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PJSIP IPV6 Transport does not bind to ::

December 12, 2014, 8:53 am
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Hm,

I compiled it straight as it came from the asterisk repository and made no change to any configure file.

I will look if I can figure it out.

Statistics : Posted by rbasche • on Fri Dec 12, 2014 10:53 am • Replies 8 • Views 191
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PJSIP IPV6 Transport does not bind to ::

December 12, 2014, 8:53 am
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PJSIP IPV6 Transport does not bind to ::

December 12, 2014, 8:53 am
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It also does not bind to a fully qualified address:

[Dec 12 18:03:16] DEBUG[11475] config.c: Parsing /etc/asterisk/pjsip.conf
[Dec 12 18:03:16] VERBOSE[11475] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Dec 12 18:03:16] DEBUG[11475] config.c: extract int from [0] in [0, 65535] gives [0](0)
[Dec 12 18:03:16] DEBUG[11475] config.c: extract int from [100] in [1, 2147483647] gives [100](0)
[Dec 12 18:03:16] DEBUG[11475] config.c: extract int from [1] in [-2147483648, 2147483647] gives [1](0)
[Dec 12 18:03:16] DEBUG[11475] config.c: extract int from [0] in [-2147483648, 2147483647] gives [0](0)
[Dec 12 18:03:16] ERROR[11475] config_options.c: Error parsing bind=fe80::ba27:ebff:feda:bbb6 at line 8 of
[Dec 12 18:03:16] ERROR[11475] res_sorcery_config.c: Could not create an object of type 'transport' with id 'udp-ipv6' from configuration file 'pjsip.conf'

Statistics : Posted by rbasche • on Fri Dec 12, 2014 10:53 am • Replies 8 • Views 191
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PJSIP IPV6 Transport does not bind to ::

December 12, 2014, 8:53 am
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Hello,

I have newly installed Asterisk 13.0.1 on my raspberry box.

I am triyng to set up PJSIP using an IPV6 transport.

The box has a global address and two local addresses:

asterisk@raspbx ~ $ ifconfig -a
eth0 Link encap:Ethernet HWaddr b8:27:eb:da:bb:b6
inet addr:192.168.178.99 Bcast:192.168.178.255 Mask:255.255.255.0
inet6 addr: fe80::ba27:ebff:feda:bbb6/64 Scope:Link
inet6 addr: 2a02:8070:8680:9520:ba27:ebff:feda:bbb6/64 Scope:Global
inet6 addr: fd00::ba27:ebff:feda:bbb6/64 Scope:Global
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:23373 errors:0 dropped:0 overruns:0 frame:0
TX packets:26482 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:2431636 (2.3 MiB) TX bytes:11339366 (10.8 MiB)


The transport definition in pjsip.conf is:

[udp-ipv6]
type=transport
protocol=udp
bind=::

as it is described in the documentation to have it binded to all available addresses

I get the following message when starting up asterisk:

[Dec 12 00:58:23] Asterisk 13.0.1 built by root @ raspberrypi on a armv6l running Linux on 2014-12-06 17:25:37 UTC
[Dec 12 00:58:24] NOTICE[10157] cdr.c: CDR simple logging enabled.
[Dec 12 00:58:26] NOTICE[10157] loader.c: 232 modules will be loaded.
[Dec 12 00:58:26] WARNING[10157] loader.c: Error loading module 'res_monitor.so': /usr/lib/asterisk/modules/res_monitor.so: undefined symbol: __ast_beep_stop
[Dec 12 00:58:31] ERROR[10157] config_options.c: Could not find option suitable for category 'easybell' named 'aors' at line 34 of
[Dec 12 00:58:31] ERROR[10157] res_sorcery_config.c: Could not create an object of type 'auth' with id 'easybell' from configuration file 'pjsip.conf'
[Dec 12 00:58:31] ERROR[10157] config_options.c: Error parsing bind=:: at line 8 of
[Dec 12 00:58:31] ERROR[10157] res_sorcery_config.c: Could not create an object of type 'transport' with id 'udp-ipv6' from configuration file 'pjsip.conf'

Is there an alternate way to bind asterisk to all available IPV6 addresses, I do not want to use a specific address, as the address is given by the ISP and may change over time.

