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Summary:ASTERISK-11969: Asterisk crashes when a call comes in from a Mediatrix 2102
Reporter:Daniel Lynes (dlynes)Labels:
Date Opened:2008-05-04 17:01:20Date Closed:2011-06-07 14:00:59
Priority:CriticalRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Call is made from a Mediatrix 2102, Asterisk dumps core on connect.
All registrations and authentications are fine.

****** STEPS TO REPRODUCE ******

Send a call to the asterisk box from the mediatrix device.

****** ADDITIONAL INFORMATION ******

SIP debug log:

[root@new-host asterisk]# /usr/sbin/asterisk -r
Asterisk 1.4.19.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.19.1 currently running on new-host (pid = 18303)
Verbosity is at least 3
new-host*CLI> sip debug
SIP Debugging enabled
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
new-host*CLI>
<--- SIP read from 10.1.99.107:5060 --->
REGISTER sip:10.1.99.105:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bKea65d0c03
Content-Length: 0
To: phone3 <sip:phone3@10.1.99.105:5060>
From: phone3 <sip:phone3@10.1.99.105:5060>;tag=1abd663bd78dc6b
Call-ID: 523e2787d668dbcc2cd5f12e93526adc@10.1.99.105
CSeq: 1034426160 REGISTER
Contact: phone3 <sip:phone3@10.1.99.107:5060>
Authorization:Digest response="e470bf405e3ee5b714e4ae05de426728",username="phone3",realm="asterisk",nonce="3a745798",algorithm=MD5,uri="sip:10.1.99.105:5060"
User-Agent: MxSipApp/5.0.14.83 MxSF/v3.2.1.1


<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.99.107 : 5060 (no NAT)

<--- Transmitting (no NAT) to 10.1.99.107:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bKea65d0c03;received=10.1.99.107
From: phone3 <sip:phone3@10.1.99.105:5060>;tag=1abd663bd78dc6b
To: phone3 <sip:phone3@10.1.99.105:5060>
Call-ID: 523e2787d668dbcc2cd5f12e93526adc@10.1.99.105
CSeq: 1034426160 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:phone3@10.1.99.105>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 10.1.99.107:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bKea65d0c03;received=10.1.99.107
From: phone3 <sip:phone3@10.1.99.105:5060>;tag=1abd663bd78dc6b
To: phone3 <sip:phone3@10.1.99.105:5060>;tag=as60bbda63
Call-ID: 523e2787d668dbcc2cd5f12e93526adc@10.1.99.105
CSeq: 1034426160 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d7b45e1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '523e2787d668dbcc2cd5f12e93526adc@10.1.99.105' in 32000 ms (Method: REGISTER)
new-host*CLI>
<--- SIP read from 10.1.99.107:5060 --->
REGISTER sip:10.1.99.105:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bKd1315889c
Content-Length: 0
To: phone4 <sip:phone4@10.1.99.105:5060>
From: phone4 <sip:phone4@10.1.99.105:5060>;tag=8c56483330037ad
Call-ID: 92556dffa9f806ab61e5eb5c082129b9@10.1.99.105
CSeq: 1799072873 REGISTER
Contact: phone4 <sip:phone4@10.1.99.107:5060>
Authorization:Digest response="167f7caca093bedcdce2a076c66dca3a",username="phone4",realm="asterisk",nonce="561aae22",algorithm=MD5,uri="sip:10.1.99.105:5060"
User-Agent: MxSipApp/5.0.14.83 MxSF/v3.2.1.1


<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.99.107 : 5060 (no NAT)

