[Asterisk-Users] No voice - one way - both ways

[Rudolf Ladyzhenskii]

Firewall/NAT problem?

Are all phones on same subnet?

Rudolf

----- Original Message ----- 
From: "Ronald Wiplinger" <ronald at elmit.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Sunday, October 16, 2005 9:29 PM
Subject: [Asterisk-Users] No voice - one way - both ways


>I got four phones:
>
> 601 is a SIP phone (no brand)
> 615 is Snom 190
> 621 is a Grand stream
> 628 is a remote SIP phone (no brand)
>
> 601, 615, 628 can call each other without any problems
>
> 621 used to be able to call remote 628, but after upgrade to CVS Head Nov. 
> 11 the remote party cannot hear me.
> 615 never could call remote 628, both party hear nothing.
> 601 can always call 628
>
>
>
> [Oct 16 00:52:13] -- Executing Dial("SIP/621-673f", "SIP/628|60|r") in new 
> stack
> [Oct 16 00:52:13] -- Called 628
> [Oct 16 00:52:13] -- SIP/628-9d23 is ringing
> [Oct 16 00:52:15] -- SIP/628-9d23 answered SIP/621-673f
> [Oct 16 00:52:15] -- Attempting native bridge of SIP/621-673f and 
> SIP/628-9d23
>
> She cannot here me!!!
>
>
> [Oct 16 00:52:30] == Spawn extension (default, 628, 1) exited non-zero on 
> 'SIP/621-673f'
> [Oct 16 00:52:30] -- Executing Hangup("SIP/621-673f", "") in new stack
> [Oct 16 00:52:30] == Spawn extension (default, h, 1) exited non-zero on 
> 'SIP/621-673f'
> [Oct 16 00:53:06] -- Executing Playback("SIP/621-88e8", "demo-echotest") 
> in new stack
> [Oct 16 00:53:06] -- Playing 'demo-echotest' (language 'en')
> [Oct 16 00:53:26] -- Executing Echo("SIP/621-88e8", "") in new stack
> [Oct 16 00:53:33] == Spawn extension (default, 690, 2) exited non-zero on 
> 'SIP/621-88e8'
> [Oct 16 00:53:33] -- Executing Hangup("SIP/621-88e8", "") in new stack
> [Oct 16 00:53:33] == Spawn extension (default, h, 1) exited non-zero on 
> 'SIP/621-88e8'
>
> Echo test no problem, means phone is ok!!
>
>
> [Oct 16 00:53:41] -- Executing Dial("SIP/621-b113", "SIP/628|60|r") in new 
> stack
> [Oct 16 00:53:41] -- Called 628
> [Oct 16 00:53:41] -- SIP/628-b3b6 is ringing
> [Oct 16 00:53:51] -- SIP/628-b3b6 answered SIP/621-b113
> [Oct 16 00:53:51] -- Attempting native bridge of SIP/621-b113 and 
> SIP/628-b3b6
> [Oct 16 00:53:58] == Spawn extension (default, 628, 1) exited non-zero on 
> 'SIP/621-b113'
> [Oct 16 00:53:58] -- Executing Hangup("SIP/621-b113", "") in new stack
> [Oct 16 00:53:58] == Spawn extension (default, h, 1) exited non-zero on 
> 'SIP/621-b113'
>
> She cannot hear me
>
>
>
> [Oct 16 00:55:19] -- Executing Hangup("SIP/615-a5bd", "") in new stack
> [Oct 16 00:55:19] == Spawn extension (default, h, 1) exited non-zero on 
> 'SIP/615-a5bd'
> [Oct 16 00:55:23] == Spawn extension (VoIP_customer_Phone_routes, 621, 2) 
> exited non-zero on 'SIP/628-aba4'
> [Oct 16 00:55:35] -- Executing Dial("SIP/615-31a8", "SIP/628|60|r") in new 
> stack
> [Oct 16 00:55:35] -- Called 628
> [Oct 16 00:55:36] -- SIP/628-7293 is ringing
> [Oct 16 00:55:42] -- SIP/628-7293 answered SIP/615-31a8
> [Oct 16 00:55:42] -- Attempting native bridge of SIP/615-31a8 and 
> SIP/628-7293
> [Oct 16 00:55:51] == Spawn extension (default, 628, 1) exited non-zero on 
> 'SIP/615-31a8'
> [Oct 16 00:55:51] -- Executing Hangup("SIP/615-31a8", "") in new stack
> [Oct 16 00:55:51] == Spawn extension (default, h, 1) exited non-zero on 
> 'SIP/615-31a8'
>
> We both cannot hear
>
>
> [Oct 16 00:56:08] -- Executing Dial("SIP/601-bb26", "SIP/628|60|r") in new 
> stack
> [Oct 16 00:56:08] -- Called 628
> [Oct 16 00:56:09] -- SIP/628-0be9 is ringing
> [Oct 16 00:56:16] -- SIP/628-0be9 answered SIP/601-bb26
> [Oct 16 00:56:16] -- Attempting native bridge of SIP/601-bb26 and 
> SIP/628-0be9
> [Oct 16 00:58:36] == Spawn extension (default, 628, 1) exited non-zero on 
> 'SIP/601-bb26'
> [Oct 16 00:58:36] -- Executing Hangup("SIP/601-bb26", "") in new stack
> [Oct 16 00:58:36] == Spawn extension (default, h, 1) exited non-zero on 
> 'SIP/601-bb26'
>
> Call ok!!!
>
>
>
> SIP.conf:
>
> [601]
> type=friend
> username=601
> secret=youdontneedtoknow
> canreinvite=no
> host=dynamic
> dtmfmode=rfc2833
> mailbox=601 at other
> nat=yes
> callgroup=1
> pickupgroup=1
> callerid="Ronald Hotline",<601>
> qualify=1000
>
> [615] ; snom 190
> type=friend ; Friends place calls and receive calls
> username=615 ; Username to use in INVITE until peer registers
> secret=youdontneedtoknow
> host=dynamic ; This peer register with us
> dtmfmode=rfc2833
> qualify=1000
> mailbox=615 at other ; Mailboxes for message waiting indicator
> restrictcid=yes ; To have the callerid restriced -> sent as ANI
> disallow=all
> allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
> allow=alaw
> allow=g729
> callerid="Ronald Snom",<615>
> callgroup=1
> pickupgroup=1
>
>
> 621 and 628 are in realtime and have similar settings. Important I think 
> is only the codec:
> 621: ulaw;alaw
> 628: g729;ulaw;alaw
>
>
> How can I solve it?
>
>
> bye
>
> Ronald Wiplinger
>
> ===============================
> First they ignore you, then they laugh at you,
> then they fight you, then you win.
> —Mahatma Gandhi
>
>
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