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

[Matthew J. Roth]

Asmaa Ahmed wrote: 
> 
> Indeed I missed your previous message! 
> After changing the externip, it worked successfully... The sip
> session is established with the complete three-way handshake, and
> the voice packet is exchanged with no problem! 
> 
> Many thanks.


Asmaa,

That's great news!!  I guess the firewall settings were already
correct and it was just a matter of configuring Asterisk properly.

In my experience, the first call is always the hardest one to get
working.  Now that you've done that you can really start seeing what
Asterisk can do.  Have fun, but remember to take it step by step and
don't hesitate to ask the list if you run into any problems.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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

[Asmaa Ahmed]

Hi Matthew,
Indeed I missed your previous message!After changing the externip, it worked 
successfully... The sip session is established with the complete  three-way 
handshake, and the voice packet is exchanged with no problem!
Many thanks.   
> Date: Fri, 20 Sep 2013 10:01:52 -0500
> From: mr...@imminc.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] The call is established but without exchanged 
> voice packets
> 
> Asmaa, 
> 
> You're getting ahead of yourself.  How do you expect audio to work if
> your firewall/NAT settings aren't even configured correctly to
> establish SIP sessions?
> 
> Go back and read the message that I sent yesterday.  Fix the SIP 
> three-way handshake problem.  That is step 1 and you'll know you have
> it right when you stop seeing 'Retransmission timeout reached on
> transmission' errors.
> 
> You still won't have audio but that's step 2.  It requires properly
> configuring Asterisk's NAT settings and the firewall(s) between the
> phones and the server to allow RTP traffic to flow, but don't worry
> about it until step 1 is complete.
> 
> Regards,
> 
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
> 
> --
> _
> -- 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
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_
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Re: [asterisk-users] The call is established but without exchanged voice packets

[Asghar Mohammad]

Hello,
paste you extension context.


On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed  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/7001-0001??
> [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 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=
> 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 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=
>
> [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.filec

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

[Asmaa Ahmed]

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  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/7001-0001??
[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  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=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  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=


[7001]


type=friend


host=dynamic


secret=123


context=internal


[7002]


type=friend


host=dyn

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

[Matthew J. Roth]

Asmaa, 

You're getting ahead of yourself.  How do you expect audio to work if
your firewall/NAT settings aren't even configured correctly to
establish SIP sessions?

Go back and read the message that I sent yesterday.  Fix the SIP 
three-way handshake problem.  That is step 1 and you'll know you have
it right when you stop seeing 'Retransmission timeout reached on
transmission' errors.

You still won't have audio but that's step 2.  It requires properly
configuring Asterisk's NAT settings and the firewall(s) between the
phones and the server to allow RTP traffic to flow, but don't worry
about it until step 1 is complete.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

--
_
-- 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

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

[Asghar Mohammad]

Hello,
i think your logic is wrong please explain me what are you trying to do?
[internal]
exten => 7002,1,Answer()
exten => 7002,n,Playback(vm-nobodyavail)
exten => 7002,n,Hangup()

exten => 7001,1,Dial(SIP/7001,60)
exten => 7001,n,Hangup()

try this dial 7002 and you should listen vm-nobodyavail or 7001 to 7001
extension.


On Fri, Sep 20, 2013 at 4:31 PM, Asmaa Ahmed  wrote:

> 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 mailbox
> exten => 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 retreival
> exten => 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 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/7001-0001??
> [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 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=
> 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 

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

[Asmaa Ahmed]

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/7001-0001??[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  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=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  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=

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

[Asghar Mohammad]

Hello,
If Asterisk version is > 1.6 use nat=force_rport,comedia


On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed  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=
> 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 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=
>
> [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.
>
>
> --
> _
> -- 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:
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> New to Asterisk? Join us for a live

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

[Asmaa Ahmed]

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] 
sectioncontext=internalallowguest=noallowoverlap=notransport=udpbindport=5060bindaddr=0.0.0.0directmedia=nosrvlookup=nodisallow=allallow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=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+RetransmissionsPacket 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  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=
[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

**
Muhammad Salman
***



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

[Matthew J. Roth]

Asmaa Ahmed wrote:
> 
> 
> 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, a nd 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) 

The SIP trace you provided breaks down as follows:

  X-Lite   Asterisk
  ---  ---
  INVITE(No Auth) ---> 
  <--- 401 Unauthorized
  ACK --->
  INVITE(Auth)--->
  <--- 100 Trying
  <--- 200 OK
  <--- 200 OK (Retransmitted 10 Times)
  <--- BYE
  OK  --->

This shows that the three-way handshake (INVITE/200 OK/ACK) used to
establish SIP sessions is not completed because Asterisk never
receives an ACK from X-Lite.  After retransmitting the 200 OK 10 times
Asterisk gives up and disconnects the call.

> 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= 

>From the SIP trace, I believe 'externip=41.46.164.96' is set.  If that
is the case, try changing it to 'externip=54.241.129.14'.  You should
also set localnet as follows:

  ; RFC 1918 addresses
  localnet=192.168.0.0/255.255.0.0
  localnet=10.0.0.0/255.0.0.0
  localnet=172.16.0.0/12

If that doesn't work you can also try setting 'nat=force_rport'
instead of 'nat=yes'.

> [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
>  

At this point, you should be able to establish a call between the two
X-Lite phones that won't get disconnected due to failing to complete
the three-way handshake.  There may still not be voice because the
firewall(s) between Asterisk and the X-Lite phones may block the RTP
traffic.  The phones appear to be on the same network, so you can try
setting 'canreinvite=yes' to workaround this problem until the
firewall(s) are configured to allow RTP traffic on the UDP port range
specified in 'rtp.conf' (the default range is 1-2).

Good luck and please report your progress back to the list.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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

[Salman Zafar]

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

[Asmaa Ahmed]

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 thischan_sip.c:3641 retrans_pkt: Retransmission timeout reached 
on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 
(Critical Response) Here's my  simple sip 
configuration[general]context=internalallowguest=noallowoverlap=nobindport=5060bindaddr=0.0.0.0srvlookup=nodisallow=allallow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=[7001]type=friendhost=dynamicsecret=123context=internal[7002]type=friendhost=dynamicsecret=456context=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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