PJSIP : Nobody picked up

May 28, 2015, 8:06 am

Thanks for the reply,

For using PJSIP:
-- When an endpoint registers I get only this :
"-- Added contact 'sip:6001@My.IP.IP.IP:14737;transport=UDP;rinstance=68516eaae9d9ff89' to AOR '6001' with expiration of 3600 seconds
[Jun 1 15:32:27] NOTICE[7757]: res_pjsip_mwi.c:679 mwi_new_subscribe: AOR 6001 has no configured mailboxes. MWI subscription failed
[Jun 1 15:32:27] WARNING[7758]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
"
-- When try to make a call from 6003 extension to 6001extension, I get this:
"[Jun 1 15:37:52] WARNING[7839]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Jun 1 15:37:52] WARNING[7839]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Jun 1 15:37:52] WARNING[7839]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
[Jun 1 15:37:52] WARNING[7839]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
-- Executing [6001@from-internal:1] Dial("PJSIP/6003-00000002", "PJSIP/6001,20") in new stack
-- Called PJSIP/6001
-- Nobody picked up in 20000 ms
-- Auto fallthrough, channel 'PJSIP/6003-00000002' status is 'NOANSWER'
[Jun 1 15:38:13] WARNING[7839]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Jun 1 15:38:13] WARNING[7839]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
[Jun 1 15:38:13] WARNING[7839]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Jun 1 15:38:13] WARNING[7840]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo"

==> This is a failed call.
----------------------------------------------------------------------------------------------------------------------------------------------
Comparing to PJSIP, when using the old "chan_sip", I get :
-- When an extension registers, I get this :
" Registered SIP '6001' at 10.82.3.1:45665
> Saved useragent "Z 3.3.25608 r25552" for peer 6001"

-- When try to make a call from 6003 extension to 6001extension, I get this :
" == Using SIP RTP CoS mark 5
-- Executing [6001@from-internal:1] Dial("SIP/6003-00000004", "SIP/6001,20") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6001
-- SIP/6001-00000005 is ringing
-- SIP/6001-00000005 answered SIP/6003-00000004
-- Channel SIP/6003-00000004 joined 'simple_bridge' basic-bridge <a2a0cb6d-ca53-43ba-949b-665417370969>
-- Channel SIP/6001-00000005 joined 'simple_bridge' basic-bridge <a2a0cb6d-ca53-43ba-949b-665417370969>
> Bridge a2a0cb6d-ca53-43ba-949b-665417370969: switching from simple_bridge technology to native_rtp
> Remotely bridged 'SIP/6001-00000005' and 'SIP/6003-00000004' - media will flow directly between them
> Remotely bridged 'SIP/6001-00000005' and 'SIP/6003-00000004' - media will flow directly between them
-- Started music on hold, class 'default', on channel 'SIP/6001-00000005'
> 0x7faee40046c0 -- Probation passed - setting RTP source address to 10.82.3.1:8002
> 0x7faee40046c0 -- Probation passed - setting RTP source address to 10.82.3.1:8002
-- Stopped music on hold on SIP/6001-00000005
-- Channel SIP/6001-00000005 left 'native_rtp' basic-bridge <a2a0cb6d-ca53-43ba-949b-665417370969>
-- Channel SIP/6003-00000004 left 'native_rtp' basic-bridge <a2a0cb6d-ca53-43ba-949b-665417370969>
== Spawn extension (from-internal, 6001, 1) exited non-zero on 'SIP/6003-00000004'"

===> And of course using the old chan_sip never failed.

And thanks in advance.

Statistics : Posted by wilwil • on Thu May 28, 2015 8:06 am • Replies 15 • Views 598

↧

PJSIP : Nobody picked up

May 28, 2015, 8:06 am

≫ Next: PJSIP : Nobody picked up

≪ Previous: PJSIP : Nobody picked up

Ah! yes it's true one of my tests was to call 6001 to 6001.
-- But I tried from/to different endpoints but the same , the problem it's once the call can pass and often not .
-- By the way with the old "chan_sip" as you said , it's working perfectly.
-- Also, I'm testing within within a VPN network which I set using OPEN VPN: so softphones are installed in clients computers (different from Asterisk machine): each has different IP.

For PJSIP.conf is :
"""""""""""""""""""""""""""
[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.x.x.x/255.255.255.255

[6001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
transport=transport-udp-nat
auth=6001
aors=6001
direct_media=no
rtp_symmetric=yes
force_rport=yes

[6001]
type=auth
auth_type=userpass
password=xxxxxx
username=6001

[6001]
type=aor
max_contacts=2

[6003]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
transport=transport-udp-nat
auth=6003
aors=6003
direct_media=no
rtp_symmetric=yes
force_rport=yes

[6003]
type=auth
auth_type=userpass
password=xxxxxxx
username=6003

[6003]
type=aor
max_contacts=2
""""""""""""""""""""""""""""""""""""""""""""""""""""

Statistics : Posted by wilwil • on Thu May 28, 2015 8:06 am • Replies 15 • Views 598

↧

PJSIP : Nobody picked up

May 28, 2015, 8:06 am

≫ Next: PJSIP : Nobody picked up

≪ Previous: PJSIP : Nobody picked up

You have not done "pjsip set logger on" and thus there is no SIP signaling present. There's nothing else I can add to my previous reply until that is shown.

