Hi
I have had issues with wiring for incoming calls causing what looks like a
hangup when answered but in those cases the call stays up and asterisk thinks
its a new call. Have seen it on Avaya too
If it is wiring can you test a different incoming line?
Cheers duncan
On 19/04/2012, at 1:54 AM, Tech <t...@digital-select.com> wrote:
> Thanks Dhaval for taking the time to look at my question.
>
> I have tried to print the hangup cause however as you can see below it
> doesn't show that section of the dialplan.
> I have ammended below the CLI and extensions.conf with the changes I made.
>
> ASTERISK CLI
> == Using SIP RTP CoS mark 5
> -- Executing [01493857917@sipofficephone:1]
> Verbose("SIP/lewisphone-0000000d", "2,Call from VoIP network to 01493857917")
> in new stack
> == Call from VoIP network to 01493857917
> -- Executing [01493857917@sipofficephone:2]
> Dial("SIP/lewisphone-0000000d", "DAHDI/1/01493857917") in new stack
> -- Called DAHDI/1/01493857917
> -- DAHDI/1-1 answered SIP/lewisphone-0000000d
> -- Hanging up on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
> == Spawn extension (sipofficephone, 01493857917, 2) exited non-zero on
> 'SIP/lewisphone-0000000d'
>
>
> extensions.conf
> [sipofficephone]
>
> exten => _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})
> same => n,Dial(DAHDI/1/${EXTEN})
> same => n,Verbose(2, Hangup Cause ${HANGUPCAUSE})
> same => n,Hangup()
>
> [pstnincomming]
>
> exten => s,1,Answer()
> same => n,Dial(SIP/lewisphone)
> same => n,Hangup()
>
> Best Regards
>
>
>
> Lewis
> <image001.gif>
>
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
> Sent: 18 April 2012 13:18
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FXO -> GSM Gateway Problem
>
> Hi,
>
> It can be codec negotiation error or else plese try to print hangupcause sent
> from telco
>
>
>
> On Wed, Apr 18, 2012 at 4:27 PM, Tech <t...@digital-select.com> wrote:
> Hi,
>
> I have a problem where calling "out" of asterisk when the call is answered
> dahdi hangs up immediately.
> For example: Sip Client A calls external number. Route: SIP -> FXO -> GSM
> Gateway ->External Landline.
> However when that external landline answers the call dahdi hangs up
> immediately .
>
> Going the other way is fine (External Landline -> GSM Gateway -> FXO -> SIP).
>
> I've tried multiple different internet searches and can't seem to find any
> information on this problem.
>
> Below are my config files.
>
> Sip.conf
> [office-phone](!)
> type=friend
> context=sipofficephone
> host=dynamic
> nat=yes
> #secret=xxxx
> dtmfmode=auto
> disallow=all
> ;allow=ulaw
> allow=alaw
> allow=GSM
>
> [lewisphone](office-phone);lewis mobile
> secret=xxxx
>
> Chan_dahdi.conf
> [channels]
> signalling=fxs_ks
> context=pstnincomming
> group=0
> channel => 1
>
>
> Extensions.conf
> [sipofficephone]
> exten => _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})
> same => n,Dial(DAHDI/1/${EXTEN})
> same => n,Hangup()
>
> [pstnincomming]Diamon
> exten => s,1,Answer()
> same => n,Dial(SIP/lewisphone)
> same => n,Hangup()
>
>
> Asterisk CLI Output (Verbose 3)
> My comments bold.
>
> == Using SIP RTP CoS mark 5
> -- Executing [xxxx@sipofficephone:1] Verbose("SIP/lewisphone-0000000a",
> "2,Call from VoIP network to xxxx") in new stack
> == Call from VoIP network to xxxx
> -- Executing [xxxx@sipofficephone:2] Dial("SIP/lewisphone-0000000a",
> "DAHDI/1/xxxx") in new stack
> -- Called DAHDI/1/xxxx
> -- DAHDI/1-1 answered SIP/lewisphone-0000000a GSM Gateway Answering Call
> then Sending it out.
> -- Hanging up on 'DAHDI/1-1' Dest answering call to which DAHDI hangs up
> -- Hungup 'DAHDI/1-1'
> == Spawn extension (sipofficephone, xxxx, 2) exited non-zero on
> 'SIP/lewisphone-0000000a'
>
>
>
> Best Regards
>
>
>
> Lewis
> <image001.gif>
> www.Digital-Select.com
>
>
>
>
>
> --
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