[asterisk-users] FXO -> GSM Gateway Problem

[Tech]

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 

digitalselect-e

 


 

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 

digitalselect-e

www.Digital-Select.com <http://www.digital-select.com/> 

 


 


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