Statistics : Posted by wilwil • on Thu May 28, 2015 8:06 am • Replies 15 • Views 598

↧

PJSIP : Nobody picked up

May 28, 2015, 8:06 am

≫ Next: PJSIP : Nobody picked up

≪ Previous: PJSIP : Nobody picked up

sorry for that, I enabled now :

-- This is the log when 6003 endpoint registered from 10.82.3.6:

Code: <--- Received SIP request (688 bytes) from UDP:10.82.3.6:36769 --->
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.82.3.6:36769;branch=z9hG4bK-524287-1---aae6fd08617f7759
Max-Forwards: 70
Contact: <sip:6001@10.82.3.6:36769;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=2f3da15b
Call-ID: M0DJjW30HWu2EsecxsurAQ..
CSeq: 1 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

<--- Transmitting SIP response (493 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.82.3.6:36769;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---aae6fd08617f7759
Call-ID: M0DJjW30HWu2EsecxsurAQ..
From: <sip:6001@10.82.3.1>;tag=2f3da15b
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---aae6fd08617f7759
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1433229264/3699d0f42141836e0d749ddd4b93cec0",opaque="6c74d1a4699cbd39",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (982 bytes) from UDP:10.82.3.6:36769 --->
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.82.3.6:36769;branch=z9hG4bK-524287-1---ec490396e34fb232
Max-Forwards: 70
Contact: <sip:6001@10.82.3.6:36769;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=2f3da15b
Call-ID: M0DJjW30HWu2EsecxsurAQ..
CSeq: 2 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username="6001",realm="asterisk",nonce="1433229264/3699d0f42141836e0d749ddd4b93cec0",uri="sip:6001@10.82.3.1;transport=UDP",response="1469ffe1e59b71534a6c743df382de01",cnonce="e9cc7dff896e1d17ca52c89c71efe418",nc=00000001,qop=auth,algorithm=md5,opaque="6c74d1a4699cbd39"
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

[Jun 2 08:14:24] NOTICE[3882]: res_pjsip_mwi.c:679 mwi_new_subscribe: AOR 6001 has no configured mailboxes. MWI subscription failed
<--- Transmitting SIP response (343 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.82.3.6:36769;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---ec490396e34fb232
Call-ID: M0DJjW30HWu2EsecxsurAQ..
From: <sip:6001@10.82.3.1>;tag=2f3da15b
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---ec490396e34fb232
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (648 bytes) from UDP:10.82.3.6:36769 --->
REGISTER sip:10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1---1e24cd15822f1a35
Max-Forwards: 70
Contact: <sip:6001@41.227.185.41:22741;rinstance=f87d40b39d8500af;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=50473f16
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg..
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Allow-Events: presence, kpml
Content-Length: 0

<--- Transmitting SIP response (496 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---1e24cd15822f1a35
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg..
From: <sip:6001@10.82.3.1>;tag=50473f16
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---1e24cd15822f1a35
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1433229288/af00ed84c0264288745ad15aba761c1f",opaque="04f2cce67f4e0a18",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (937 bytes) from UDP:10.82.3.6:36769 --->
REGISTER sip:10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1---217f49534a0c019f
Max-Forwards: 70
Contact: <sip:6001@41.227.185.41:22741;rinstance=f87d40b39d8500af;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=50473f16
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg..
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username="6001",realm="asterisk",nonce="1433229288/af00ed84c0264288745ad15aba761c1f",uri="sip:10.82.3.1;transport=UDP",response="8e9035dea9cb6c1371ae0e36633d62f4",cnonce="4f98156ce965bb9d9792521edc895e28",nc=00000001,qop=auth,algorithm=md5,opaque="04f2cce67f4e0a18"
Allow-Events: presence, kpml
Content-Length: 0

-- Added contact 'sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af' to AOR '6001' with expiration of 3600 seconds
<--- Transmitting SIP response (486 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---217f49534a0c019f
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg..
From: <sip:6001@10.82.3.1>;tag=50473f16
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---217f49534a0c019f
CSeq: 2 REGISTER
Date: Tue, 02 Jun 2015 07:14:48 GMT
Contact: <sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af>;expires=3599
Expires: 3600
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (935 bytes) from UDP:10.82.3.6:36769 --->
PUBLISH sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1---c0e9d546d0d4b7f7
Max-Forwards: 70
Contact: <sip:6001@41.227.185.41:22741;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=7635ce49
Call-ID: ID4jdnY-bPyJ1Eu8u3uBVg..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Event: presence
Allow-Events: presence, kpml
Content-Length: 257

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:6001@10.82.3.1;transport=UDP">
<tuple id="6001" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>