(I did not yet try to bind asterisk to the global address)

Statistics : Posted by rbasche • on Fri Dec 12, 2014 10:53 am • Replies 8 • Views 191
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Extesion for ip Deskphnone

December 12, 2014, 11:28 am
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Ok for now i work with only some computers with sip client installed,i think soon i will purchase ip phone designed for Asterisk PBX.

Thanks,

Statistics : Posted by tella • on Fri Dec 12, 2014 1:28 pm • Replies 4 • Views 115
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Extesion for ip Deskphnone

December 12, 2014, 11:28 am
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It is generally inadvisable to use phones designed for proprietary PABXes on anything but that PABX. Although they may be technically SIP compliant, they may depend on the PABX for configuration. I can't say whether or not the Avaya 96xx phones suffer from such problems, although it does look as though some are intended to mainly be used with H.323, rather than SIP.

Statistics : Posted by tella • on Fri Dec 12, 2014 1:28 pm • Replies 4 • Views 115
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Extesion for ip Deskphnone

December 12, 2014, 11:28 am
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Alright thank you,for now i only focus on internall call between a few extensions,it's my first try on ASTERISK FREE PBX that i want to increase my knowledge Image ... And then it is also the same procedure to make a SIP Deskphone works with Asterisk
Do you think that a 9600 series Avaya phone going to match with Astrerisk Server Image ?

Thanks,

Statistics : Posted by tella • on Fri Dec 12, 2014 1:28 pm • Replies 4 • Views 115
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Extesion for ip Deskphnone

December 12, 2014, 11:28 am
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You are using terminology which suggests you are using FreePBX, but using a SIP desk phone with Asterisk is probably still more common than using mobile devices.

Statistics : Posted by tella • on Fri Dec 12, 2014 1:28 pm • Replies 4 • Views 115
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Extesion for ip Deskphnone

December 12, 2014, 11:28 am
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Hello guys,
A few days ago i configured two SIP extensions on a Asterisk server,on a mobile smart phone i installed a SIP client such as CSipSimple,registered on Asterisk server , on Asterisk console the client SIP has been registered on with the Server,between these two mobile phone i can make and receive call,the comunication was perfect on local Image Image .
Now i want to configure extension for ip phone such as Deskphone,for exemple 2 extensions,my question is :
For IP Phone such as Deskphone is it the same procedure to make an ip phone registered to the Asterisk Server,for exemple if i set a SIP extension,does it going to work ?

Thanks,

Statistics : Posted by tella • on Fri Dec 12, 2014 1:28 pm • Replies 4 • Views 115
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Inbound call rejected on SIP trunk

December 12, 2014, 12:20 pm
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Thanks to all of you for your helpful comments. I will try to give you all the info that you need to help me solve the problem.
My SIP trunk is connected to a Digium G200 gateway, which connects to another PBX on ISDN trunks. When I make a call from my other PBX to an Asterik extension over the SIP trunk, the call gets rejected with below error. "2365" in that case is the extension I am calling from, and "Cloud" is the username of the SIP endpoint on the G200, and also part of the register string Cloud:secret@10.1.1.125.
So Asterisk confuses the extension number that I am calling from with the username of the SIP endpoint of the gateway.
Setting type to "user" in peer settings, returns the same error.
I also have H.323 trunks between the two PBXs and everything works there.
I noticed, when I disable sending callerID on the ISDN trunk to the gateway, my call completes.

David55, I would like to provide you the first INVITE, any 401 and any second INVITE. Where do I get that information exactly?
(Relatively new to Asterisk)

Statistics : Posted by avayax • on Fri Dec 12, 2014 2:20 pm • Replies 10 • Views 309
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Inbound call rejected on SIP trunk

December 12, 2014, 12:20 pm
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Thinking more about this, I suspect you are seeing why using type=friend is usually a bad thing. Your incoming calls are probably showing the same callerID as a local extension, and therefore matching the entry for a local extension in the type=user role of type=friend.

Unfortunately, you have generally insisted on providing too little information to see what was going on (e.g. seeing that the trunk device name wasn't 2385 would have been a good clue), so one could not confirm this from the information you have provided. Like blind use of insecure=very, blind use of type=friend is so common that one often tends to ignore it's questionable use.

