[asterisk-users] The call is established but without exchanged voice packets
Asmaa Ahmed
asabatgirl at hotmail.com
Fri Sep 20 09:31:30 CDT 2013
Hello,
Here is my extension context,
[internal]exten => 7001,1,Answer()exten => 7001,2,Dial(SIP/7001,60)exten => 7001,3,Playback(vm-nobodyavail)exten => 7001,4,VoiceMail(7001 at main) ;forward to voicemail mailboxexten => 7001,5,Hangup()
exten => 7002,1,Answer()exten => 7002,2,Dial(SIP/7002,60)exten => 7002,3,Playback(vm-nobodyavail)exten => 7002,4,VoiceMail(7002 at main)exten => 7002,5,Hangup()
exten => 7003,1,Answer()exten => 7003,2,Dial(SIP/7003,60)exten => 7003,3,Playback(vm-nobodyavail)exten => 7003,4,VoiceMail(7003 at main)exten => 7003,5,Hangup()
exten => 8001,1,VoicemailMain(7001 at main) ;voicemail retreivalexten => 8001,2,Hangup()
exten => 8002,1,VoicemailMain(7002 at main)exten => 8002,2,Hangup()
Date: Fri, 20 Sep 2013 16:25:42 +0200
From: asghar144 at gmail.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] The call is established but without exchanged voice packets
Hello,paste you extension context.
On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed <asabatgirl at hotmail.com> wrote:
Hello,
I have Asterisk 1.8.10.1Moving to nat=force_rport,comedia hasn't solved the problem. Still having the same error!
I am not sure if this is related to the problem here, but I was trying to test my voicemail and got this error "No audio available).
[Sep 20 14:05:41] WARNING[11424]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)[Sep 20 14:05:54] WARNING[11424]: app.c:855 __ast_play_and_record: No audio available on SIP/7001-00000001??
[Sep 20 14:06:13] WARNING[11387]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkNTY0YzZjMTcwNzcwYTg0NWRiMjlhYzQ4ZjFkOTc for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Thanks.
Date: Fri, 20 Sep 2013 16:05:35 +0200
From: asghar144 at gmail.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] The call is established but without exchanged voice packets
Hello,If Asterisk version is > 1.6 use nat=force_rport,comedia
On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed <asabatgirl at hotmail.com> wrote:
Hello,
I have set the direct media to be off, but still doesn't work. I am not sure about NAT configuration!
SIP.conf, [general] section
context=internalallowguest=noallowoverlap=notransport=udpbindport=5060bindaddr=0.0.0.0directmedia=nosrvlookup=nodisallow=all
allow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=<IP>localnet=172.16.0.255/255.255.255.0
The error messages
[Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7002[Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response[Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority)
Thanks.
Date: Thu, 19 Sep 2013 13:14:59 +0500
From: msalman212 at gmail.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] The call is established but without exchanged voice packets
Choose suitable NAT settings from sip.conf
turn direct media in sip.conf or per peer off
On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed <asabatgirl at hotmail.com> wrote:
Hello,
I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see this
chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response)
Here's my simple sip configuration
[general]
context=internal
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
alwaysauthreject=yes
canreinvite=no
nat=yes
session-timers=refuse
externip=<IP>
[7001]
type=friend
host=dynamic
secret=123
context=internal
[7002]
type=friend
host=dynamic
secret=456
context=internal
A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here!http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992
Thanks.
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Regards
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Muhammad Salman
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