<--- Transmitting SIP response (495 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---c0e9d546d0d4b7f7
Call-ID: ID4jdnY-bPyJ1Eu8u3uBVg..
From: <sip:6001@10.82.3.1>;tag=7635ce49
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---c0e9d546d0d4b7f7
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1433229288/af00ed84c0264288745ad15aba761c1f",opaque="2b05028c0fa8f159",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (687 bytes) from UDP:10.82.3.6:36769 --->
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1---7eda2b41003d4900
Max-Forwards: 70
Contact: <sip:6001@41.227.185.41:22741;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=920f5a46
Call-ID: PI5K6X2W0E_MmwMrk_LhUQ..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<--- Transmitting SIP response (497 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---7eda2b41003d4900
Call-ID: PI5K6X2W0E_MmwMrk_LhUQ..
From: <sip:6001@10.82.3.1>;tag=920f5a46
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---7eda2b41003d4900
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1433229288/af00ed84c0264288745ad15aba761c1f",opaque="61d6d4d74040ebb4",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (1229 bytes) from UDP:10.82.3.6:36769 --->
PUBLISH sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1---a25e216e831f79b3
Max-Forwards: 70
Contact: <sip:6001@41.227.185.41:22741;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=7635ce49
Call-ID: ID4jdnY-bPyJ1Eu8u3uBVg..
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username="6001",realm="asterisk",nonce="1433229288/af00ed84c0264288745ad15aba761c1f",uri="sip:6001@10.82.3.1;transport=UDP",response="18c4dc04b29040f32f8191806e4fc99b",cnonce="b2cbbb25f1b6196f321c1a87d4a213c8",nc=00000001,qop=auth,algorithm=md5,opaque="2b05028c0fa8f159"
Event: presence
Allow-Events: presence, kpml
Content-Length: 257

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:6001@10.82.3.1;transport=UDP">
<tuple id="6001" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>

[Jun 2 08:14:48] WARNING[3882]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
<--- Transmitting SIP response (345 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---a25e216e831f79b3
Call-ID: ID4jdnY-bPyJ1Eu8u3uBVg..
From: <sip:6001@10.82.3.1>;tag=7635ce49
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---a25e216e831f79b3
CSeq: 2 PUBLISH
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (696 bytes) from UDP:10.82.3.6:36769 --->
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1---db522e91d922bbfd
Max-Forwards: 70
Contact: <sip:6001@41.227.185.41:22741;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=87708c2d
Call-ID: OqoWdMUgmKwWX79QbYRWgw..
CSeq: 1 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

<--- Transmitting SIP response (497 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---db522e91d922bbfd
Call-ID: OqoWdMUgmKwWX79QbYRWgw..
From: <sip:6001@10.82.3.1>;tag=87708c2d
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---db522e91d922bbfd
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1433229288/af00ed84c0264288745ad15aba761c1f",opaque="62dc3a765cff63e9",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (981 bytes) from UDP:10.82.3.6:36769 --->
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1---fc55bb12bada1719
Max-Forwards: 70
Contact: <sip:6001@41.227.185.41:22741;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=920f5a46
Call-ID: PI5K6X2W0E_MmwMrk_LhUQ..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username="6001",realm="asterisk",nonce="1433229288/af00ed84c0264288745ad15aba761c1f",uri="sip:6001@10.82.3.1;transport=UDP",response="503741850430785ca5a2cdb0a3a2803e",cnonce="4a86d18d8f6eb388d37fcddb7287218e",nc=00000001,qop=auth,algorithm=md5,opaque="61d6d4d74040ebb4"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

[Jun 2 08:14:48] WARNING[3882]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
<--- Transmitting SIP response (347 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---fc55bb12bada1719
Call-ID: PI5K6X2W0E_MmwMrk_LhUQ..
From: <sip:6001@10.82.3.1>;tag=920f5a46
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---fc55bb12bada1719
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (990 bytes) from UDP:10.82.3.6:36769 --->
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1---94ebda0155275009
Max-Forwards: 70
Contact: <sip:6001@41.227.185.41:22741;transport=UDP>
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=87708c2d
Call-ID: OqoWdMUgmKwWX79QbYRWgw..
CSeq: 2 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username="6001",realm="asterisk",nonce="1433229288/af00ed84c0264288745ad15aba761c1f",uri="sip:6001@10.82.3.1;transport=UDP",response="28f3dd18a082f3941397f3aa6f7bd014",cnonce="528f98f4ffbf24005b79553a7fed6797",nc=00000001,qop=auth,algorithm=md5,opaque="62dc3a765cff63e9"
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

[Jun 2 08:14:48] NOTICE[3882]: res_pjsip_mwi.c:679 mwi_new_subscribe: AOR 6001 has no configured mailboxes. MWI subscription failed
<--- Transmitting SIP response (347 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---94ebda0155275009
Call-ID: OqoWdMUgmKwWX79QbYRWgw..
From: <sip:6001@10.82.3.1>;tag=87708c2d
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---94ebda0155275009
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 13.3.0
Content-Length: 0

-- This is when I try to call 6001 from 6003:

Code: -- Executing [6001@from-internal:1] Dial("PJSIP/6003-00000002", "PJSIP/6001,20") in new stack
-- Called PJSIP/6001
<--- Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 --->
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: <sip:6003@192.168.1.2>;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: <sip:6001@41.227.185.41;rinstance=f87d40b39d8500af>
Contact: <sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060>
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 --->
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: <sip:6003@192.168.1.2>;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: <sip:6001@41.227.185.41;rinstance=f87d40b39d8500af>
Contact: <sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060>
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 --->
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: <sip:6003@192.168.1.2>;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: <sip:6001@41.227.185.41;rinstance=f87d40b39d8500af>
Contact: <sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060>
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 --->
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: <sip:6003@192.168.1.2>;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: <sip:6001@41.227.185.41;rinstance=f87d40b39d8500af>
Contact: <sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060>
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (984 bytes) to UDP:41.227.185.41:22741 --->
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj637cbe2e-65d6-473e-af34-262392012214
From: <sip:6003@192.168.1.2>;tag=0d40f5de-1785-4587-b09c-21931bf8c27b
To: <sip:6001@41.227.185.41;rinstance=f87d40b39d8500af>
Contact: <sip:f436ecb0-ba1c-4ff1-ab4e-ba3e54746837@41.227.246.48:5060>
Call-ID: 8fedc553-3bd9-4c40-b3e4-d096e4735834
CSeq: 3370 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 235

