Re: [asterisk-users] FXO -> GSM Gateway Problem
Hi Alec and Duncan
Thank you both for taking the time to reply.
Firstly regarding Duncan's post.
The asterisk makes a call fine when the FXO card is connected to a analogue
landline and the GSM Gateway works fine when connected to standard phone
handset. The problem only arises when I connect to GSM Gateway to the FXO
card. Within all 3 scenarios I'm using the same RJ11 to RJ11 cable.
Secondly regarding Alec's post.
I made the changes you suggested in the chan_dahdi.conf, rebooted and
tested. Unfortunately it didn't solve the issue.
Shown below is the new chan_dahdi.conf file and the Asterisk CLI for the
call using them setting.
chan_dahdi.conf
[channels]
signalling=fxs_ks
context=pstnincomming
group=0
answeronpolarityswitch=no
hanguponpolarityswitch=no
channel => 1
Asterisk CLI
== Using SIP RTP CoS mark 5
-- Executing [@sipofficephone:1] Verbose("SIP/lewisphone-0006",
"2,Call from VoIP network to ") in new stack
== Call from VoIP network to
-- Executing [@sipofficephone:2] Dial("SIP/lewisphone-0006",
"DAHDI/1/") in new stack
-- Called DAHDI/1/
-- DAHDI/1-1 answered SIP/lewisphone-0006
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
== Spawn extension (sipofficephone, , 2) exited non-zero on
'SIP/lewisphone-0006'
Best Regards
Lewis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alec Davis
Sent: 19 April 2012 00:10
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] FXO -> GSM Gateway Problem
>>
>> I have a problem where calling "out" of asterisk when the call is
answered dahdi hangs up immediately.
I'd make sure both answeronpolarityswitch and hanguponpolarityswitch are
either commented out or set to no.
from chan_dahdi.conf;
; Use a polarity reversal to mark when a outgoing call is answered by the ;
remote party.
;
;answeronpolarityswitch=yes
;
; In some countries, a polarity reversal is used to signal the disconnect of
a ; phone line. If the hanguponpolarityswitch option is selected, the call
will ; be considered "hung up" on a polarity reversal.
;
;hanguponpolarityswitch=yes
;
Alec Davis
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
--
_
-- 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
Re: [asterisk-users] FXO -> GSM Gateway Problem
>> >> I have a problem where calling "out" of asterisk when the call is answered dahdi hangs up immediately. I'd make sure both answeronpolarityswitch and hanguponpolarityswitch are either commented out or set to no. from chan_dahdi.conf; ; Use a polarity reversal to mark when a outgoing call is answered by the ; remote party. ; ;answeronpolarityswitch=yes ; ; In some countries, a polarity reversal is used to signal the disconnect of a ; phone line. If the hanguponpolarityswitch option is selected, the call will ; be considered "hung up" on a polarity reversal. ; ;hanguponpolarityswitch=yes ; Alec Davis -- _ -- 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
Re: [asterisk-users] FXO -> GSM Gateway Problem
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-000d", "2,Call from VoIP network to 01493857917")
> in new stack
> == Call from VoIP network to 01493857917
> -- Executing [01493857917@sipofficephone:2]
> Dial("SIP/lewisphone-000d", "DAHDI/1/01493857917") in new stack
> -- Called DAHDI/1/01493857917
> -- DAHDI/1-1 answered SIP/lewisphone-000d
> -- Hanging up on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
> == Spawn extension (sipofficephone, 01493857917, 2) exited non-zero on
> 'SIP/lewisphone-000d'
>
>
> 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-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 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=
> dtmfmode=auto
> disallow=all
> ;allow=ulaw
> allow=alaw
> allow=GSM
>
> [lewisphone](office-phone);lewis mobile
> secret=
>
> 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 [@sipofficephone:1] Verbose("SIP/lewisphone-000a",
> "2,Call from VoIP network to ") in new stack
> == Call from VoIP network to
> -- Executing [@sipofficephone:2] Dial("SIP/lewisphone-000a",
> "DAHDI/1/") in new stack
> -- Called DAHDI/1/
> -- DAHDI/1-1 answered SIP/lewisphone-00
Re: [asterisk-users] FXO -> GSM Gateway Problem
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-000d", "2,Call from VoIP network to
01493857917") in new stack
== Call from VoIP network to 01493857917
-- Executing [01493857917@sipofficephone:2]
Dial("SIP/lewisphone-000d", "DAHDI/1/01493857917") in new stack
-- Called DAHDI/1/01493857917
-- DAHDI/1-1 answered SIP/lewisphone-000d
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
== Spawn extension (sipofficephone, 01493857917, 2) exited non-zero on
'SIP/lewisphone-000d'
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-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 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=
dtmfmode=auto
disallow=all
;allow=ulaw
allow=alaw
allow=GSM
[lewisphone](office-phone);lewis mobile
secret=
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 [@sipofficephone:1] Verbose("SIP/lewisphone-000a",
"2,Call from VoIP network to ") in new stack
== Call from VoIP network to
-- Executing [@sipofficephone:2] Dial("SIP/lewisphone-000a",
"DAHDI/1/") in new stack
-- Called DAHDI/1/
-- DAHDI/1-1 answered SIP/lewisphone-000a 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, , 2) exited non-zero on
'SIP/lewisphone-000a'
Best Regards
Lewis
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
<>--
_
-- 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
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=
>
> dtmfmode=auto
>
> disallow=all
>
> ;allow=ulaw
>
> allow=alaw
>
> allow=GSM
>
> ** **
>
> [lewisphone](office-phone);lewis mobile
>
> secret=
>
> ** **
>
> *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 [@sipofficephone:1]
> Verbose("SIP/lewisphone-000a", "2,Call from VoIP network to ") in
> new stack
>
> == Call from VoIP network to
>
> -- Executing [@sipofficephone:2] Dial("SIP/lewisphone-000a",
> "DAHDI/1/") in new stack
>
> -- Called DAHDI/1/
>
> -- DAHDI/1-1 answered SIP/lewisphone-000a *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, , 2) exited non-zero on
> 'SIP/lewisphone-000a'
>
> ** **
>
> ** **
>
> ** **
>
> Best Regards
>
> *
>
> *
>
> Lewis
>
> [image: digitalselect-e]
>
> www.Digital-Select.com
[asterisk-users] FXO -> GSM Gateway Problem
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=
dtmfmode=auto
disallow=all
;allow=ulaw
allow=alaw
allow=GSM
[lewisphone](office-phone);lewis mobile
secret=
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 [@sipofficephone:1] Verbose("SIP/lewisphone-000a",
"2,Call from VoIP network to ") in new stack
== Call from VoIP network to
-- Executing [@sipofficephone:2] Dial("SIP/lewisphone-000a",
"DAHDI/1/") in new stack
-- Called DAHDI/1/
-- DAHDI/1-1 answered SIP/lewisphone-000a 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, , 2) exited non-zero on
'SIP/lewisphone-000a'
Best Regards
Lewis
digitalselect-e
www.Digital-Select.com