Statistics : Posted by avayax • on Fri Dec 12, 2014 2:20 pm • Replies 10 • Views 309
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Inbound call rejected on SIP trunk

December 12, 2014, 12:20 pm
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[2014-12-12 18:08:26] WARNING[2443][C-000002ac]: chan_sip.c:16491 check_auth: username mismatch, have <2365>, digest has <Cloud>

On the trunk setting in your freepbx replace Cloud name for 2365.. This would fix the issue

Statistics : Posted by avayax • on Fri Dec 12, 2014 2:20 pm • Replies 10 • Views 309
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Inbound call rejected on SIP trunk

December 12, 2014, 12:20 pm
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The bit in square brackets is important. It should match one of two identifiers given and I want to know which.

With insecure=invite, Asterisk should not send 401, so there should be no authentication data. Maybe the peer is sending authentication data without being properly challenged for it.

In any case, you need to provide the first INVITE, any 401 and any second INVITE.

Statistics : Posted by avayax • on Fri Dec 12, 2014 2:20 pm • Replies 10 • Views 309
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Inbound call rejected on SIP trunk

December 12, 2014, 12:20 pm
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Thank you.
I have changed the peer details. Still no inbound calls with same error message. Any other ideas?
host=10.1.1.125
type=friend
context=from-trunk
insecure=invite
defaultuser=Cloud
remotesecret=.................

[2014-12-12 18:08:26] WARNING[2443][C-000002ac]: chan_sip.c:16491 check_auth: username mismatch, have <2365>, digest has <Cloud>
[2014-12-12 18:08:26] NOTICE[2443][C-000002ac]: chan_sip.c:25611 handle_request_invite: Failed to authenticate device "▒Jonathan Klein " <sip:2365@10.1.1.125>;tag=as3b386d47

Statistics : Posted by avayax • on Fri Dec 12, 2014 2:20 pm • Replies 10 • Views 309
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Inbound call rejected on SIP trunk

December 12, 2014, 12:20 pm
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The bit in square brackets is important.

However, the normal reason for this is neither using insecure=invite, nor using remotesecret instead of secret. insecure=very was deprecated and may well have been discontinued.

Statistics : Posted by avayax • on Fri Dec 12, 2014 2:20 pm • Replies 10 • Views 309
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Inbound call rejected on SIP trunk

December 12, 2014, 12:20 pm
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I have Asterisk 11.9.0 and Free PBX 12.0.18 on CentOS.
I have a SIP trunk connected to a Digium G200 Gateway. Inbound calls fail and I get following error:

WARNING[2443][C-00000275]: chan_sip.c:16491 check_auth: username mismatch, have <2365>, digest has <Cloud>
[2014-12-12 15:00:52] NOTICE[2443][C-00000275]: chan_sip.c:25611 handle_request_invite: Failed to authenticate device "▒Jonathan Klein " <sip:2365@10.1.1.125>;tag=as4a614331

These are my peer details in outgoing settings in free PBX:
host=10.1.1.125
type=friend
context=from-trunk
insecure=very
defaultuser=Cloud
secret=.................

Inbound settings are empty.
Strangely, I had had it working with these settings for a while. Didn't know what changes I made apart from upgrading Free PBX.
Can you help?

Statistics : Posted by avayax • on Fri Dec 12, 2014 2:20 pm • Replies 10 • Views 309
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Problem with Audio

December 14, 2014, 9:32 am
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Here is the Debug
Code: ---
Scheduling destruction of SIP dialog 'K5zIElTdR7dddcjn2TcVvuEc9YfbXMlI' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:64.170.xxx.xxx:29377 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.170.xxx.xxx:28377;rport=28377;received=64.170.xxx.xxx;branch=z9hG4bK6fe1bc74
Call-ID: 7159e80355a9149e298319dd2f9721ba@64.170.xxx.xxx
From: "asterisk" <sip:asterisk@64.170.xxx.xxx>;tag=as6b90e63e
To: <sip:417@64.170.xxx.xxx;ob>;tag=z9hG4bK6fe1bc74
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: Digium D50 1_4_2_0_63880
Content-Type: application/sdp
Content-Length: 425

v=0
o=- 113959953 113959953 IN IP4 64.170.xxx.xxx
s=digphn
c=IN IP4 64.170.98.168
t=0 0
m=audio 4000 RTP/AVP 0 8 9 111 18 58 118 58 96
a=rtcp:4001 IN IP4 64.170.xxx.xxx
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:58 L16/16000
a=rtpmap:118 L16/8000
a=rtpmap:58 L16-256/16000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
<------------->
--- (13 headers 18 lines) ---
Really destroying SIP dialog '7159e80355a9149e298319dd2f9721ba@64.170.98.15:28377' Method: OPTIONS