v=0
o=- 1398057981 1398057981 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 10762 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 --->
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: <sip:6003@192.168.1.2>;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: <sip:6001@41.227.185.41;rinstance=f87d40b39d8500af>
Contact: <sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060>
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 --->
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: <sip:6003@192.168.1.2>;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: <sip:6001@41.227.185.41;rinstance=f87d40b39d8500af>
Contact: <sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060>
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Nobody picked up in 20000 ms
-- Auto fallthrough, channel 'PJSIP/6003-00000002' status is 'NOANSWER'
<--- Transmitting SIP response (394 bytes) to UDP:10.82.3.1:52170 --->
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-ef571ce00d5612c5-1---d8754z-
Call-ID: ZmJkNWNhZThjMTI3NDZhMzkxY2Q0ODgzMTYwOWRkYzc.
From: <sip:6003@10.82.3.1>;tag=1bc1c118
To: <sip:6001@10.82.3.1>;tag=156db271-1bfe-4b85-bd03-967cb38c407a
CSeq: 2 INVITE
Server: Asterisk PBX 13.3.0
Reason: Q.850;cause=0
Content-Length: 0

<--- Received SIP request (364 bytes) from UDP:10.82.3.1:52170 --->
ACK sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-ef571ce00d5612c5-1---d8754z-
Max-Forwards: 70
To: <sip:6001@10.82.3.1>;tag=156db271-1bfe-4b85-bd03-967cb38c407a
From: <sip:6003@10.82.3.1;transport=UDP>;tag=1bc1c118
Call-ID: ZmJkNWNhZThjMTI3NDZhMzkxY2Q0ODgzMTYwOWRkYzc.
CSeq: 2 ACK
Content-Length: 0

<--- Received SIP request (947 bytes) from UDP:10.82.3.1:52170 --->
PUBLISH sip:6003@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-1d2ea0e06f9ddf6b-1---d8754z-
Max-Forwards: 70
Contact: <sip:6003@41.227.185.41:22377;transport=UDP>
To: <sip:6003@10.82.3.1;transport=UDP>
From: <sip:6003@10.82.3.1;transport=UDP>;tag=80d73a07
Call-ID: NGE3YzI0MWQ4OWQwY2YzMWRkZWY1NWJlNjE5N2NlMTU.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 257

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:6003@10.82.3.1;transport=UDP">
<tuple id="6003" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>

<--- Received SIP request (699 bytes) from UDP:10.82.3.1:52170 --->
SUBSCRIBE sip:6003@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-7dbade96a2574170-1---d8754z-
Max-Forwards: 70
Contact: <sip:6003@41.227.185.41:22377;transport=UDP>
To: <sip:6003@10.82.3.1;transport=UDP>
From: <sip:6003@10.82.3.1;transport=UDP>;tag=cc230f28
Call-ID: M2EwNWZkZDAzMzhlNzcwMDIwZTk0MmNlYmE4YmMxNDE.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<--- Transmitting SIP response (531 bytes) to UDP:10.82.3.1:52170 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-1d2ea0e06f9ddf6b-1---d8754z-
Call-ID: NGE3YzI0MWQ4OWQwY2YzMWRkZWY1NWJlNjE5N2NlMTU.
From: <sip:6003@10.82.3.1>;tag=80d73a07
To: <sip:6003@10.82.3.1>;tag=z9hG4bK-d8754z-1d2ea0e06f9ddf6b-1---d8754z-
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1433229420/95a537f084952af3a3ea68d64c087584",opaque="42a2d1f4470b23b1",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Transmitting SIP response (533 bytes) to UDP:10.82.3.1:52170 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-7dbade96a2574170-1---d8754z-
Call-ID: M2EwNWZkZDAzMzhlNzcwMDIwZTk0MmNlYmE4YmMxNDE.
From: <sip:6003@10.82.3.1>;tag=cc230f28
To: <sip:6003@10.82.3.1>;tag=z9hG4bK-d8754z-7dbade96a2574170-1---d8754z-
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1433229420/95a537f084952af3a3ea68d64c087584",opaque="5bad92ba26acf5d2",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (1241 bytes) from UDP:10.82.3.1:52170 --->
PUBLISH sip:6003@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-865df2942948418d-1---d8754z-
Max-Forwards: 70
Contact: <sip:6003@41.227.185.41:22377;transport=UDP>
To: <sip:6003@10.82.3.1;transport=UDP>
From: <sip:6003@10.82.3.1;transport=UDP>;tag=80d73a07
Call-ID: NGE3YzI0MWQ4OWQwY2YzMWRkZWY1NWJlNjE5N2NlMTU.
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6003",realm="asterisk",nonce="1433229420/95a537f084952af3a3ea68d64c087584",uri="sip:6003@10.82.3.1;transport=UDP",response="7d19d5516b062c4950af3b66e422daa8",cnonce="69e0be5ff1cedcd3f3fc1f6c4d2768b6",nc=00000001,qop=auth,algorithm=md5,opaque="42a2d1f4470b23b1"
Event: presence
Allow-Events: presence, kpml
Content-Length: 257

