thr3ads.net - asterisk users - [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing? [Feb 2009]

If this information is useful, please help other people find it:
Share via:

joekane at gmail.com

2009-Feb-12 08:56 UTC

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

Hi all,

I have a connect between a siemens hipath & Asterisk system over PRI
The connection works perfectly I can call from the Hipath to an Asterisk
Extension.

I want to allow the hipath extensions to dial out over a SIP trunk on
asterisk but I keep getting "The number you have dialed is not in
service"

In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk) then
the number 1905 (Freefone number in Ireland)

Please help I cant figure this one out.

Thanks, Joe

CLI -

[Feb 11 17:45:25] VERBOSE[4526] logger.c:     -- Accepting overlap call from
'0339' to '<unspecified>' on channel 0/31, span 1
[Feb 11 17:45:25] VERBOSE[5764] logger.c:     -- Starting simple switch on
'Zap/31-1'
[Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing [91905 at
from-pstn:1]
Set("Zap/31-1", "__FROM_DID=91905") in new stack
[Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing [91905 at
from-pstn:2]
NoOp("Zap/31-1", "Received an unknown call with DID set to
91905") in new
stack
[Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing [91905 at
from-pstn:3]
Goto("Zap/31-1", "s|a2") in new stack
[Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Goto (from-pstn,s,2)
[Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing [s at from-pstn:2]
Answer("Zap/31-1", "") in new stack
[Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing [s at from-pstn:3]
Wait("Zap/31-1", "2") in new stack
[Feb 11 17:45:33] VERBOSE[5764] logger.c:     -- Executing [s at from-pstn:4]
Playback("Zap/31-1", "ss-noservice") in new stack
[Feb 11 17:45:33] VERBOSE[5764] logger.c:     -- <Zap/31-1> Playing
'ss-noservice' (language 'en')
[Feb 11 17:45:38] VERBOSE[5764] logger.c:     -- Executing [s at from-pstn:5]
SayAlpha("Zap/31-1", "91905") in new stack
[Feb 11 17:45:38] VERBOSE[5764] logger.c:     -- <Zap/31-1> Playing
'digits/9' (language 'en')
[Feb 11 17:45:39] VERBOSE[4526] logger.c:     -- Channel 0/31, span 1 got
hangup request, cause 16
[Feb 11 17:45:39] WARNING[5764] file.c: Failed to write frame
[Feb 11 17:45:39] VERBOSE[5764] logger.c:   == Spawn extension (from-pstn,
s, 5) exited non-zero on 'Zap/31-1'
[Feb 11 17:45:39] VERBOSE[5764] logger.c:     -- Executing [h at from-pstn:1]
Hangup("Zap/31-1", "") in new stack
[Feb 11 17:45:39] VERBOSE[5764] logger.c:   == Spawn extension (from-pstn,
h, 1) exited non-zero on 'Zap/31-1'
[Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value:
ON(1) on Zap/31-1
[Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Not yet hungup...  Calling
hangup once with icause, and clearing call
[Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value:
OFF(0) on Zap/31-1
[Feb 11 17:45:39] VERBOSE[5764] logger.c:     -- Hungup 'Zap/31-1'
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090212/023e4a93/attachment.htm

Rob Hillis

2009-Feb-12 09:10 UTC

head link

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

Which line of code is generating this log entry?

[Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing
[91905 at from-pstn:3] Goto("Zap/31-1", "s|a2") in new stack

...because this appears to be where your problem lies.


