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 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
>
>
>
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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 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
>
> 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
>
> –
> _____________________________________________________________________
> — 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

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

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 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
>

I have a compatibilty problem between asterisk 1.4 and 1.6.2 In my 1.4 asterisk I have a custom application that users call and make recordings which recording I save to a file with the caller Id.

Below is the config file which works perfectly in 1.4

In my 1.6 version I use the same configuration in extensions_custom.conf but I get the error below. It seems like 1.6 does not recognize the button the user has pressed.

As I’ve occasionally posted here before, I have user terminals which can
accept SIP text messages to an SMS-like interface.

After upgrading to Asterisk 10, I do indeed have external processes
generating these messages. But it’s a bit ugly. What I’d _like_ to do is
simply generate a callfile, and something like this almost works:

Channel: Local/8902
Application: MessageSend
Set: MESSAGE(body)=messagebody
Data: sip:glowworm
Data: sip:glowworm

but (a) I need that reserved local number to let the call work at all
(the number just does an Answer(), Wait(10), Hangup) and (b) I can’t
seem to set the sender’s name. That ought to be the second Data
parameter; actually the second one seems to determine where the message
goes, and whatever I set the first one to the sender name always comes
up as “asterisk”. (Specifically, in the packet capture, I have

From: “asterisk”

.) Now, I _can_ achieve the desired result, but only by having _another_
local number that does

exten => 8901,n,SET(MESSAGE(body)=${msg_out_body})
exten => 8901,n,MessageSend(${msg_out_to},${msg_out_from})

and setting up the callfile with:

Extension: 8901
Set: msg_out_to=glowworm
Set: msg_out_from=

at which point the message will appear to originate from FROM (note that
if I put a display name component in the msg_out_from it gets ignored -
but that is the terminals’ peculiarity). But that’s ugly. Has anyone got
this working with a relatively straight callfile setup?

While I’m writing, does Asterisk 10 have any way to send a SIP message
that isn’t text/plain?

Roger