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:6003@10.82.3.1;transport=UDP">
<tuple id="6003" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>

<--- Received SIP request (993 bytes) from UDP:10.82.3.1:52170 --->
SUBSCRIBE sip:6003@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-e84caa37b87f1926-1---d8754z-
Max-Forwards: 70
Contact: <sip:6003@41.227.185.41:22377;transport=UDP>
To: <sip:6003@10.82.3.1;transport=UDP>
From: <sip:6003@10.82.3.1;transport=UDP>;tag=cc230f28
Call-ID: M2EwNWZkZDAzMzhlNzcwMDIwZTk0MmNlYmE4YmMxNDE.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6003",realm="asterisk",nonce="1433229420/95a537f084952af3a3ea68d64c087584",uri="sip:6003@10.82.3.1;transport=UDP",response="47585f59499796484eea24703badf14d",cnonce="cd9fcbea6e3890f70d61057bc85362f8",nc=00000001,qop=auth,algorithm=md5,opaque="5bad92ba26acf5d2"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

[Jun 2 08:17:00] WARNING[4028]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
<--- Transmitting SIP response (381 bytes) to UDP:10.82.3.1:52170 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-865df2942948418d-1---d8754z-
Call-ID: NGE3YzI0MWQ4OWQwY2YzMWRkZWY1NWJlNjE5N2NlMTU.
From: <sip:6003@10.82.3.1>;tag=80d73a07
To: <sip:6003@10.82.3.1>;tag=z9hG4bK-d8754z-865df2942948418d-1---d8754z-
CSeq: 2 PUBLISH
Server: Asterisk PBX 13.3.0
Content-Length: 0

[Jun 2 08:17:00] WARNING[4028]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
<--- Transmitting SIP response (383 bytes) to UDP:10.82.3.1:52170 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-e84caa37b87f1926-1---d8754z-
Call-ID: M2EwNWZkZDAzMzhlNzcwMDIwZTk0MmNlYmE4YmMxNDE.
From: <sip:6003@10.82.3.1>;tag=cc230f28
To: <sip:6003@10.82.3.1>;tag=z9hG4bK-d8754z-e84caa37b87f1926-1---d8754z-
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (928 bytes) from UDP:10.82.3.6:36769 --->
REGISTER sip:10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.82.3.6:36769;branch=z9hG4bK-524287-1---f16af0e0002d56d8
Max-Forwards: 70
Contact: <sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af>;expires=0
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=50473f16
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg..
CSeq: 3 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username="6001",realm="asterisk",nonce="1433229288/af00ed84c0264288745ad15aba761c1f",uri="sip:10.82.3.1;transport=UDP",response="6780609ec66cbce6620932772d983410",cnonce="2a74c6f7e67c8f59db2b5d04ba4658b4",nc=00000002,qop=auth,algorithm=md5,opaque="04f2cce67f4e0a18"
Allow-Events: presence, kpml
Content-Length: 0

<--- Transmitting SIP response (503 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.82.3.6:36769;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---f16af0e0002d56d8
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg..
From: <sip:6001@10.82.3.1>;tag=50473f16
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---f16af0e0002d56d8
CSeq: 3 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1433229429/a5a58229b5db5ac03171753460566da4",opaque="7fb07b9807a7f2ec",stale=true,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Received SIP request (928 bytes) from UDP:10.82.3.6:36769 --->
REGISTER sip:10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.82.3.6:36769;branch=z9hG4bK-524287-1---1ed6831cf6a85375
Max-Forwards: 70
Contact: <sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af>;expires=0
To: <sip:6001@10.82.3.1;transport=UDP>
From: <sip:6001@10.82.3.1;transport=UDP>;tag=50473f16
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg..
CSeq: 4 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username="6001",realm="asterisk",nonce="1433229429/a5a58229b5db5ac03171753460566da4",uri="sip:10.82.3.1;transport=UDP",response="ba95d4b1ecd561328befe44065529665",cnonce="a94932f2d5daa9657d6dea6bcf81c2d1",nc=00000001,qop=auth,algorithm=md5,opaque="7fb07b9807a7f2ec"
Allow-Events: presence, kpml
Content-Length: 0

-- Attempted to remove non-existent contact 'sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af' from AOR '6001' by request
<--- Transmitting SIP response (467 bytes) to UDP:10.82.3.6:36769 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.82.3.6:36769;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1---1ed6831cf6a85375
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg..
From: <sip:6001@10.82.3.1>;tag=50473f16
To: <sip:6001@10.82.3.1>;tag=z9hG4bK-524287-1---1ed6831cf6a85375
CSeq: 4 REGISTER
Date: Tue, 02 Jun 2015 07:17:09 GMT
Contact: <sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af>;expires=3458
Server: Asterisk PBX 13.3.0
Content-Length: 0

<--- Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 --->
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: <sip:6003@192.168.1.2>;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: <sip:6001@41.227.185.41;rinstance=f87d40b39d8500af>
Contact: <sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060>
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Statistics : Posted by wilwil • on Thu May 28, 2015 8:06 am • Replies 15 • Views 598

↧

PJSIP : Nobody picked up

May 28, 2015, 8:06 am

≫ Next: PJSIP : Nobody picked up

≪ Previous: PJSIP : Nobody picked up

The NOTICE is is occurring because the endpoint is trying to subscribe to receive voicemail information, but no mailboxes have been configured.

