| View previous topic :: View next topic |
| Author |
Message |
tzieleniewski at gmail...
Guest
|
 Posted: Fri Jan 04, 2008 4:58 am Post subject: [asterisk-users] Unable to forward call on SIP channel after |
 |
|
Hi,
I have the following problem that when asterisk receives SIP response 302 it
cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to
create local channel for call forward to 'Local/poczta at routing-sip' (cause =
66)
What could be the reason for this?
Thank You in advance
bests
-tomasz
<--- SIP read from 192.168.0.165:7060 --->
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.0.165:7070;branch=z9hG4bK274b7d50;rport=7070
Contact: <sip:poczta at voip.rd.touk.pl:7170>
To: <sip:hellboy at voip.rd.touk.pl:7060>;tag=12711d6c
From: "IPFon"<sip:0225761853 at 192.168.0.165:7070>;tag=as66773d49
Call-ID: 08027b6d45abc7a00c2c6a8630d2de47 at 192.168.0.165
CSeq: 102 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
<------------->
[Jan 4 10:43:27] --- (9 headers 0 lines) ---
[Jan 4 10:43:27] -- Got SIP response 302 "Moved Temporarily" back from
192.168.0.165
[Jan 4 10:43:27] Transmitting (no NAT) to 192.168.0.165:7060:
ACK sip:hellboy at voip.rd.touk.pl:7060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.165:7070;branch=z9hG4bK274b7d50;rport
From: "IPFon" <sip:0225761853 at 192.168.0.165:7070>;tag=as66773d49
To: <sip:hellboy at voip.rd.touk.pl:7060>;tag=12711d6c
Contact: <sip:0225761853 at 192.168.0.165:7070>
Call-ID: 08027b6d45abc7a00c2c6a8630d2de47 at 192.168.0.165
Seq: 102 ACK
User-Agent: TouK S.K.A
Max-Forwards: 70
Content-Length: 0
---
[Jan 4 10:43:27] -- Now forwarding SIP/213.218.117.72-00837b20 to '
Local/poczta at routing-sip' (thanks to SIP/voip.rd.touk.pl-00843c10)
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to
create local channel for call forward to 'Local/poczta at routing-sip' (cause =
66)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080104/21305151/attachment.htm |
|
| Back to top |
|
 |
b.benchev at gmail.com
Guest
|
| Posted: Fri Jan 04, 2008 5:50 am Post subject: [asterisk-users] Unable to forward call on SIP channel after |
|
|
On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote:
| Quote: | Hi,
I have the following problem that when asterisk receives SIP response 302
it cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to
create local channel for call forward to 'Local/poczta at routing-sip' (cause
= 66)
|
Maybe this:
"Local channel
Description: Local Proxy Channel Driver
Syntax: Local/extension at context/n
Configuration file: none
chan_local is a pseudo-channel. Use of this channel simply loops calls back
into the dialplan in a different context. Useful for recursive routing.
Notes: Adding "/n" at the end of the string will make the Local channel not
do a native transfer (the "n" stands for "n"o release) upon the remote end
answering the line. This is an esoteric, but important feature if you
expect the Local channel to handle calls _exactly_ like a normal channel.
If you do not have the "no release" feature set, then as soon as the
destination (inside of the Local channel0 answers the line, the variables
and dial plan will revert back to that of the original call, and the Local
channel will become a zombie and be removed from the active channels list.
This is desirable in some circumstances, but can result in unexpected
dialplan behavior if you are doing fancy things with variables in your call
handling. "
Boyko |
|
| Back to top |
|
|
oej at edvina.net
Guest
|
| Posted: Fri Jan 04, 2008 7:06 am Post subject: [asterisk-users] Unable to forward call on SIP channel after |
|
|
4 jan 2008 kl. 11.50 skrev Benchev:
| Quote: | On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote:
| Quote: | Hi,
I have the following problem that when asterisk receives SIP
response 302
it cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No
channel
type registered for 'Local'
[Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer:
Unable to
create local channel for call forward to 'Local/poczta at routing-
sip' (cause
= 66)
|
Maybe this:
"Local channel
Description: Local Proxy Channel Driver
Syntax: Local/extension at context/n
Configuration file: none
chan_local is a pseudo-channel. Use of this channel simply loops
calls back
into the dialplan in a different context. Useful for recursive
routing.
|
You have to enable chan_local in menuselect (1.4) and make sure it's
not disabled
in modules.conf.
This is not a developer question, so please take this kind of
questions to
asterisk-users in the future. Thank you!
Best regards,
/Olle |
|
| Back to top |
|
|
oej at edvina.net
Guest
|
| Posted: Fri Jan 04, 2008 7:08 am Post subject: [asterisk-users] Unable to forward call on SIP channel after |
|
|
4 jan 2008 kl. 13.06 skrev Johansson Olle E:
| Quote: |
4 jan 2008 kl. 11.50 skrev Benchev:
| Quote: | On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote:
| Quote: | Hi,
I have the following problem that when asterisk receives SIP
response 302
it cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No
channel
type registered for 'Local'
[Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer:
Unable to
create local channel for call forward to 'Local/poczta at routing-
sip' (cause
= 66)
|
Maybe this:
"Local channel
Description: Local Proxy Channel Driver
Syntax: Local/extension at context/n
Configuration file: none
chan_local is a pseudo-channel. Use of this channel simply loops
calls back
into the dialplan in a different context. Useful for recursive
routing.
|
You have to enable chan_local in menuselect (1.4) and make sure it's
not disabled
in modules.conf.
This is not a developer question, so please take this kind of
questions to
asterisk-users in the future. Thank you!
|
My fault, really sorry. You were asking on -users!
Really time to get some energy. Lunch!
/Olle |
|
| Back to top |
|
|
tzieleniewski at gmail...
Guest
|
| Posted: Fri Jan 04, 2008 10:00 am Post subject: [asterisk-users] Unable to forward call on SIP channel after |
|
|
Thanks it helped, I had the noload in modules.conf.
But now I have another problem:
When 302 response is received by asterisk it falls in to some context.
according to rfc 3261 uac which receives 302 should retry the request at the
address given by the contact header filed.
I am not able to make the same routing decision because conditions are
different.
What can I do here.
I have for instance such problem that my asterisk works as a gateway.
When there is an external call this call is forwarded to some internal sip
address.
After this my sip client responses with 302 which point to his voicemail
(sip uri in the contact).
What can be done in such situation to make is work??
On Jan 4, 2008 1:06 PM, Johansson Olle E <oej at edvina.net> wrote:
| Quote: |
4 jan 2008 kl. 11.50 skrev Benchev:
| Quote: | On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote:
| Quote: | Hi,
I have the following problem that when asterisk receives SIP
response 302
it cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No
channel
type registered for 'Local'
[Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer:
Unable to
create local channel for call forward to 'Local/poczta at routing-
sip' (cause
= 66)
|
Maybe this:
"Local channel
Description: Local Proxy Channel Driver
Syntax: Local/extension at context/n
Configuration file: none
chan_local is a pseudo-channel. Use of this channel simply loops
calls back
into the dialplan in a different context. Useful for recursive
routing.
|
You have to enable chan_local in menuselect (1.4) and make sure it's
not disabled
in modules.conf.
This is not a developer question, so please take this kind of
questions to
asterisk-users in the future. Thank you!
Best regards,
/Olle
_______________________________________________
--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
|
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080104/8acedd68/attachment.htm |
|
| Back to top |
|
|
|
Search
