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@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@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@main)exten => 
7003,5,Hangup()
exten => 8001,1,VoicemailMain(7001@main) ;voicemail retreivalexten => 
8001,2,Hangup()
exten => 8002,1,VoicemailMain(7002@main)exten => 8002,2,Hangup()
Date: Fri, 20 Sep 2013 16:25:42 +0200
From: asghar...@gmail.com
To: asterisk-users@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 <asabatg...@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: asghar...@gmail.com
To: asterisk-users@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 <asabatg...@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: msalman...@gmail.com
To: asterisk-users@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 <asabatg...@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.
                                          

--

_____________________________________________________________________

-- 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



asterisk-users mailing list

To UNSUBSCRIBE or update options visit:

   http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Regards

**************************
Muhammad Salman
***************************



--
_____________________________________________________________________
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users                      
                  

--

_____________________________________________________________________

-- 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



asterisk-users mailing list

To UNSUBSCRIBE or update options visit:

   http://lists.digium.com/mailman/listinfo/asterisk-users



--
_____________________________________________________________________
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users                      
                  

--

_____________________________________________________________________

-- 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



asterisk-users mailing list

To UNSUBSCRIBE or update options visit:

   http://lists.digium.com/mailman/listinfo/asterisk-users



--
_____________________________________________________________________
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users                      
                  
--
_____________________________________________________________________
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to