[asterisk-users] callback and bridge problem

[Dovid B]

Are you behind NAT ? Do you have canreinvite=yes ?

----- Original Message ----- 
From: "Adam KOSA" <adamk at 3a.hu>
To: <asterisk-users at lists.digium.com>
Sent: Monday, June 25, 2007 6:37 PM
Subject: [asterisk-users] callback and bridge problem


> Hi guys,
>
> sorry for the long e-mail, i'm only trying to give as much information
> as i think is relevant to my problem (console log, sip.conf and
> extension.conf parts).
>
> i've been practicing with callback for a while, but i'm at a dead end.
> I hope somebody can help me to move on.
>
> i have troubles getting two calls bridged together.  Scenario is the
> following:
>
> - asterisk calls my cell via a SIP provider called neophone
> - my cell rings, i pick up, and i find myself in:
>
> [internal]
> ; callback is directed here
> exten => s,1,WaitExten,50
> include => voicemail-context
> include => internal_extensions-context
> include => dialout_prefix-context
>
>
> because my call file looks like this:
>
> Channel: SIP/06202222222 at neophonex
> Context: internal
> Extension: s
> Priority: 1
>
> where 06202222222 is my cell.
>
> - after picking up, i dial 95206301111111 where 952 is the dialing
> prefix, 06301111... is another cell.  952 is a prefix for another
> registered account at the same provider (one account is allowed to place
> one call at a time).
>
> After this as you can see, the second number (1111..) is dialed.
> However when i pick up the phone, the call hangs up.
>
> This also happens when i use another prefix (another provider, even
> PSTN) for the second call too.
>
> The relevant part from asterisk console is at the end of this e-mail, i
> don't really understand the warning messages.
>
> ----- configs:
>
> In sip.conf, the configuration for the two SIP accounts are:
>
> register => 0621380....:password at sip.neophonex.hu
> register => 0621381....:password at sip.neophonex.hu
>
> [neophonex]
> type=friend
> host=sip.neophonex.hu
> context=dialout_prefix-context
> username=0621380....
> authname=0621380....
> fromuser=0621380....
> secret=password
> callerid=0621380....
> fromdomain=sip.neophonex.hu
> disallow=all
> allow=alaw
> allow=g723
> dtmfmode=inband
> nat=no
>
> [neophonex-out]
> type=friend
> host=sip.neophonex.hu
> context=dialout_prefix-context
> username=0621381....
> authname=0621381....
> fromuser=0621381....
> secret=password
> callerid=0621381....
> fromdomain=sip.neophonex.hu
> disallow=all
> allow=alaw
> allow=g723
> dtmfmode=inband
> nat=no
>
>
> extension.conf:
>
> exten => _952.,1,Playback(kapcsolas,noanswer)
> exten => _952.,n,Set(CALLERID(name)=0621380....)
> exten => _952.,n,Dial(SIP/${EXTEN:3}@neophonex-out)
>
> I have tried every possible setting i know about, but still, when i call
> outside, via 'turning around' in asterisk, both cells hung up when
> answering the call.  I have tried calling a regular landline phone
> number but still hanging up.
>
> Both accounts are valid, registered and have enough credit to dial
> outside its voice network.
>
> The only way the call does not hung up is when i dial extensions within
> asterisk.
>
> The asterisk log:
>
>     -- Called 06301111111 at neophonex-out
>     -- Call on SIP/neophonex-out-081a9cc0 left from hold
>     -- SIP/neophonex-out-081a9cc0 is making progress passing it to
> SIP/neophonex-081ab240
> [Jun 25 16:57:07] WARNING[18232]: chan_sip.c:11839
> handle_response_invite: Re-invite to non-existing call leg on other UA.
> SIP dialog '44c971692552f3245aa7b4e834bdafab at sip.neophonex.hu'. Giving up.
>     -- Call on SIP/neophonex-out-081a9cc0 left from hold
>     -- SIP/neophonex-out-081a9cc0 answered SIP/neophonex-081ab240
>     -- Native bridging SIP/neophonex-081ab240 and
> SIP/neophonex-out-081a9cc0
> [Jun 25 16:57:10] WARNING[18232]: chan_sip.c:11839
> handle_response_invite: Re-invite to non-existing call leg on other UA.
> SIP dialog '240370c953d10a75430c0e2e0d4764a6 at sip.neophonex.hu'. Giving up.
>   == Spawn extension (internal, 95206301111111, 3) exited non-zero on
> 'SIP/neophonex-081ab240'
> [Jun 25 16:57:10] NOTICE[18440]: pbx_spool.c:351 attempt_thread: Call
> completed to SIP/06202222222 at neophonex
>
>
> Please help me to figure out why the calls are hung up.
>
> Thanks
> Adam
>
>
>
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