The WARNING is occurring because the endpoint is attempting to publish presence information but this is not currently supported.

As for your problem please turn on packet output using "pjsip set logger on" and provide the console output when an endpoint registers and when you call it and it fails.

Statistics : Posted by wilwil • on Thu May 28, 2015 8:06 am • Replies 15 • Views 598

↧

PJSIP : Nobody picked up

May 28, 2015, 8:06 am

≫ Next: PJSIP : Nobody picked up

≪ Previous: PJSIP : Nobody picked up

Your problem is either that you are calling the same softphone from which you are making the call or you have multiple endpoints registered to the same AOR.

This message:

Quote:Added contact 'sip:6001@10.82.3.6:61506;rinstance=3ef662af83ebf79d' to AOR '6001' with expiration of 3600 seconds

Indicates that your softphone successfully registered with an endpoint with AOR 6001. The command to dial this device is Dial(PJSIP/6001)

This message:

Quote:[May 28 10:32:15] NOTICE[12893]: res_pjsip_mwi.c:679 mwi_new_subscribe: AOR 6001 has no configured mailboxes. MWI subscription failed"

Means that you dont have a mailbox configured for this AOR. You can ignore this for now as it wont prevent you from making or receiving calls.

This patch:https://forum.snom.com/index.php?showtopic=14573 Is not related to these warnings.

You can ignore this.

Quote:"res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[May 28 11:17:49] WARNING[13537]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo"

This is just the device trying to send a PUBLISH event that is not supported by Asterisk.

pjsip show registrations wont show because this commands lists outbound registrations and you are using inbound registrations, which is correct for registering softphones.

pjsip list endpoints is correct to say "Not in use" because that is the state of the phone when its not on a call or ringing.

This right here

Quote:Executing [6001@from-internal:1] Dial("PJSIP/6001-0000000a", "PJSIP/6001,20") in new stack
-- Called PJSIP/6001
-- Nobody picked up in 20000 ms
-- Auto fallthrough, channel 'PJSIP/6002-00000008' status is 'NOANSWER'"

This seems to say you are calling from PJSIP/6001 to PJSIP/6001. I'm not sure why there is the Auto fallthrough, channel 'PJSIP/6002-00000008' status is 'NOANSWER' You may be trying to register multiple endpoints to the same AOR? I'll bet if you look closely at your softphone, it will show an incoming call while you are making this outbound call. This is like calling your cellphone from your cellphone, it doesnt work. What you need to do is register one phone with AOR 6001 and the other with AOR 6002. Post your pjsip.conf to be sure.

As an alternative, I'd recommend (I might be outvoted on this) starting with the old sip stack rather than pjsip. It is easier for beginners to learn and understand, in my opinion. Once you have sip figured out, you can look at how it translates to pjsip.

You should start with the O'Reily Asterisk book. Edition 3 is available for free online here http://asteriskdocs.org/ See the section on Online Versions in the middle of the page. It doesn't cover chan_pjsip but it will cover chan_sip in enough detail to get you started and understand how Asterisk connects channels and devices.

Statistics : Posted by wilwil • on Thu May 28, 2015 8:06 am • Replies 15 • Views 598

↧

PJSIP : Nobody picked up

May 28, 2015, 8:06 am

≫ Next: Outbound call: Received response "Fordbidden" and Red Bar

≪ Previous: PJSIP : Nobody picked up

Hi,

I'm new in Asterisk field. But I want to learn more about this great soft.

So I'll explain the issue:

- I'm working with Asterisk 13.3 and latest pjsip version.

- I configured OpenVPN tunnel to pass audio calls through it . The tunnel works and clients can connect to OpenVPN server: and softphones show registered when connected to VPN server. (might be the cause)

- When softphones register to Asterisk only these warnings are showing; I though that my problem is like this one ""https://forum.snom.com/index.php?showtopic=14573"" , but is the same when I added the patch as described there:

I get only this :

" Added contact 'sip:6001@10.82.3.6:61506;rinstance=3ef662af83ebf79d' to AOR '6001' with expiration of 3600 seconds
[May 28 10:32:15] NOTICE[12893]: res_pjsip_mwi.c:679 mwi_new_subscribe: AOR 6001 has no configured mailboxes. MWI subscription failed"

And when softphones try to register : I get this :

"res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[May 28 11:17:49] WARNING[13537]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo"

But I really don't know if this is a complete registration:
** "pjsip show registrations" command doesn't show nothing despite softphones say registered.
** " pjsip list endpoints" command shows all endpoints but with "not in use" as status.

--- The problem is when trying to make calls between endpoints: sometimes it works and many times don't ; Asterisk showing only this :

"Executing [6001@from-internal:1] Dial("PJSIP/6001-0000000a", "PJSIP/6001,20") in new stack
-- Called PJSIP/6001
-- Nobody picked up in 20000 ms
-- Auto fallthrough, channel 'PJSIP/6002-00000008' status is 'NOANSWER'"

And if it worked, when called endpoint "hang up the call", the caller endpoint don't figure out that (I mean that the call keeps running in caller side) !!!!