joekane at gmail.com wrote:> Hi all,
>
> I have a connect between a siemens hipath & Asterisk system over PRI
> The connection works perfectly I can call from the Hipath to an
> Asterisk Extension.
>
> I want to allow the hipath extensions to dial out over a SIP trunk on
> asterisk but I keep getting "The number you have dialed is not in
service"
>
> In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)
> then the number 1905 (Freefone number in Ireland)
>
> Please help I cant figure this one out.
>
> Thanks, Joe
>
> CLI -
>
> [Feb 11 17:45:25] VERBOSE[4526] logger.c:     -- Accepting overlap
> call from '0339' to '<unspecified>' on channel 0/31,
span 1
> [Feb 11 17:45:25] VERBOSE[5764] logger.c:     -- Starting simple
> switch on 'Zap/31-1'
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing
> [91905 at from-pstn:1] Set("Zap/31-1",
"__FROM_DID=91905") in new stack
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing
> [91905 at from-pstn:2] NoOp("Zap/31-1", "Received an unknown
call with
> DID set to 91905") in new stack
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing
> [91905 at from-pstn:3] Goto("Zap/31-1", "s|a2") in new
stack
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Goto (from-pstn,s,2)
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing
> [s at from-pstn:2] Answer("Zap/31-1", "") in new stack
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing
> [s at from-pstn:3] Wait("Zap/31-1", "2") in new stack
> [Feb 11 17:45:33] VERBOSE[5764] logger.c:     -- Executing
> [s at from-pstn:4] Playback("Zap/31-1", "ss-noservice")
in new stack
> [Feb 11 17:45:33] VERBOSE[5764] logger.c:     -- <Zap/31-1> Playing
> 'ss-noservice' (language 'en')
> [Feb 11 17:45:38] VERBOSE[5764] logger.c:     -- Executing
> [s at from-pstn:5] SayAlpha("Zap/31-1", "91905") in new
stack
> [Feb 11 17:45:38] VERBOSE[5764] logger.c:     -- <Zap/31-1> Playing
> 'digits/9' (language 'en')
> [Feb 11 17:45:39] VERBOSE[4526] logger.c:     -- Channel 0/31, span 1
> got hangup request, cause 16
> [Feb 11 17:45:39] WARNING[5764] file.c: Failed to write frame
> [Feb 11 17:45:39] VERBOSE[5764] logger.c:   == Spawn extension
> (from-pstn, s, 5) exited non-zero on 'Zap/31-1'
> [Feb 11 17:45:39] VERBOSE[5764] logger.c:     -- Executing
> [h at from-pstn:1] Hangup("Zap/31-1", "") in new stack
> [Feb 11 17:45:39] VERBOSE[5764] logger.c:   == Spawn extension
> (from-pstn, h, 1) exited non-zero on 'Zap/31-1'
> [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE,
> value: ON(1) on Zap/31-1
> [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Not yet hungup...  Calling
> hangup once with icause, and clearing call
> [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE,
> value: OFF(0) on Zap/31-1
> [Feb 11 17:45:39] VERBOSE[5764] logger.c:     -- Hungup 'Zap/31-1'
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

turby at canistec.com

2009-Feb-12 09:38 UTC

head link

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

I thing, you have bad routing configuration in extensions.conf. Send me 
"from-pstn" context configuration.

turby

joekane at gmail.com napsal(a):> Hi all,
>
> I have a connect between a siemens hipath & Asterisk system over PRI
> The connection works perfectly I can call from the Hipath to an 
> Asterisk Extension.
>
> I want to allow the hipath extensions to dial out over a SIP trunk on 
> asterisk but I keep getting "The number you have dialed is not in
service"
>
> In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk) 
> then the number 1905 (Freefone number in Ireland)
>
> Please help I cant figure this one out.
>
> Thanks, Joe
>
> CLI -
>
> [Feb 11 17:45:25] VERBOSE[4526] logger.c:     -- Accepting overlap 
> call from '0339' to '<unspecified>' on channel 0/31,
span 1
> [Feb 11 17:45:25] VERBOSE[5764] logger.c:     -- Starting simple 
> switch on 'Zap/31-1'
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing 
> [91905 at from-pstn:1] Set("Zap/31-1",
"__FROM_DID=91905") in new stack
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing 
> [91905 at from-pstn:2] NoOp("Zap/31-1", "Received an unknown
call with
> DID set to 91905") in new stack
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing 
> [91905 at from-pstn:3] Goto("Zap/31-1", "s|a2") in new
stack
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Goto (from-pstn,s,2)
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing 
> [s at from-pstn:2] Answer("Zap/31-1", "") in new stack
> [Feb 11 17:45:31] VERBOSE[5764] logger.c:     -- Executing 
> [s at from-pstn:3] Wait("Zap/31-1", "2") in new stack
> [Feb 11 17:45:33] VERBOSE[5764] logger.c:     -- Executing 
> [s at from-pstn:4] Playback("Zap/31-1", "ss-noservice")
in new stack
> [Feb 11 17:45:33] VERBOSE[5764] logger.c:     -- <Zap/31-1> Playing 
> 'ss-noservice' (language 'en')
> [Feb 11 17:45:38] VERBOSE[5764] logger.c:     -- Executing 
> [s at from-pstn:5] SayAlpha("Zap/31-1", "91905") in new
stack
> [Feb 11 17:45:38] VERBOSE[5764] logger.c:     -- <Zap/31-1> Playing 
> 'digits/9' (language 'en')
> [Feb 11 17:45:39] VERBOSE[4526] logger.c:     -- Channel 0/31, span 1 
> got hangup request, cause 16
> [Feb 11 17:45:39] WARNING[5764] file.c: Failed to write frame
> [Feb 11 17:45:39] VERBOSE[5764] logger.c:   == Spawn extension 
> (from-pstn, s, 5) exited non-zero on 'Zap/31-1'
> [Feb 11 17:45:39] VERBOSE[5764] logger.c:     -- Executing 
> [h at from-pstn:1] Hangup("Zap/31-1", "") in new stack
> [Feb 11 17:45:39] VERBOSE[5764] logger.c:   == Spawn extension 
> (from-pstn, h, 1) exited non-zero on 'Zap/31-1'
> [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, 
> value: ON(1) on Zap/31-1
> [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Not yet hungup...  Calling 
> hangup once with icause, and clearing call
> [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, 
> value: OFF(0) on Zap/31-1
> [Feb 11 17:45:39] VERBOSE[5764] logger.c:     -- Hungup 'Zap/31-1'
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

