Hi,
I have Installed Asterisk on Synology.
After configuring Asterisk on Synology, i am able to make calls and receive calls with in the extns, but when I answer the call, call connects but no voice is heard. Only way to hear the voice from other end is to pause the call for a sec and then un-pause this. Then you can hear the voice.
Any help would be appreciated.
As I recall it, this is a closed system with a version of Asterisk provided by its vendors. I think you need to take this up with the vendors.
Otherwise you are going to have to provide all the usual information, e.g. versions, configuration, debugging logs.
Hi,
I have tried the Astrix support, but no help, since the support is a paid version. I have tried Synology and they suggested talk to Astrix. This was going in the loop.
Here is the version :
Asterisk/1.8.13.1
Asterisk GUI-version : 2.1.0-rc1
Configuration:
There is not much configuration done. I have set up just 2 extns and tried to calling each other.
There is a ring on each extns (no issues there)…
Calls gets answered (no issues there)…
After the call connection, no voice is heard. But if I put the call on hold and un-hold this then, the voice is audible.
Debug:
Either Debug log is not getting generated nor Astrix log is been generated.
THanks
Those are not the default names for the logs. You will need to edit logger.conf and also do:
core set debug 5
core set verbose 5
sip set debug on
at the Asterisk CLI.
The non-standard names suggest extensive customisation by Synology, so you really need to talk to them.
No much help from Synology.
It looks like you will need to do debugs from the Asterisk CLI. I hope you have access to that.
First set the verbosity level to 3 or more. After that make a call and copy/paste the output here. You can also issue a “sip set debug on”, do a call and copy/paste the output here.
What SIP clients are you using?
i Have been trying to debug, not able to do so. Can you please advise how to debug.
I have access to Asterix CLI. So what do we need to do here ?
Where do I get the Debug logs ?
After I make call, do i look at Asterix Logs ? I have been trying to look at this logs, but they are always empty.
Basically, I have just set up 2 users and trying to call each other.
1 User : 6001
2 User : 6002
The standard full log location is /var/log/asterisk/full. This normally needs to be enabled in /etc/asterisk/logger.conf. If these do not exist, please ask Synology.
Most people manage to produce debug logging without being given detailed instruction.
You can set verbosity level in Asterisk CLI with command:
core set verbose 3
When you make a call, some debugs should be printed directly in Asterisk CLI interface.
Also please copy/paste the output of the command:
sip show peers
Even After setting core set verbose 3 in CLI and then making some calls , I get the below Message shown in CLI:
Command>core show version
Asterisk 1.8.13.1 built by root @ ubuntu-pkg2 on a x86_64 running Linux on 2012-11-08 14:41:38 UTC
With the Statement sip show peers I get the below message:
Name/username Host Dyn Forcerport ACL Port Status
6001/6001 192.168.1.12 D N 5060 Unmonitored
6002/6002 192.168.1.3 D N 5066 Unmonitored
6004/6004 (Unspecified) D N 0 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline]
Please do:
sip set debug on
and make a call. If you still do not see any messages in the Asterisk CLI, you probably have a firewall issue.
Hi,
Even with statement " sip set debug on".
Still cannot generate Debugging logs .
I am not on external network. I am testing this process via internal Network.
Hi,
I managed to get the Sip log from Client rather than Server.
Please see if this helps.