Please have you any ideas, I have been blocked here for several days??? knowing that all was fine and worked when I tried with asterisk and endpoints in the same subnet with the same pjsip version.

Statistics : Posted by wilwil • on Thu May 28, 2015 8:06 am • Replies 15 • Views 598

↧

Outbound call: Received response "Fordbidden" and Red Bar

June 5, 2015, 12:36 pm

≫ Next: SBO: VoIP Bandwidth Optimizer-Do you have any recommendation

≪ Previous: PJSIP : Nobody picked up

Hi!

I'm trying to make outbound call, but I received the message Received response: "Fordbidden"

Code: == Using SIP RTP CoS mark 5
-- Executing [01145550711@ramais:1] Dial("SIP/3028-00000048", "SIP/tellfree/01145550711,30,tT") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/tellfree/01145550711
[Jun 5 15:31:48] WARNING[23638][C-00000039]: chan_sip.c:23159 handle_response_invite: Received response: "Forbidden" from '<sip:=sip.tellfree.net@sip.tellfree.net:5089>;tag=as1d2e0d26'
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/3028-00000048' status is 'CHANUNAVAIL'

See my sip.conf

Code: [general]
allowguest=no
autocreatepeer=no
awayssauthreject=yes
udpbindaddr=0.0.0.0:5089
context=ramais
disallow=all
allow=ulaw;gsm
externhost=mydomain.noip.us:5089
localnet=192.3.1.2/255.255.255.0
register => ******:******@sip.tellfree.net
[tellfree]
type=peer
defaultuser=******
secret=******
context=ramais
host=sip.tellfree.net
insecure=port,invite
qualify=yes
fromdomain=sip.tellfree.net
fromuser==sip.tellfree.net
allow=gsm
dtmfmode=rfc2833
directmedia=no
;;

Why appear "Fordbidden"? And Can't I make any calls?

And I have one question, when I put the address in my host, appear a strange red bar plaese see below

[img]
https://uploaddeimagens.com.br/imagens/ ... png--11994
[/img]

When I use

Code: host=

appear this red bar, but it works, but when I use

Code: hosts=

the red bar disapear but it not work. Why this happen?

Statistics : Posted by vitormazuco • on Fri Jun 05, 2015 12:36 pm • Replies 8 • Views 285

↧

SBO: VoIP Bandwidth Optimizer-Do you have any recommendation

September 25, 2014, 4:54 am

≫ Next: SBO: VoIP Bandwidth Optimizer-Do you have any recommendation

≪ Previous: Outbound call: Received response "Fordbidden" and Red Bar

mehroz wrote:Here is something that I have used , page_id=68

Thanks Mehroz, but I am a little bit confused about your propose products. So for I know that SBO has introduced by SYNCHRONOUS ICT in global market with the brand name of SBO which means Synchronous Bandwidth Optimizer but your propose company also using same brand name. I this this is fake and you know it is illegal to use other company's brand name.

Statistics : Posted by synchronous • on Thu Sep 25, 2014 4:54 am • Replies 2 • Views 1571

↧

SBO: VoIP Bandwidth Optimizer-Do you have any recommendation

September 25, 2014, 4:54 am

≫ Next: SBO: VoIP Bandwidth Optimizer-Do you have any recommendation

≪ Previous: SBO: VoIP Bandwidth Optimizer-Do you have any recommendation

Here is something that I have used , http://www.voipography.com/?page_id=68

Statistics : Posted by synchronous • on Thu Sep 25, 2014 4:54 am • Replies 2 • Views 1571

↧

SBO: VoIP Bandwidth Optimizer-Do you have any recommendation

September 25, 2014, 4:54 am

≫ Next: Access your IP PBX using voice channels

≪ Previous: SBO: VoIP Bandwidth Optimizer-Do you have any recommendation

I got some idea about SBO, that it reduce bandwidth cost, increase ACD, Block free and support all types of internet.

They offer service as monthly rental basis and price is also very reasonable.

So, could you guys tell me which one is best in global market? I think SBO!

webpage: http://www.syncswitch.com/content/sbo-m ... h-features

Thanks

Statistics : Posted by synchronous • on Thu Sep 25, 2014 4:54 am • Replies 2 • Views 1571

↧

Access your IP PBX using voice channels

June 7, 2015, 2:08 am

≫ Next: Access your IP PBX using voice channels

≪ Previous: SBO: VoIP Bandwidth Optimizer-Do you have any recommendation

Thanks for your comment.

Indeed this action might be considered hostile in some situations such as very restrictive policied netwoks. I agree with you, thank you for pointed out.
Hoping that in other scenarios this still could be useful, I'll wait to read other comments.
At last, please consider that: using this solution you do not need any additional hardware, such as additional analog modems.

Regards
AC

Statistics : Posted by acarminati • on Sun Jun 07, 2015 2:08 am • Replies 2 • Views 65

↧

Access your IP PBX using voice channels

June 7, 2015, 2:08 am

≫ Next: Access your IP PBX using voice channels

≪ Previous: Access your IP PBX using voice channels

If there are such restrictive policies, trying to circumvent them may be considered a breach of the policy. However you can certainly connect a modem to a voice channel and I believe there are modec implementations that are suitable for direct connection.

