ISDN - Problem

[andy]

Hi Michael

After I found the solution how to get the miniPCI-ISDN-Card to work.. i've troubles with configuring it.

Hardware:
Alix-Board
OpenVox B100M miniPCI-ISDN-Card

Software:
Askozia 2.0 (Deciso-Image)

Provider:
Swisscom (ISDN-Basisanschluss)

Problem:
Can't make any outgoing calls (seel log below)
Can't make any incomming calls (Asterisk will not recognized the call)

After trying to make a call, i'm unable to edit/access the provider-page (accounts_providers.php) again. The page is loading, but no response/error.

Any idea what the problem is? Configuration Issue?

Thanks

Andy


Logfile:
May 27 08:19:23 asterisk[1048]: VERBOSE[1230]: -- Executing [0551231234@SIP-PHONE-17856380524bfd3d00f3efc:1] NoOp("SIP/101-00000000", "Direct dialing via SIP uri : 0551231234@192.168.3.91") in new stack
May 27 08:19:23 asterisk[1048]: VERBOSE[1230]: -- Executing [0551231234@SIP-PHONE-17856380524bfd3d00f3efc:2] Dial("SIP/101-00000000", "SIP/0551231234@192.168.3.91,,To") in new stack
May 27 08:19:23 asterisk[1048]: VERBOSE[1230]: == Using SIP RTP CoS mark 5
May 27 08:19:23 asterisk[1048]: VERBOSE[1230]: == Using SIP VRTP CoS mark 6
May 27 08:19:23 asterisk[1048]: VERBOSE[1230]: -- Called 0551231234@192.168.3.91
May 27 08:19:23 asterisk[1048]: VERBOSE[1060]: -- Got SIP response 482 "Loop Detected" back from 192.168.3.91
May 27 08:19:23 asterisk[1048]: VERBOSE[1230]: -- Now forwarding SIP/101-00000000 to 'Local/0551231234@public-direct-dial' (thanks to SIP/192.168.3.91-00000001)
May 27 08:19:23 asterisk[1048]: NOTICE[1230]: chan_local.c:550 in local_call: No such extension/context 0551231234@public-direct-dial while calling Local channel
May 27 08:19:23 asterisk[1048]: NOTICE[1230]: app_dial.c:571 in do_forward: Failed to dial on local channel for call forward to 'Local'
May 27 08:19:23 asterisk[1048]: VERBOSE[1230]: == Everyone is busy/congested at this time (1:0/0/1)

[Michael]

I'm not sure what's happening here. Can you reliably reproduce the problem?

[andy]

Yes, it's allways the same. After reboot, ... whatever.

> May 27 08:19:23 asterisk[1048]: VERBOSE[1060]: -- Got SIP response 482 "Loop Detected" back from 192.168.3.91

This will maybe happen because:
- I just have one Provider (ISDN Swisscom)
- In the Provider (ISDN Swisscom) the Provider (ISDN Swisscom) is specified as Failover

Without Failover, the logs reads:

May 28 07:43:17 asterisk[1048]: VERBOSE[14258]: -- Executing [0551231234@SIP-PHONE-17856380524bfd3d00f3efc:1] NoOp("SIP/101-00000006", "Direct dialing via SIP uri : 0551231234@192.168.3.91") in new stack
May 28 07:43:17 asterisk[1048]: VERBOSE[14258]: -- Executing [0551231234@SIP-PHONE-17856380524bfd3d00f3efc:2] Dial("SIP/101-00000006", "SIP/0551231234@192.168.3.91,,To") in new stack
May 28 07:43:17 asterisk[1048]: VERBOSE[14258]: == Using SIP RTP CoS mark 5
May 28 07:43:17 asterisk[1048]: VERBOSE[14258]: == Using SIP VRTP CoS mark 6
May 28 07:43:17 asterisk[1048]: VERBOSE[14258]: -- Called 0551231234@192.168.3.91
May 28 07:43:17 asterisk[1048]: VERBOSE[1060]: -- Got SIP response 482 "Loop Detected" back from 192.168.3.91
May 28 07:43:17 asterisk[1048]: VERBOSE[14258]: -- Now forwarding SIP/101-00000006 to 'Local/0551231234@public-direct-dial' (thanks to SIP/192.168.3.91-00000007)
May 28 07:43:17 asterisk[1048]: NOTICE[14258]: chan_local.c:550 in local_call: No such extension/context 0551231234@public-direct-dial while calling Local channel
May 28 07:43:17 asterisk[1048]: NOTICE[14258]: app_dial.c:571 in do_forward: Failed to dial on local channel for call forward to 'Local'
May 28 07:43:17 asterisk[1048]: VERBOSE[14258]: == Everyone is busy/congested at this time (1:0/0/1)
May 28 07:43:17 asterisk[1048]: VERBOSE[14258]: -- Auto fallthrough, channel 'SIP/101-00000006' status is 'CHANUNAVAIL'

If you have time, and if you want, I can provide remote-access to the device.

[Michael]

It doesn't look like your outgoing call is matching any patterns in that provider. What outgoing patterns do you have defined and what numbers are you dialing.

Having that provider be a failover for itself is a bad idea. I will remove that ability in the next release.

[andy]

Ouu.. you are right... i've chaned it.. because it didn't work...  below the correct error:  (Numer to dial: 0551231234, is replaced, but same format)

May 28 09:35:25 asterisk[1048]: VERBOSE[19630]: -- Executing [90551231234@SIP-PHONE-17856380524bfd3d00f3efc:1] NoOp("SIP/101-00000008", "outgoing call to provider: Test Linie") in new stack
May 28 09:35:25 asterisk[1048]: VERBOSE[19630]: -- Executing [90551231234@SIP-PHONE-17856380524bfd3d00f3efc:2] Set("SIP/101-00000008", "ORIGEXTENSION=90551231234") in new stack
May 28 09:35:25 asterisk[1048]: VERBOSE[19630]: -- Executing [90551231234@SIP-PHONE-17856380524bfd3d00f3efc:3] NoOp("SIP/101-00000008", "original extension = 90551231234") in new stack
May 28 09:35:25 asterisk[1048]: VERBOSE[19630]: -- Executing [90551231234@SIP-PHONE-17856380524bfd3d00f3efc:4] Dial("SIP/101-00000008", "DAHDI/g1/0551231234,,To") in new stack
May 28 09:35:25 asterisk[1048]: WARNING[19630]: app_dial.c:1547 in dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
May 28 09:35:25 asterisk[1048]: VERBOSE[19630]: == Everyone is busy/congested at this time (1:0/1/0)
May 28 09:35:25 asterisk[1048]: VERBOSE[19630]: -- Executing [90551231234@SIP-PHONE-17856380524bfd3d00f3efc:5] Hangup("SIP/101-00000008", "") in new stack
May 28 09:35:25 asterisk[1048]: VERBOSE[19630]: == Spawn extension (SIP-PHONE-17856380524bfd3d00f3efc, 90551231234, 5) exited non-zero on 'SIP/101-00000008'

[Michael]

Closed and continued at: http://forums.askozia.com/index.php/topic,1182.0.html