Why the "404 Not found" from SIP INVITE generated from "channel originate"?
The phone seems to register OK. I have "senrpid = yes" in sip.conf
as was suggested by another user.
I'm guessing the problem is with the INVITE generated by the "channel originate" call.
1. I'm not sure what's incorrect, if anything, with the INVITE generated.
2. If that is true, I don't know how to configure Asterisk to generate the correct INVITE.
The general difference between these originate examples which fail with the "404 Not Found" response and the originate examples which I've gotten to work successfully is that the failed cases have user phones which have a more complicated registration process: chalange - authentication and they also use a different registration port "5260" instead of the default "5060".
I would certainly appreciate any suggestions / ideas - I'm stumped?
From CLI output:
channel originate SIP/62021@192.168.90.50:5260 extension 51207
== Using SIP RTP CoS mark 5
Audio is at 10648
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.90.50:5260:
INVITE sip:62021@192.168.90.50:5260 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.224:5060;branch=z9hG4bK1c0d0bf1
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5d3410a7
To: <sip:62021@192.168.90.50:5260>
Contact: <sip:anonymous@192.168.15.224:5060>
Call-ID: 2ab335f15deec5525a50158f7e2df117@192.168.15.224:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.1
Date: Tue, 09 Dec 2014 21:27:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 513210126 513210126 IN IP4 192.168.15.224
s=Asterisk PBX 11.12.1
c=IN IP4 192.168.15.224
t=0 0
m=audio 10648 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
*CLI>
<--- SIP read from UDP:192.168.90.50:5260 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.15.224:5060;branch=z9hG4bK1c0d0bf1;received=192.168.15.224
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5d3410a7
To: <sip:62021@192.168.90.50:5260>;tag=00955CA4-2A7B-145D-AA6D-325AA8C0AA77-615809
Call-ID: 2ab335f15deec5525a50158f7e2df117@192.168.15.224:5060
CSeq: 102 INVITE
Content-Length: 0
From sip.conf:
; sip.conf 04dec2014
;
[general]
registertimeout => 3600
defaultexpiry => 3600
allowexternaldomains = yes
sendrpid = yes
; rpid_update = no ; In certain cases, the only method by which a connected line chg ...
;
; ICM
;
register=>62021:password@192.168.90.50:5260/62021
; OXE
;
register=>51206:password@152.148.200.236:5060/51206
register=>51207:password@152.148.200.236:5060/51207
From CLI output (earlier) - The Registration:
[Dec 9 16:24:32] NOTICE[32535]: chan_sip.c:14991 sip_reregister: -- Re-registration for 62021@192.168.90.50
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.90.50:5260:
REGISTER sip:192.168.90.50:5260 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.224:5060;branch=z9hG4bK4c99839e
Max-Forwards: 70
From: <sip:62021@192.168.90.50>;tag=as5767814a
To: <sip:62021@192.168.90.50>
Call-ID: 39cb26c54a84669a2c61731e26e499d7@192.168.15.224
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 11.12.1
Authorization: Digest username="62021", realm="switch", algorithm=MD5, uri="sip:192.168.90.50:5260", nonce="008475F6-6879-1487-AA6D-325AA8C0AA77", response="f222e74872d8a0af2b81c3011f7ed353", opaque="737769746368", qop=auth, cnonce="015f1404", nc=00000002
Expires: 3600
Contact: <sip:62021@192.168.15.224:5060>
Content-Length: 0
---
<--- SIP read from UDP:192.168.90.50:5260 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.224:5060;branch=z9hG4bK4c99839e;received=192.168.15.224
From: <sip:62021@192.168.90.50>;tag=as5767814a
To: <sip:62021@192.168.90.50>;tag=00955CA4-2A7B-145D-AA6D-325AA8C0AA77-615767
Call-ID: 39cb26c54a84669a2c61731e26e499d7@192.168.15.224
CSeq: 104 REGISTER
Server: SIP Server, 2.0.044.01
Expires: 1800
Contact: <sip:62021@192.168.15.224:5060>;expires=1800
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
[Dec 9 16:24:32] NOTICE[32535]: chan_sip.c:23504 handle_response_register: Outbound Registration: Expiry for 192.168.90.50 is 1800 sec (Scheduling reregistration in 1785 s)
Really destroying SIP dialog '39cb26c54a84669a2c61731e26e499d7@192.168.15.224' Method: REGISTER
-UUU:----F1 originate_no_ip_in_from.txt
Statistics : Posted by crs • on Tue Dec 09, 2014 4:34 pm • Replies 5 • Views 139
