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 <[email protected]> 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 **** > > [image: digitalselect-e]**** > > www.Digital-Select.com <http://www.digital-select.com/>**** > > * > > ***** > > ** ** > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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