<--- SIP read from UDP:64.170.xxx.xxx --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.170.xxx.xxx;rport=xxxx;received=64.170.xxx.xxx;branch=z9hG4bK74dbeaa3
Call-ID: 5251d21704fdeece0822bcd5141db0e7@64.170.xxx.xxx:
From: "asterisk" <sip:asterisk@64.170.xxx.xxx>;tag=as2d8fb84d
To: <sip:417@64.170.xxx.xxx;ob>;tag=z9hG4bK74dbeaa3
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '5251d21704fdeece0822bcd5141db0e7@64.170.xxx.xxx' Method: NOTIFY
Reliably Transmitting (NAT) to 162.243.35.55:44121:
OPTIONS sip:418@10.0.1.3:44121;rinstance=11C2CFFD SIP/2.0
Via: SIP/2.0/UDP 64.170.xxx.xxx;branch=z9hG4bK00d086c4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@64.170.xxx.xxx>;tag=as62d6e907
To: <sip:418@10.0.1.3:44121;rinstance=11C2CFFD>
Contact: <sip:asterisk@64.170.xxx.xxx>
Call-ID: 36b68090474f523768743ac86fac96cf@64.170.98.15:28377
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.0
Date: Sun, 14 Dec 2014 20:13:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---

<--- SIP read from UDP:162.243.35.55:44121 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.170.98.15:28377;branch=z9hG4bK00d086c4;rport=28377;received=64.170.98.15
Contact: <sip:418@10.0.1.3:44121;rinstance=11C2CFFD>
From: "asterisk" <sip:asterisk@64.170.xxx.xxx>;tag=as62d6e907
Call-ID: 36b68090474f523768743ac86fac96cf@64.170.xxx.xxx
CSeq: 102 OPTIONS
To: <sip:418@10.0.1.3:44121;rinstance=11C2CFFD>
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '36b68090474f523768743ac86fac96cf@64.170.98.15:28377' Method: OPTIONS
Reliably Transmitting (NAT) to 71.198.111.249:29377:
OPTIONS sip:419@10.0.0.24:29377 SIP/2.0
Via: SIP/2.0/UDP 64.170.xxx.xxx;branch=z9hG4bK765708b6;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@64.170.xxx.xxx>;tag=as1b0b4477
To: <sip:419@10.0.0.24:29377>
Contact: <sip:asterisk@64.170.xxx.xxx>
Call-ID: 747a88d6679f14784fbce1f242fcfd05@64.170.98.15:28377
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.0
Date: Sun, 14 Dec 2014 20:13:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---

<--- SIP read from UDP:71.198.111.249:29377 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.170.xxx.xxx;branch=z9hG4bK765708b6;rport
From: "asterisk" <sip:asterisk@64.170.xxx.xxx>;tag=as1b0b4477
To: <sip:419@10.0.0.24:29377>;tag=3209036821
Call-ID: 747a88d6679f14784fbce1f242fcfd05@64.170.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-W52P 25.73.0.20
Content-Length: 0


Statistics : Posted by aristech • on Sun Dec 14, 2014 11:32 am • Replies 2 • Views 50
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Problem with Audio

December 14, 2014, 9:32 am
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Insufficient information. For a start there are many different ways in which transfers can be done.

Statistics : Posted by aristech • on Sun Dec 14, 2014 11:32 am • Replies 2 • Views 50
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Problem with Audio

December 14, 2014, 9:32 am
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What is happening is when a call comes in on the main line and the caller dials the extension the caller can hear nothing but the call receiver can hear the caller and speak. Also If I dial direct to the extension audio works perfect. It only happens when the call is transferred

Statistics : Posted by aristech • on Sun Dec 14, 2014 11:32 am • Replies 2 • Views 50
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Hardware requierements for 8 x E1 Calls (240)

December 15, 2014, 3:29 pm
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Hello guys,

How to know my hardware requierements (cpu and ram) when i'm going to use 2 cards (3 x E1 each one) then i will have 240 concurrent calls.

Thanks a lot.

Statistics : Posted by swimmercol • on Mon Dec 15, 2014 5:29 pm • Replies 0 • Views 17
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