[asterisk-users] The call is established but without exchanged voice packets

Fri Sep 20 09:25:42 CDT 2013

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 1.8.10.1
> Moving 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=internal
> allowguest=no
> allowoverlap=no
> transport=udp
> bindport=5060
> bindaddr=0.0.0.0
> directmedia=no
> srvlookup=no
> disallow=all
> allow=ulaw
> alwaysauthreject=yes
> canreinvite=no
> nat=yes
> session-timers=refuse
> externip=<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 <http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992>
> Thanks.
>
>
> --
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>
>
>
> --
> Regards
>
> **************************
> Muhammad Salman
> ***************************
>
>
> -- _____________________________________________________________________
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>
> -- _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
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> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
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