REGISTER sip:192.168.1.25:5060 SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bKccb70f833a2ab2e02
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=df02dcf0b1
To: “6001” sip:6001@192.168.1.25:5060
Call-ID: 7b95686627da8fb1
CSeq: 282475283 REGISTER
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“6198fe3e”,uri=“sip:192.168.1.25:5060”,response=“1d6aac171554edf0cf3a15d817681b2d”,algorithm=MD5
Contact: sip:6001@192.168.1.1:5060;expires=120
User-Agent: Media5-fone/3.6.1.961
Content-Length: 0
– 2013-04-02 19:18:18 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bKccb70f833a2ab2e02;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=df02dcf0b1
To: “6001” sip:6001@192.168.1.25:5060;tag=as76e72fc8
Call-ID: 7b95686627da8fb1
CSeq: 282475283 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="303bc0d6"
Content-Length: 0
– 2013-04-02 19:18:18 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
REGISTER sip:192.168.1.25:5060 SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK9a17373340a4fd171
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=df02dcf0b1
To: “6001” sip:6001@192.168.1.25:5060
Call-ID: 7b95686627da8fb1
CSeq: 282475284 REGISTER
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“303bc0d6”,uri=“sip:192.168.1.25:5060”,response=“1dc1bbffdc101acee185c48682c23476”,algorithm=MD5
Contact: sip:6001@192.168.1.1:5060;expires=120
User-Agent: Media5-fone/3.6.1.961
Content-Length: 0
– 2013-04-02 19:18:18 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK9a17373340a4fd171;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=df02dcf0b1
To: “6001” sip:6001@192.168.1.25:5060;tag=as76e72fc8
Call-ID: 7b95686627da8fb1
CSeq: 282475284 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:6001@192.168.1.1:5060;expires=120
Date: Tue, 02 Apr 2013 08:18:53 GMT
Content-Length: 0
– 2013-04-02 19:18:29 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
INVITE sip:6002@192.168.1.25 SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK332e000a14971ddee
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
To: sip:6002@192.168.1.25
Call-ID: 318d090f1d755f61
CSeq: 1107080020 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Contact: sip:6001@192.168.1.1:5060
Supported: replaces
User-Agent: Media5-fone/3.6.1.961
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 241
– 2013-04-02 19:18:29 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK332e000a14971ddee;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
To: sip:6002@192.168.1.25;tag=as57aaace8
Call-ID: 318d090f1d755f61
CSeq: 1107080020 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="17d70e95"
Content-Length: 0
– 2013-04-02 19:18:29 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
ACK sip:6002@192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK332e000a14971ddee
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
To: sip:6002@192.168.1.25;tag=as57aaace8
Call-ID: 318d090f1d755f61
CSeq: 1107080020 ACK
User-Agent: Media5-fone/3.6.1.961
Content-Length: 0
– 2013-04-02 19:18:29 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
INVITE sip:6002@192.168.1.25 SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK5a39fe4b1763f4b38
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
To: sip:6002@192.168.1.25
Call-ID: 318d090f1d755f61
CSeq: 1107080021 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“17d70e95”,uri="sip:6002@192.168.1.25",response=“d33f642f9a2cd986df40de54e24c4701”,algorithm=MD5
Contact: sip:6001@192.168.1.1:5060
Supported: replaces
User-Agent: Media5-fone/3.6.1.961
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 241
– 2013-04-02 19:18:29 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK5a39fe4b1763f4b38;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
To: sip:6002@192.168.1.25
Call-ID: 318d090f1d755f61
CSeq: 1107080021 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:6002@192.168.1.25:5060
Content-Length: 0
– 2013-04-02 19:18:29 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK5a39fe4b1763f4b38;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
To: sip:6002@192.168.1.25;tag=as6567666d
Call-ID: 318d090f1d755f61
CSeq: 1107080021 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:6002@192.168.1.25:5060
Content-Length: 0
– 2013-04-02 19:18:31 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK5a39fe4b1763f4b38;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
To: sip:6002@192.168.1.25;tag=as6567666d
Call-ID: 318d090f1d755f61
CSeq: 1107080021 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:6002@192.168.1.25:5060
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 296924430 296924430 IN IP4 192.168.1.25
s=Asterisk PBX 1.8.13.1
c=IN IP4 192.168.1.25
t=0 0
m=audio 12592 RTP/AVP 8 125
a=rtpmap:8 PCMA/8000
a=rtpmap:125 telephone-event/8000
a=fmtp:125 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
– 2013-04-02 19:18:31 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
ACK sip:6002@192.168.1.25:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK77fcf7295bc668da0
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
To: sip:6002@192.168.1.25;tag=as6567666d
Call-ID: 318d090f1d755f61
CSeq: 1107080021 ACK
Allow-Events: refer
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“17d70e95”,uri="sip:6002@192.168.1.25",response=“d33f642f9a2cd986df40de54e24c4701”,algorithm=MD5
User-Agent: Media5-fone/3.6.1.961
Content-Length: 0
– 2013-04-02 19:18:57 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
BYE sip:6001@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK407b9442;rport
Max-Forwards: 70
From: sip:6002@192.168.1.25;tag=as6567666d
To: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
Call-ID: 318d090f1d755f61
CSeq: 102 BYE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username=“6001”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.25”, nonce="", response="7968def91e957bf060700af3250c3576"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
– 2013-04-02 19:18:57 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK407b9442;rport=5060;received=192.168.1.25
From: sip:6002@192.168.1.25;tag=as6567666d
To: “6001” sip:6001@192.168.1.25:5060;tag=e0230f0d12
Call-ID: 318d090f1d755f61
CSeq: 102 BYE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Server: Media5-fone/3.6.1.961
Supported: replaces
Content-Length: 0
RTCP statistics for Call: 318d090f1d755f61
Local Min Jitter: 0
Local Max Jitter: 0
Local Avg Jitter: 0
Local Packet Sent: 1141
Local Packet Lost: 0
Local Packet Recv: 0
Local Min Latency: 0
Local Max Latency: 0
Local Avg Latency: 0
Remote Jitter: 0
Remote Packet Sent: 0
Remote Packet Lost: 0
This looks like an Asterisk log, so I don’t understand what you mean when you say, that you got if from a Client. What do you mean by that? What Client are you using and how you made the debug?
