Question

sipdroid + asterisk

Asked by: csupp_1

When connecting to an asterisk box using sipdroid on an android based phone, you can make calls but not receive. The phone gets UNREACHABLE after just seconds.

If you use pbxes.org then you are able to make and receive calls without problem.

What's the difference between the SIP protocol on pbxes.org and the one you get on an asterisk box and/or how to configure asterisk to be compatible with sipdroid.

[2009-09-21 06:12:39] NOTICE[2140] chan_sip.c: Peer '180' is now UNREACHABLE!  Last qualify: 253

1:

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Asked On: 2009-09-21 at 02:48:27ID24747808
Tags: voip sip asterisk sipdroid android
Topics: Voice Over IP
,; Asterisk Open Source Telephony
,; IP PBX Systems
,; Android
Participating Experts: 1
Points: 0
Comments: 6

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Answers

by: denisdsr20Posted on 2009-09-21 at 03:20:16ID: 25381492

Hi, could you give us a trace of the sip messages received on asterisk ?

At asterisk CLI prompt enter : "sip debug on" then issue a call from sipdroid and when the call is abnormally cleared "sip debug off"

Insert the SIP messages you got at asterisk CLI in your question it will be easier to give you an accurate answer.

Regards

Denis DIDIER
SR20 Service / FRANCE

by: csupp_1Posted on 2009-09-21 at 04:28:40ID: 25381790

Hi, thanks for the reply, here are the requested details:

- Extension 198: android with sipdroid connected to asterisk box
- Extension 199: test ipphone

1. Registering the phone:
pbx1-uk*CLI> [2009-09-21 12:18:01] NOTICE[2140] chan_sip.c: Peer '198' is now UNREACHABLE!  Last qualify: 0
    -- Registered SIP '198' at 85.14.193.197 port 5060 expires 3600
    -- Saved useragent "Sipdroid/1.0.7 beta" for peer 198
 