Statistics : Posted by acarminati • on Sun Jun 07, 2015 2:08 am • Replies 2 • Views 65

↧

Access your IP PBX using voice channels

June 7, 2015, 2:08 am

≫ Next: BT ISDN30 and Digium Wildcard TE 133F

≪ Previous: Access your IP PBX using voice channels

This post is to propose you an experiment I did not so long ago.
I had the need to access, for support reasons, an Asterisk IP PBX which does not have any IP connection, or to better say, which is not allowed to be connected through the IP connection already sitting there.

In spite this might sound absurd, it happen more than one can imagine, at least in my experience.

Reasons are many, very restrictive network policies, not so cooperative network administrators or also PBX is just installed in places where there's only voice connection. This is the case, for example, of installation of PBXes in small shops or small laboratory of various nature.

In this cases, do any change on the configuration of those devices may require you to move yourself to the device location.

When there's no other way, why don't use the voice channels already sitting there to access the server?

I don't want to hide that this solution is slow by nature, the best you can do is 38400bps, and this is not quite simple to reach, in my case I couldn't exceded 9600 bps. However you can remotely manage your installation without beg anyone's help for a remote access!

Hoping someone else can find this useful, I wrote an article on my blog. http://carminatialessandro.blogspot.it/ ... nnels.html

I'd be really glad to read your comments.
Regards
AC

Statistics : Posted by acarminati • on Sun Jun 07, 2015 2:08 am • Replies 2 • Views 65

↧

BT ISDN30 and Digium Wildcard TE 133F

June 6, 2015, 6:35 pm

≫ Next: BT ISDN30 and Digium Wildcard TE 133F

≪ Previous: Access your IP PBX using voice channels

The issue is /etc/modprobe.d/dahdi.conf is overwritten by freepbx

-----------------------------------------------------------------------------$
Do NOT edit this file as it is auto-generated by FreePBX. All modifications t$
this file must be done via the web gui. There are alternative files to make $
custom modifications, details at: http://freepbx.org/configuration_files $
-----------------------------------------------------------------------------$
I am expecting {modprobe wcte13xp default_linemode=e1} is set in the GUI of freepbx somewhere.

How do you set the card to be e1 in freepbx if the config file is overwriten by the gui?

Thanks

Statistics : Posted by londonnet • on Sat Jun 06, 2015 6:35 pm • Replies 2 • Views 81

↧

BT ISDN30 and Digium Wildcard TE 133F

June 6, 2015, 6:35 pm

≫ Next: BT ISDN30 and Digium Wildcard TE 133F

≪ Previous: BT ISDN30 and Digium Wildcard TE 133F

Why don't you qualify for support from Digium? The proper place to get configuration support for current Digium products is through their commercial support channels.

In any case, what is your problem with the description of how to do this is in https://www.digium.com/sites/digium/fil ... manual.pdf

PS I don't understand the reference to FXS and FXO. These basically refer to analogue signalling and should not be relevant to primary rate devices, particularly at they physical level. The nearest equivalent is which end provides the timing.

Statistics : Posted by londonnet • on Sat Jun 06, 2015 6:35 pm • Replies 2 • Views 81

↧

BT ISDN30 and Digium Wildcard TE 133F

June 6, 2015, 6:35 pm

≫ Next: PJSIP : Nobody picked up

≪ Previous: BT ISDN30 and Digium Wildcard TE 133F

Hi. I am installing asteriskNow and using a Digium Wildcard TE 133F card. The card is auto detected in dahdi config the card shows up as a Digium Wildcard TE131/TE133 card 0 [1]

However the settings for this card are set by default as a T1 and not as E1 as required for UK BT ISDN30

The closest matching card I can see in Modprobe Settings is the wcte13xp however the config options for this card look they are for a FXS card and not a PRI card.

If I set this to UK etc save the settings and reboot the card is not longer listed for further configuration.

Can somone tell me what the correct way is to configure a Digium Wildcard TE 133F is for the UK and more specificaly for a BT ISDN30 PRI E1?

Many thanks

Statistics : Posted by londonnet • on Sat Jun 06, 2015 6:35 pm • Replies 2 • Views 81

↧

PJSIP : Nobody picked up

May 28, 2015, 8:06 am

≫ Next: PJSIP : Nobody picked up

≪ Previous: BT ISDN30 and Digium Wildcard TE 133F

Yes I'm using a private network which I configured using OPEN VPN to put all endpoints under the same subnet.
For the port:
- I configured my router to be able to send/receive from the standard Asterisk ports 'udp 5060' and "10000 - 20000".
- I configured my firewall also to be able to receive/send from VPN interface.

Statistics : Posted by wilwil • on Thu May 28, 2015 8:06 am • Replies 22 • Views 932

↧

PJSIP : Nobody picked up

May 28, 2015, 8:06 am

≫ Next: PJSIP : Nobody picked up

≪ Previous: PJSIP : Nobody picked up

The register has specified a public address and a non-standard port number, but has come from a private address (and a different port number). I would look at the configuration of the peer and any firewall and NAT devices on the way.

Statistics : Posted by wilwil • on Thu May 28, 2015 8:06 am • Replies 22 • Views 932

↧

PJSIP : Nobody picked up

May 28, 2015, 8:06 am

≫ Next: PJSIP : Nobody picked up

≪ Previous: PJSIP : Nobody picked up

↧