<--- Transmitting (no NAT) to 10.1.99.107:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bKd1315889c;received=10.1.99.107
From: phone4 <sip:phone4@10.1.99.105:5060>;tag=8c56483330037ad
To: phone4 <sip:phone4@10.1.99.105:5060>
Call-ID: 92556dffa9f806ab61e5eb5c082129b9@10.1.99.105
CSeq: 1799072873 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:phone4@10.1.99.105>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 10.1.99.107:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bKd1315889c;received=10.1.99.107
From: phone4 <sip:phone4@10.1.99.105:5060>;tag=8c56483330037ad
To: phone4 <sip:phone4@10.1.99.105:5060>;tag=as75ca92af
Call-ID: 92556dffa9f806ab61e5eb5c082129b9@10.1.99.105
CSeq: 1799072873 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6bf4d472"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '92556dffa9f806ab61e5eb5c082129b9@10.1.99.105' in 32000 ms (Method: REGISTER)
new-host*CLI>
<--- SIP read from 10.1.99.107:5060 --->
REGISTER sip:10.1.99.105:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bKca424b3e8
Content-Length: 0
To: phone3 <sip:phone3@10.1.99.105:5060>
From: phone3 <sip:phone3@10.1.99.105:5060>;tag=1abd663bd78dc6b
Call-ID: 523e2787d668dbcc2cd5f12e93526adc@10.1.99.105
CSeq: 1034426161 REGISTER
Contact: phone3 <sip:phone3@10.1.99.107:5060>
Authorization:Digest response="4ff210fcec3ea1aa72c203a4ed1092f9",username="phone3",realm="asterisk",nonce="3d7b45e1",algorithm=MD5,uri="sip:10.1.99.105:5060"
User-Agent: MxSipApp/5.0.14.83 MxSF/v3.2.1.1


<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.99.107 : 5060 (no NAT)

<--- Transmitting (no NAT) to 10.1.99.107:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bKca424b3e8;received=10.1.99.107
From: phone3 <sip:phone3@10.1.99.105:5060>;tag=1abd663bd78dc6b
To: phone3 <sip:phone3@10.1.99.105:5060>
Call-ID: 523e2787d668dbcc2cd5f12e93526adc@10.1.99.105
CSeq: 1034426161 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:phone3@10.1.99.105>
Content-Length: 0


<------------>
new-host*CLI>
<--- Transmitting (no NAT) to 10.1.99.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bKca424b3e8;received=10.1.99.107
From: phone3 <sip:phone3@10.1.99.105:5060>;tag=1abd663bd78dc6b
To: phone3 <sip:phone3@10.1.99.105:5060>;tag=as60bbda63
Call-ID: 523e2787d668dbcc2cd5f12e93526adc@10.1.99.105
CSeq: 1034426161 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:phone3@10.1.99.107:5060>;expires=120
Date: Mon, 05 May 2008 03:06:19 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '523e2787d668dbcc2cd5f12e93526adc@10.1.99.105' in 32000 ms (Method: REGISTER)
new-host*CLI>
<--- SIP read from 10.1.99.107:5060 --->
REGISTER sip:10.1.99.105:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bK12a06723a
Content-Length: 0
To: phone4 <sip:phone4@10.1.99.105:5060>
From: phone4 <sip:phone4@10.1.99.105:5060>;tag=8c56483330037ad
Call-ID: 92556dffa9f806ab61e5eb5c082129b9@10.1.99.105
CSeq: 1799072874 REGISTER
Contact: phone4 <sip:phone4@10.1.99.107:5060>
Authorization:Digest response="b6d93d57dcc402177f7461a96bca5fec",username="phone4",realm="asterisk",nonce="6bf4d472",algorithm=MD5,uri="sip:10.1.99.105:5060"
User-Agent: MxSipApp/5.0.14.83 MxSF/v3.2.1.1


<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.99.107 : 5060 (no NAT)

<--- Transmitting (no NAT) to 10.1.99.107:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bK12a06723a;received=10.1.99.107
From: phone4 <sip:phone4@10.1.99.105:5060>;tag=8c56483330037ad
To: phone4 <sip:phone4@10.1.99.105:5060>
Call-ID: 92556dffa9f806ab61e5eb5c082129b9@10.1.99.105
CSeq: 1799072874 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:phone4@10.1.99.105>
Content-Length: 0