The call shows a totally valid call leg from a Client (192.168.1.1) to Asterisk Server (192.168.1.25). I don’t see anything wrong here any I can’t understand how you can’t see any of the SIP messages in Asterisk server debugs, if the servers answers all the SIP messages.
I feel that you have a very strange topology issue. Can you please explain the exact topology (including the IP addresses of devices)?
He’s not using client and server in the SIP sense.
The Synology is a turnkey system, that includes a pre-configured instance of Asterisk. Unfortunately, it looks like Synology do not support this instance of Asterisk, even though the system is not a general purpose computer, hence people are coming here.
The system from which he eventually took the logs looks like another instance of Asterisk, with a modified user agent string. It may or may not be part of another turnkey system.
Hi All,
I am using Iphone with an app Media5-fone.
I have configured 6001 on Media5-fone on iphone
and configured 6002 on Media5-fone on another iphone.
192.168.1.25 is the Asterisk server.
There is no evidence of a pause in that log. If there was actually a pause, it has been actioned locally within the “client” and the fault lies there.
That is valid Point from David, there is no Pause in the log.
Should I try the log with out Pressing the Pause and post the log here ?
Hi,
Here is the log, without pressing pause button.
– 2013-04-08 06:23:57 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
INVITE sip:6002@192.168.1.25 SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK659057954077024da
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
To: sip:6002@192.168.1.25
Call-ID: 1d100b51f0f67cf3
CSeq: 1189254371 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Contact: sip:6001@192.168.1.1:5060
Supported: replaces
User-Agent: Media5-fone/3.7.1.1118
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 241
– 2013-04-08 06:23:57 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK659057954077024da;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
To: sip:6002@192.168.1.25;tag=as5c82d3ed
Call-ID: 1d100b51f0f67cf3
CSeq: 1189254371 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3a078ea6"
Content-Length: 0
– 2013-04-08 06:23:57 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
ACK sip:6002@192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK659057954077024da
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
To: sip:6002@192.168.1.25;tag=as5c82d3ed
Call-ID: 1d100b51f0f67cf3
CSeq: 1189254371 ACK
User-Agent: Media5-fone/3.7.1.1118
Content-Length: 0
– 2013-04-08 06:23:57 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
INVITE sip:6002@192.168.1.25 SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK63afbb57dc4f0e695
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
To: sip:6002@192.168.1.25
Call-ID: 1d100b51f0f67cf3
CSeq: 1189254372 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“3a078ea6”,uri="sip:6002@192.168.1.25",response=“d04a3c0f8587b39c65ce7b8fd382321c”,algorithm=MD5
Contact: sip:6001@192.168.1.1:5060
Supported: replaces
User-Agent: Media5-fone/3.7.1.1118
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 241
– 2013-04-08 06:23:57 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK63afbb57dc4f0e695;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
To: sip:6002@192.168.1.25
Call-ID: 1d100b51f0f67cf3
CSeq: 1189254372 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:6002@192.168.1.25:5060
Content-Length: 0
– 2013-04-08 06:23:57 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK63afbb57dc4f0e695;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
To: sip:6002@192.168.1.25;tag=as277662a8
Call-ID: 1d100b51f0f67cf3
CSeq: 1189254372 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:6002@192.168.1.25:5060
Content-Length: 0
– 2013-04-08 06:24:00 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK63afbb57dc4f0e695;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
To: sip:6002@192.168.1.25;tag=as277662a8
Call-ID: 1d100b51f0f67cf3
CSeq: 1189254372 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:6002@192.168.1.25:5060
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 2036306626 2036306626 IN IP4 192.168.1.25