2. Making a call from ext 199 to ext 198:
    -- Executing [198@from-internal:1] Macro("SIP/199-08406d40", "exten-vm|novm|198") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/199-08406d40", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/199-08406d40", "AMPUSER=199") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/199-08406d40", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/199-08406d40", "1|Set|REALCALLERIDNUM=199") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/199-08406d40", "AMPUSER=199") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/199-08406d40", "AMPUSERCIDNAME=TEST") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/199-08406d40", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/199-08406d40", "AMPUSERCID=199") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/199-08406d40", "CALLERID(all)="TEST" <199>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/199-08406d40", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/199-08406d40", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/199-08406d40", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/199-08406d40", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/199-08406d40", "Using CallerID "TEST" <199>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/199-08406d40", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/199-08406d40", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/199-08406d40", "EXTTOCALL=198") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/199-08406d40", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/199-08406d40", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/199-08406d40", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/199-08406d40", "record-enable|198|IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/199-08406d40", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/199-08406d40", "recordingcheck|20090921-121836|1253531916.50") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20090921-121836|1253531916.50: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/199-08406d40", "") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/199-08406d40", "dial||tr|198") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/199-08406d40", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/199-08406d40", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_additional.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: Caller ID name is 'TEST' number is '199'
  dialparties.agi: USE_CONFIRMATION:  'FALSE'
  dialparties.agi: RINGGROUP_INDEX:   ''
  dialparties.agi: Methodology of ring is  'none'
    --  dialparties.agi: Added extension 198 to extension map
    --  dialparties.agi: Extension 198 cf is disabled
    --  dialparties.agi: Extension 198 do not disturb is disabled
       >  dialparties.agi: extnum 198 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
  dialparties.agi: ExtensionState: 4
    --  dialparties.agi: dbset CALLTRACE/198 to 199
    --  dialparties.agi: Filtered ARG3: 198
  == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/199-08406d40", "SIP/198||tr") in new stack
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dial:8] Set("SIP/199-08406d40", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-dial:9] GosubIf("SIP/199-08406d40", "0?CHANUNAVAIL|1") in new stack
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/199-08406d40", "0?exit|return") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/199-08406d40", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/199-08406d40", "0?docfu|1") in new stack
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/199-08406d40", "0?docfb|1") in new stack
    -- Executing [s@macro-exten-vm:14] Set("SIP/199-08406d40", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/199-08406d40", "Voicemail is novm") in new stack
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/199-08406d40", "1?s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/199-08406d40", "IVR_RETVM:  IVR_CONTEXT: ") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/199-08406d40", "0?exit|1") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/199-08406d40", "congestion") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:4] Congestion("SIP/199-08406d40", "10") in new stack
  == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 4) exited non-zero on 'SIP/199-08406d40' in macro 'exten-vm'
  == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 4) exited non-zero on 'SIP/199-08406d40'
pbx1-uk*CLI> [2009-09-21 12:18:36] VERBOSE[24762] logger.c:     -- Executing [s@macro-dial:8] Set("SIP/199-08406d40", "DIALSTATUS=CHANUNAVAIL") in new stack
[2009-09-21 12:18:36] VERBOSE[24762] logger.c:     -- Executing [s@macro-dial:9] GosubIf("SIP/199-08406d40", "0?CHANUNAVAIL|1") in new stack
[2009-09-21 12:18:36] VERBOSE[24762] logger.c:     -- Executing [s@macro-exten-vm:11] Set("SIP/199-08406d40", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
[2009-09-21 12:18:36] VERBOSE[24762] logger.c:     -- Executing [s@macro-exten-vm:14] Set("SIP/199-08406d40", "DIALSTATUS=CHANUNAVAIL") in new stack
[2009-09-21 12:18:36] VERBOSE[24762] logger.c:     -- Executing [s@macro-exten-vm:16] GotoIf("SIP/199-08406d40", "1?s-CHANUNAVAIL|1") in new stack
[2009-09-21 12:18:37] VERBOSE[24762] logger.c:     -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
[2009-09-21 12:18:37] VERBOSE[24762] logger.c:     -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/199-08406d40", "IVR_RETVM:  IVR_CONTEXT: ") in new stack
[2009-09-21 12:18:37] VERBOSE[24762] logger.c:     -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/199-08406d40", "0?exit|1") in new stack
[2009-09-21 12:18:37] VERBOSE[24762] logger.c:     -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/199-08406d40", "congestion") in new stack
[2009-09-21 12:18:37] VERBOSE[24762] logger.c:     -- Executing [s-CHANUNAVAIL@macro-exten-vm:4] Congestion("SIP/199-08406d40", "10") in new stack
[2009-09-21 12:18:37] VERBOSE[24762] logger.c:   == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 4) exited non-zero on 'SIP/199-08406d40' in macro 'exten-vm'
[2009-09-21 12:18:37] VERBOSE[24762] logger.c:   == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 4) exited non-zero on 'SIP/199-08406d40'

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by: denisdsr20Posted on 2009-09-21 at 05:30:43ID: 25382193

It seems you do not report the SIP messages perphas you forget "sip debug on" command ?

Denis DIDIER

by: csupp_1Posted on 2009-09-21 at 05:50:43ID: 25382321

Sorry,

I managed to get the debug data. I can see the problem is while registering. Check logs attached after just registering the phone to the asterisk box. It keeps retrying over and over again.

<--- SIP read from 85.14.192.197:5060 --->
REGISTER sip:pbx1-uk.cxl.net SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK51232
Max-Forwards: 70
To: <sip:198@pbx1-uk.cxl.net>
From: <sip:198@pbx1-uk.cxl.net>;tag=z9hG4bK32123045
Call-ID: 541529033227@127.0.0.1
CSeq: 1 REGISTER
Contact: <sip:198@127.0.0.1>
Expires: 3600
User-Agent: Sipdroid/1.0.7 beta
Content-Length: 0
 
 
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 85.14.192.197 : 5060 (NAT)
 