joekane at gmail.com

2009-Feb-13 09:21 UTC

head link

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

Default FreePBX context,

[from-pstn]
include => from-pstn-custom                     ; create this context in
extensions_custom.conf to include customizations
include => ext-did
include => ext-did-post-custom
include => from-did-direct        ; MODIFICATOIN (PL) for findmefollow if
enabled, should be bofore ext-local
include => ext-did-catchall        ; THIS MUST COME AFTER ext-did
exten => fax,1,Goto(ext-fax,in_fax,1)

The call seems to be going here

[ext-did-catchall]
include => ext-did-catchall-custom
exten => s,1,Noop(No DID or CID Match)
exten => s,n(a2),Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,SayAlpha(${FROM_DID})
exten => s,n,Hangup
exten => _.,1,Set(__FROM_DID=${EXTEN})
exten => _.,n,Noop(Received an unknown call with DID set to ${EXTEN})
exten => _.,n,Goto(s,a2)
exten => h,1,Hangup

; end of [ext-did-catchall]

------------------------------------------------------------------------------
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090213/2db8bbf0/attachment.htm

Philipp Kempgen

2009-Feb-14 16:58 UTC

head link

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

joekane at gmail.com schrieb:> Default FreePBX context,
> 
> [from-pstn]
> The call seems to be going here
> 
> [ext-did-catchall]
So?


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--

Marco Mouta

2009-Feb-15 13:26 UTC

head link

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

try to set in your zapata.conf

overlapdial=yes

then in your asterisk cli

reload chan_zap.so


--
Marco Mouta



On Fri, Feb 13, 2009 at 9:21 AM,  <joekane at gmail.com>
wrote:> Default FreePBX context,
>
> [from-pstn]
> include => from-pstn-custom                     ; create this context in
> extensions_custom.conf to include customizations
> include => ext-did
> include => ext-did-post-custom
> include => from-did-direct        ; MODIFICATOIN (PL) for findmefollow
if
> enabled, should be bofore ext-local
> include => ext-did-catchall        ; THIS MUST COME AFTER ext-did
> exten => fax,1,Goto(ext-fax,in_fax,1)
>
> The call seems to be going here
>
> [ext-did-catchall]
> include => ext-did-catchall-custom
> exten => s,1,Noop(No DID or CID Match)
> exten => s,n(a2),Answer
> exten => s,n,Wait(2)
> exten => s,n,Playback(ss-noservice)
> exten => s,n,SayAlpha(${FROM_DID})
> exten => s,n,Hangup
> exten => _.,1,Set(__FROM_DID=${EXTEN})
> exten => _.,n,Noop(Received an unknown call with DID set to ${EXTEN})
> exten => _.,n,Goto(s,a2)
> exten => h,1,Hangup
>
> ; end of [ext-did-catchall]
>
>
------------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

Apparently Analagous Threads

Search for more maybe matching threads

asterisk users - Feb 2009 - Siemens Hipath PRI to Asterisk Call Routing?

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

Apparently Analagous Threads