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

[Rich Adamson]

> 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?

Try canreinvite=no on each sip phone definition and test again.

It sounds like some of these phones are behind nat boxes, and 
that type of info is somewhat important for anyone to guess at
what you're doing.