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Home > Archive > Voice over IP Cisco > January 2007 > SIP /SCCP call forwarding
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SIP /SCCP call forwarding
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| Ahmad Cheikh-Moussa 2007-01-28, 1:11 pm |
| Hi Guys, =
I have a strange problem. Perhaps someone had a similar problem and =
could solve it. I've got an asterisk server, a H.323 gateway and =
a 7940 SCCP Phone. The callmanager version is 4.1.3(Sr1).
Asterisk mailbox : 111
7940 : 222
ISDN30-Number : 444.
For example if someone from the pstn wants to call the =
7940, then the user calls 444-222.
The Problem is, when I make a callforward all on the 7940 to
the asterisk number 111, all calls from external doesn't work.
If I make the callforward all to 0444-111, then the callforward works.
Did anyone experienced such a problem. Here a list of calls, which works.
PSTN -> mailbox (444-111) =3D works
PSTN -> 7940 (444-222) =3D works
7940 -> mailbox (111) =3D works
7940 -> mailbox (0444-111) =3D works
PSTN -> 7940 (444-222) call forward all is set to 111 =3D doesn't work
PSTN -> 7940 (444-222) call forward all is set to 0444-111 =3D work
Here a cut of my config:
voice call convert-discpi-to-prog
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
sip
!
dial-peer voice 800 voip
destination-pattern 111
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
sip-ua
set sip-status 401 pstn-cause 127
set sip-status 407 pstn-cause 127
set sip-status 410 pstn-cause 22
set sip-status 415 pstn-cause 127
set sip-status 480 pstn-cause 19
set sip-status 503 pstn-cause 127
set sip-status 580 pstn-cause 127
retry invite 3
retry register 3
timers register 150
registrar ipv4:1.2.3.51 expires 3600
sip-server ipv4:1.2.3.51
!
During the debugging, I could see by the call, which doesn't work,
that the H.323 Gateway send a cancel message and the call is terminated.
1. solero01 -> astvoice: INVITE
2. astvoice -> solero01: SIP/2.0 100 Trying
3. astvoice -> solero01: SIP/2.0 200 OK
4. solero01 -> astvoice: CANCEL sip:111@1.2.3.51:5060 SIP/2.0
5. astvoice -> solero01: SIP/2.0 487 Request Terminated
6. astvoice -> solero01: SIP/2.0 200 OK [...] CSeq: 101 CANCEL
7. solero01 -> astvoice: ACK sip:111@1.2.3.51:5060 SIP/2.0
I made the same debugging on the H.323 Gateway, but I could not see a reason
for the cancel. =
Regards,
Ahmad
-- =
Ahmad Cheikh-Moussa =
NetUSE AG
Dr.-Hell-Stra=DFe, 24107 Kiel, Germany
Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499
Service: Service@NetUSE.DE -- http://NetUSE.DE/
| |
| Vince Loschiavo 2007-01-28, 1:11 pm |
| I think you'll need a media termination point for doing forwards, transfers=
, etc with h323. =
Regards,
Vince Loschiavo
Senior cisco Engineer
DATACORP
7862821164
Sent via blackberry from T-Mobile =
-----Original Message-----
From: Ahmad Cheikh-Moussa <acm@netuse.de>
Date: Sun, 28 Jan 2007 15:26:38 =
To:cisco-voip@puck.nether.net
Subject: [cisco-voip] SIP /SCCP call forwarding
Hi Guys, =
I have a strange problem. Perhaps someone had a similar problem and =
could solve it. I've got an asterisk server, a H.323 gateway and =
a 7940 SCCP Phone. The callmanager version is 4.1.3(Sr1).
Asterisk mailbox : 111
7940 : 222
ISDN30-Number : 444.
For example if someone from the pstn wants to call the =
7940, then the user calls 444-222.
The Problem is, when I make a callforward all on the 7940 to
the asterisk number 111, all calls from external doesn't work.
If I make the callforward all to 0444-111, then the callforward works.
Did anyone experienced such a problem. Here a list of calls, which works.
PSTN -> mailbox (444-111) =3D works
PSTN -> 7940 (444-222) =3D works
7940 -> mailbox (111) =3D works
7940 -> mailbox (0444-111) =3D works
PSTN -> 7940 (444-222) call forward all is set to 111 =3D doesn't work
PSTN -> 7940 (444-222) call forward all is set to 0444-111 =3D work
Here a cut of my config:
voice call convert-discpi-to-prog
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
sip
!
dial-peer voice 800 voip
destination-pattern 111
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
sip-ua
set sip-status 401 pstn-cause 127
set sip-status 407 pstn-cause 127
set sip-status 410 pstn-cause 22
set sip-status 415 pstn-cause 127
set sip-status 480 pstn-cause 19
set sip-status 503 pstn-cause 127
set sip-status 580 pstn-cause 127
retry invite 3
retry register 3
timers register 150
registrar ipv4:1.2.3.51 expires 3600
sip-server ipv4:1.2.3.51
!
During the debugging, I could see by the call, which doesn't work,
that the H.323 Gateway send a cancel message and the call is terminated.
1. solero01 -> astvoice: INVITE
2. astvoice -> solero01: SIP/2.0 100 Trying
3. astvoice -> solero01: SIP/2.0 200 OK
4. solero01 -> astvoice: CANCEL sip:111@1.2.3.51:5060 SIP/2.0
5. astvoice -> solero01: SIP/2.0 487 Request Terminated
6. astvoice -> solero01: SIP/2.0 200 OK [...] CSeq: 101 CANCEL
7. solero01 -> astvoice: ACK sip:111@1.2.3.51:5060 SIP/2.0
I made the same debugging on the H.323 Gateway, but I could not see a reason
for the cancel. =
Regards,
Ahmad
-- =
Ahmad Cheikh-Moussa =
NetUSE AG
Dr.-Hell-Stra=DFe, 24107 Kiel, Germany
Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499
Service: Service@NetUSE.DE -- http://NetUSE.DE/
________________________________________
_______
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
| |
| Ahmad Cheikh Moussa 2007-01-29, 1:11 pm |
| Hi Vince,
Vince Loschiavo wrote:
> I think you'll need a media termination point for doing forwards, transfe=
rs, etc with h323. =
> =
why do I need a media termination point. When I make a call from the
7940 to the asterisk voicebox, it works without a problem.
Where is the difference between a direct call and a forwarded call ?
Regards,
Ahmad
-- =
Ahmad Cheikh-Moussa
NetUSE AG
Dr.-Hell-Stra=DFe, 24107 Kiel, Germany
Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499
Service: Service@NetUSE.DE -- http://NetUSE.DE/
| |
| Matt Slaga \(US\) 2007-01-29, 1:11 pm |
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