[asterisk-users] FXO -> GSM Gateway Problem

[Duncan Turnbull]

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 <tech at 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 at sipofficephone:1] Verbose("SIP/lewisphone-0000000d", "2,Call from VoIP network to 01493857917") in new stack
>   == Call from VoIP network to 01493857917
>     -- Executing [01493857917 at 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-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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 <tech at 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 at sipofficephone:1] Verbose("SIP/lewisphone-0000000a", "2,Call from VoIP network to xxxx") in new stack
>   == Call from VoIP network to xxxx
>     -- Executing [xxxx at 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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