<--- Transmitting (NAT) to 85.14.192.197:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK51232;received=85.14.192.197;rport=5060
From: <sip:198@pbx1-uk.cxl.net>;tag=z9hG4bK32123045
To: <sip:198@pbx1-uk.cxl.net>
Call-ID: 541529033227@127.0.0.1
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:198@85.13.233.232>
Content-Length: 0
 
 
<------------>
 
<--- Transmitting (NAT) to 85.14.192.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK51232;received=85.14.192.197;rport=5060
From: <sip:198@pbx1-uk.cxl.net>;tag=z9hG4bK32123045
To: <sip:198@pbx1-uk.cxl.net>;tag=as4ecde385
Call-ID: 541529033227@127.0.0.1
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="447141bd"
Content-Length: 0
 
 
<------------>
Scheduling destruction of SIP dialog '541529033227@127.0.0.1' in 32000 ms (Method: REGISTER)
pbx1-uk*CLI> 
<--- SIP read from 85.14.192.197:5060 --->
REGISTER sip:pbx1-uk.cxl.net SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK51232
Max-Forwards: 70
To: <sip:198@pbx1-uk.cxl.net>
From: <sip:198@pbx1-uk.cxl.net>;tag=z9hG4bK32123045
Call-ID: 541529033227@127.0.0.1
CSeq: 2 REGISTER
Contact: <sip:198@127.0.0.1>
Expires: 3600
User-Agent: Sipdroid/1.0.7 beta
Authorization: Digest username="198", realm="asterisk", nonce="447141bd", uri="sip:pbx1-uk.cxl.net", algorithm=MD5, response="2b414045ca0b9f104ef54c6e99e8cf01"
Content-Length: 0
 
 
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 85.14.192.197 : 5060 (NAT)
 
<--- Transmitting (NAT) to 85.14.192.197:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK51232;received=85.14.192.197;rport=5060
From: <sip:198@pbx1-uk.cxl.net>;tag=z9hG4bK32123045
To: <sip:198@pbx1-uk.cxl.net>
Call-ID: 541529033227@127.0.0.1
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:198@85.13.233.232>
Content-Length: 0
 
 
<------------>
Reliably Transmitting (NAT) to 85.14.192.197:5060:
OPTIONS sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK78a387e8;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as2c57f7f8
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 06c732f8142bed77721ea755323809da@85.13.233.232
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Sep 2009 12:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
 
 
---
    -- Registered SIP '198' at 85.14.192.197 port 5060 expires 3600
    -- Saved useragent "Sipdroid/1.0.7 beta" for peer 198
pbx1-uk*CLI> 
<--- Transmitting (NAT) to 85.14.192.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK51232;received=85.14.192.197;rport=5060
From: <sip:198@pbx1-uk.cxl.net>;tag=z9hG4bK32123045
To: <sip:198@pbx1-uk.cxl.net>;tag=as4ecde385
Call-ID: 541529033227@127.0.0.1
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:198@127.0.0.1>;expires=3600
Date: Mon, 21 Sep 2009 12:46:36 GMT
Content-Length: 0
 
 
<------------>
Scheduling destruction of SIP dialog '541529033227@127.0.0.1' in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '634427706cda187d36d813a30a4eaf1c@85.13.233.232' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 85.14.192.197:5060:
NOTIFY sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK07931a4c;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as508ab0e0
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 634427706cda187d36d813a30a4eaf1c@85.13.233.232
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
 
Messages-Waiting: no
Message-Account: sip:*97@85.13.233.232
Voice-Message: 0/0 (0/0)
 
---
Retransmitting #1 (NAT) to 85.14.192.197:5060:
OPTIONS sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK78a387e8;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as2c57f7f8
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 06c732f8142bed77721ea755323809da@85.13.233.232
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Sep 2009 12:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
 
 
---
Retransmitting #1 (NAT) to 85.14.192.197:5060:
NOTIFY sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK07931a4c;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as508ab0e0
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 634427706cda187d36d813a30a4eaf1c@85.13.233.232
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
 
Messages-Waiting: no
Message-Account: sip:*97@85.13.233.232
Voice-Message: 0/0 (0/0)
 