s=Asterisk PBX 1.8.13.1
c=IN IP4 192.168.1.25
t=0 0
m=audio 19788 RTP/AVP 8 125
a=rtpmap:8 PCMA/8000
a=rtpmap:125 telephone-event/8000
a=fmtp:125 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
– 2013-04-08 06:24:00 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
ACK sip:6002@192.168.1.25:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bKf889d7eba95681da0
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
To: sip:6002@192.168.1.25;tag=as277662a8
Call-ID: 1d100b51f0f67cf3
CSeq: 1189254372 ACK
Allow-Events: refer
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“3a078ea6”,uri="sip:6002@192.168.1.25",response=“d04a3c0f8587b39c65ce7b8fd382321c”,algorithm=MD5
User-Agent: Media5-fone/3.7.1.1118
Content-Length: 0
– 2013-04-08 06:24:06 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
BYE sip:6001@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK7cf10420;rport
Max-Forwards: 70
From: sip:6002@192.168.1.25;tag=as277662a8
To: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
Call-ID: 1d100b51f0f67cf3
CSeq: 102 BYE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username=“6001”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.25”, nonce="", response="7968def91e957bf060700af3250c3576"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
– 2013-04-08 06:24:06 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK7cf10420;rport=5060;received=192.168.1.25
From: sip:6002@192.168.1.25;tag=as277662a8
To: “6001” sip:6001@192.168.1.25:5060;tag=f80e1bfe8f
Call-ID: 1d100b51f0f67cf3
CSeq: 102 BYE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Server: Media5-fone/3.7.1.1118
Supported: replaces
Content-Length: 0
RTCP statistics for Call: 1d100b51f0f67cf3
Local Min Jitter: 0
Local Max Jitter: 0
Local Avg Jitter: 0
Local Packet Sent: 90
Local Packet Lost: 0
Local Packet Recv: 0
Local Min Latency: 0
Local Max Latency: 0
Local Avg Latency: 0
Remote Jitter: 0
Remote Packet Sent: 0
Remote Packet Lost: 0
– 2013-04-08 06:24:07 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
REGISTER sip:192.168.1.25:5060 SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bKe27282b8ce40e7459
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=0f37520fd0
To: “6001” sip:6001@192.168.1.25:5060
Call-ID: 986f9a1f9f0dbb4a
CSeq: 282475251 REGISTER
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“31ca625d”,uri=“sip:192.168.1.25:5060”,response=“865346d0d5f0d337c1b20de8a9e0b64e”,algorithm=MD5
Contact: sip:6001@192.168.1.1:5060;expires=120
User-Agent: Media5-fone/3.7.1.1118
Content-Length: 0
– 2013-04-08 06:24:07 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bKe27282b8ce40e7459;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=0f37520fd0
To: “6001” sip:6001@192.168.1.25:5060;tag=as002132c6
Call-ID: 986f9a1f9f0dbb4a
CSeq: 282475251 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="23f3a34d"
Content-Length: 0
– 2013-04-08 06:24:07 - Sent to 192.168.1.25:5060 from 192.168.1.1:5060
REGISTER sip:192.168.1.25:5060 SIP/2.0
Accept: application/reginfo+xml, application/sdp, application/simple-message-summary, message/sipfrag, multipart/mixed, multipart/related
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK42cf14510c95846c6
Max-Forwards: 70
From: “6001” sip:6001@192.168.1.25:5060;tag=0f37520fd0
To: “6001” sip:6001@192.168.1.25:5060
Call-ID: 986f9a1f9f0dbb4a
CSeq: 282475252 REGISTER
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“23f3a34d”,uri=“sip:192.168.1.25:5060”,response=“ef786e47164e5f3fefb180eb6d777ea3”,algorithm=MD5
Contact: sip:6001@192.168.1.1:5060;expires=120
User-Agent: Media5-fone/3.7.1.1118
Content-Length: 0
– 2013-04-08 06:24:07 - Received from 192.168.1.25:5060 from 192.168.1.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK42cf14510c95846c6;received=192.168.1.1;rport=5060
From: “6001” sip:6001@192.168.1.25:5060;tag=0f37520fd0
To: “6001” sip:6001@192.168.1.25:5060;tag=as002132c6
Call-ID: 986f9a1f9f0dbb4a
CSeq: 282475252 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:6001@192.168.1.1:5060;expires=120
Date: Sun, 07 Apr 2013 20:24:44 GMT
Content-Length: 0
Hi All,
This has been resolved by modifying
progressinband=yes to sip.conf under the general area.