<------------>
new-host*CLI>
<--- Transmitting (no NAT) to 10.1.99.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.99.107:5060;branch=z9hG4bK12a06723a;received=10.1.99.107
From: phone4 <sip:phone4@10.1.99.105:5060>;tag=8c56483330037ad
To: phone4 <sip:phone4@10.1.99.105:5060>;tag=as75ca92af
Call-ID: 92556dffa9f806ab61e5eb5c082129b9@10.1.99.105
CSeq: 1799072874 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:phone4@10.1.99.107:5060>;expires=120
Date: Mon, 05 May 2008 03:06:19 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '92556dffa9f806ab61e5eb5c082129b9@10.1.99.105' in 32000 ms (Method: REGISTER)
new-host*CLI>
<--- SIP read from 10.1.99.149:5060 --->
INVITE sip:6044840153@10.1.99.105:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.99.149:5060;branch=z9hG4bKb4953d986
Content-Length: 241
To: sip:6044840153@10.1.99.105:5060
From: phone1 <sip:phone1@10.1.99.105:5060>;tag=7752027f4333ec4
Call-ID: 8364dc8753f96554c55ee6375cb905d1@10.1.99.105
CSeq: 1535308721 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: phone1 <sip:phone1@10.1.99.149:5060>
Supported: replaces
User-Agent: MxSipApp/5.0.15.92 MxSF/v3.2.1.1

v=0
o=MxSIP 765926109111005011 1044520923467861791 IN IP4 10.1.99.149
s=-
c=IN IP4 10.1.99.149
t=0 0
a=sendrecv
m=audio 5006 RTP/AVP 0 18 4 8 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000

<------------->
--- (15 headers 11 lines) ---
Sending to 10.1.99.149 : 5060 (no NAT)
Using INVITE request as basis request - 8364dc8753f96554c55ee6375cb905d1@10.1.99.105

<--- Reliably Transmitting (no NAT) to 10.1.99.149:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.99.149:5060;branch=z9hG4bKb4953d986;received=10.1.99.149
From: phone1 <sip:phone1@10.1.99.105:5060>;tag=7752027f4333ec4
To: sip:6044840153@10.1.99.105:5060;tag=as71f5494d
Call-ID: 8364dc8753f96554c55ee6375cb905d1@10.1.99.105
CSeq: 1535308721 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="611dfea1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8364dc8753f96554c55ee6375cb905d1@10.1.99.105' in 32000 ms (Method: INVITE)
Found user 'phone1'
new-host*CLI>
<--- SIP read from 10.1.99.149:5060 --->
ACK sip:6044840153@10.1.99.105:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.99.149:5060;branch=z9hG4bKb4953d986
Content-Length: 0
To: sip:6044840153@10.1.99.105:5060;tag=as71f5494d
From: phone1 <sip:phone1@10.1.99.105:5060>;tag=7752027f4333ec4
Call-ID: 8364dc8753f96554c55ee6375cb905d1@10.1.99.105
CSeq: 1535308721 ACK
User-Agent: MxSipApp/5.0.15.92 MxSF/v3.2.1.1


<------------->
--- (8 headers 0 lines) ---
new-host*CLI>
Disconnected from Asterisk server
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
[root@new-host asterisk]# mpg123: no process killed

Comments:By: Clod Patry (junky) 2008-05-04 19:35:42

Can you attach all the infos required as specified in doc/backtrace.txt?
By: Joshua C. Colp (jcolp) 2008-05-05 08:40:04

When creating a crash issue you always need to attach a backtrace, please follow the instructions in backtrace.txt of the doc directory.
By: Daniel Lynes (dlynes) 2008-05-06 14:13:29

I have determined what the issue was.

I was trying to strip down the memory footprint on this particular box, and in so doing, I did a noload on a module (res_features.so) that was apparently needed by chan_sip.so to be able to use the dial application.  However, the warning for that was not listed on the screen in core set verbose 100, nor was it listed in sip debug ip logging to the screen.

I only found the error/warning after using 'asterisk -vvvvvvvvvvvvvvvvvvvvvg'

What should happen, instead is that it should at least fail with an error to the screen in normal running mode, if it's going to exit with code 127 like it does.
By: Joshua C. Colp (jcolp) 2008-05-14 13:30:29

Closed as this was a configuration issue with modules. While menuselect will not allow you to build a module without the dependencies the module loader in Asterisk has no knowledge of modules that depend on each other when it comes to loading them.