---
Retransmitting #2 (NAT) to 85.14.192.197:5060:
OPTIONS sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK78a387e8;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as2c57f7f8
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 06c732f8142bed77721ea755323809da@85.13.233.232
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Sep 2009 12:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
 
 
---
Retransmitting #2 (NAT) to 85.14.192.197:5060:
NOTIFY sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK07931a4c;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as508ab0e0
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 634427706cda187d36d813a30a4eaf1c@85.13.233.232
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
 
Messages-Waiting: no
Message-Account: sip:*97@85.13.233.232
Voice-Message: 0/0 (0/0)
 
---
Retransmitting #3 (NAT) to 85.14.192.197:5060:
OPTIONS sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK78a387e8;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as2c57f7f8
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 06c732f8142bed77721ea755323809da@85.13.233.232
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Sep 2009 12:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
 
 
---
Retransmitting #4 (NAT) to 85.14.192.197:5060:
OPTIONS sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK78a387e8;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as2c57f7f8
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 06c732f8142bed77721ea755323809da@85.13.233.232
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Sep 2009 12:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
 
 
---
Really destroying SIP dialog '06c732f8142bed77721ea755323809da@85.13.233.232' Method: OPTIONS
Retransmitting #3 (NAT) to 85.14.192.197:5060:
NOTIFY sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK07931a4c;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as508ab0e0
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 634427706cda187d36d813a30a4eaf1c@85.13.233.232
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
 
Messages-Waiting: no
Message-Account: sip:*97@85.13.233.232
Voice-Message: 0/0 (0/0)
 
---
pbx1-uk*CLI> [2009-09-21 13:46:40] NOTICE[2140] chan_sip.c: Peer '198' is now UNREACHABLE!  Last qualify: 0
 
<--- SIP read from 85.14.192.197:2801 --->
 
 
 
<------------->
Retransmitting #4 (NAT) to 85.14.192.197:5060:
NOTIFY sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK07931a4c;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as508ab0e0
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 634427706cda187d36d813a30a4eaf1c@85.13.233.232
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
 
Messages-Waiting: no
Message-Account: sip:*97@85.13.233.232
Voice-Message: 0/0 (0/0)
 
---
Retransmitting #5 (NAT) to 85.14.192.197:5060:
NOTIFY sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK07931a4c;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as508ab0e0
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 634427706cda187d36d813a30a4eaf1c@85.13.233.232
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
 
Messages-Waiting: no
Message-Account: sip:*97@85.13.233.232
Voice-Message: 0/0 (0/0)
 
---
Reliably Transmitting (NAT) to 85.14.192.197:5060:
OPTIONS sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK5b0e5c1d;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as6f3cee3c
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 12f8eca906085b6073434b096a54fbb2@85.13.233.232
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Sep 2009 12:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
 
 
---
Retransmitting #1 (NAT) to 85.14.192.197:5060:
OPTIONS sip:198@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 85.13.233.232:5060;branch=z9hG4bK5b0e5c1d;rport
From: "Unknown" <sip:Unknown@85.13.233.232>;tag=as6f3cee3c
To: <sip:198@127.0.0.1>
Contact: <sip:Unknown@85.13.233.232>
Call-ID: 12f8eca906085b6073434b096a54fbb2@85.13.233.232
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Sep 2009 12:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

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by: denisdsr20Posted on 2009-09-22 at 01:13:19ID: 25390513

It seems your sipdroid does not respond to NOTIFY message (with Message Wait Information).
This causes asterisk to consider remote UA as unreachable.

One strange behaviour: the request should be addressed to 198@pbx1-uk.cxl.net but it refer to 198@127.0.0.1

Did you alias pbx1-uk.cxl.net and localhost somewhere in asterisk config or on asterisk box ?

I think this could be the problem.

Regards

Denis DIDIER

by: csupp_1Posted on 2009-09-27 at 11:46:05ID: 25434982

Using sipdroid 1.0.1 beta, the solutions is just use the option qualify=no on the extension ;)

20120131-EE-VQP-002

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