[asterisk-bugs] [Asterisk 0016351]: Asterisk 1.6.1 won't "answer" the phone when using a callcentric sip trunk
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Feb 12 13:43:23 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16351
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Reported By: cattelan
Assigned To:
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Project: Asterisk
Issue ID: 16351
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.6.1.10
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-11-30 11:16 CST
Last Modified: 2010-02-12 13:43 CST
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Summary: Asterisk 1.6.1 won't "answer" the phone when using a
callcentric sip trunk
Description:
I have been trying for a while to upgrade my system from a 1.4 based
asterisk to a 1.6 asterisk, but have never been able to get it to function
correctly with my sip provider (calcentric)
I finally sat down and did some comparisons of the sip debug info and I
think I have found an issue.
callcentric wants to use port 5080 vs the standard 5060 port.
It appears that 1.4 is correctly sending the sip answer message via port
5080 but 1.6 does not -- it is trying to use port 5060.
Here is the relevant sip debug messages.
asterisk 1.4.22
-- Executing [s at ivr-2:7] Answer("SIP/66.193.176.35-1ecd0160", "") in
new stack
Audio is at 10.0.0.13 port 12644
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pbx-barn-old*CLI>
<--- Reliably Transmitting (no NAT) to 204.11.192.36:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
204.11.192.36:5080;branch=z9hG4bK-26fd0d9fbdc6e35702ab0c2b568e095a;received=204.11.192.36
From: "Russell Cattelan"
<sip:16128053144 at 66.193.176.35>;tag=3468510726-41122
To: <sip:16514820379 at ss.callcentric.com>;tag=as700a9f61
Call-ID: 13715408-3468510726-41097 at msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:17772080931 at 10.0.0.13>
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 21108 21108 IN IP4 10.0.0.13
s=session
c=IN IP4 10.0.0.13
t=0 0
m=audio 12644 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Asterisk 1.6.1
---
-- SIP/203-a8007498 answered SIP/66.193.176.35-c0008898
Audio is at 10.0.0.31 port 15980
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
barn-pbx*CLI>
<--- Reliably Transmitting (no NAT) to 204.11.192.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
204.11.192.22:5060;branch=z9hG4bK-79eb04ffe8db79dc1c9a1bec67872e62;received=204.11.192.22
From: "Russell Cattelan"
<sip:16128053144 at 66.193.176.35>;tag=3468510938-450305
To: <sip:16514820379 at ss.callcentric.com>;tag=as524972e9
Call-ID: 13715775-3468510938-450277 at msw1.telengy.net
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: <sip:17772080931 at 10.0.0.31>
Content-Type: application/sdp
Content-Length: 279
v=0
o=root 233148858 233148858 IN IP4 10.0.0.31
s=Asterisk PBX 1.6.1.9
c=IN IP4 10.0.0.31
t=0 0
m=audio 15980 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
======================================================================
----------------------------------------------------------------------
(0118043) adahlquist (reporter) - 2010-02-12 13:43
https://issues.asterisk.org/view.php?id=16351#c118043
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I tried that and it didnt make a difference. I also have port=5063 in peer
details, and in the fromdomain i have host:5063. Also have port=5063 in
sip general. Here is my registration from a 1.4 asterisk box:
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: REGISTER sip:registrar.bandtel.com SIP/2.0
Message Header
Via: SIP/2.0/UDP 24.69.128.115:5061;branch=z9hG4bK3be99106;rport
Transport: UDP
Sent-by Address: 24.69.128.115
Sent-by port: 5061
Branch: z9hG4bK3be99106
RPort: rport
From: <sip:201077000007 at registrar.bandtel.com>;tag=as18fc8ea9
To: <sip:201077000007 at registrar.bandtel.com>
Call-ID: 5fd472234efe1da5762da6ef2842c4ea at 192.168.170.213
CSeq: 784 REGISTER
Sequence Number: 784
Method: REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="201077000007", realm="10.1.2.1",
algorithm=MD5, uri="sip:registrar.bandtel.com",
nonce="4b74955a810dd1992795b5b175596020aae65ca0",
response="af404f1d9aa3f34704278b3786ef91e2"
Authentication Scheme: Digest
Username: "201077000007"
Realm: "10.1.2.1"
Algorithm: MD5
Authentication URI: "sip:registrar.bandtel.com"
Nonce Value: "4b74955a810dd1992795b5b175596020aae65ca0"
Digest Authentication Response:
"af404f1d9aa3f34704278b3786ef91e2"
Expires: 120
Contact: <sip:201077000007 at 24.69.128.115:5061>
Event: registration
Content-Length: 0
And here is one from my 1.6 box which has the same config (VM image was
replicated and asterisk updated)
User Datagram Protocol, Src Port: 5063 (5063), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: REGISTER sip:registrar.bandtel.com SIP/2.0
Message Header
Via: SIP/2.0/UDP 24.69.128.115:5060;branch=z9hG4bK14588a4e;rport
Transport: UDP
Sent-by Address: 24.69.128.115
Sent-by port: 5060
Branch: z9hG4bK14588a4e
RPort: rport
Max-Forwards: 70
From: <sip:201077000006 at registrar.bandtel.com>;tag=as79b7fda4
To: <sip:201077000006 at registrar.bandtel.com>
Call-ID: 5f8d846018fe5bc579a77c025e60f8c7 at 192.168.170.171
CSeq: 173 REGISTER
Sequence Number: 173
Method: REGISTER
User-Agent: Asterisk PBX 1.6.2.2
Authorization: Digest username="201077000006", realm="10.1.12.1",
algorithm=MD5, uri="sip:registrar.bandtel.com",
nonce="4b75ac2cc400324fc2ab5f80e7d5ad072fbdfd33",
response="c5cd8f5bc3a5c6b134272b6b5987e328"
Authentication Scheme: Digest
Username: "201077000006"
Realm: "10.1.12.1"
Algorithm: MD5
Authentication URI: "sip:registrar.bandtel.com"
Nonce Value: "4b75ac2cc400324fc2ab5f80e7d5ad072fbdfd33"
Digest Authentication Response:
"c5cd8f5bc3a5c6b134272b6b5987e328"
Expires: 120
Contact: <sip:201077000006 at 24.69.128.115>
Content-Length: 0
You can see that my source port is correct in both cases.
Please let me know as soon as you have an idea if this is a bug in
asterisk so i can set expectations as to where the roadblock is on my end.
Thanks a lot for your help and prompt replies on this issue.
Issue History
Date Modified Username Field Change
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2010-02-12 13:43 adahlquist Note Added: 0118043
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