Outgoing calls blocked after reboot with analog PCI Card A400M13

[jstrebel]

Hi,
I am using A400M13 Mini PCI Cards. The card worked great under 1.0.3
After upgrading to 2.0.0 I have the following problem: (Akozia in running under PCengine embedded version)

I cant place a outgoing call after rebbot. First I need to place a incomming call from the PSTN Netork to Askozia (call my askozia with my mobile) and from then on, I can place also outgoing calls.
Let me know what kind of debuging info I need to provide you. Below is the log with default verbosity
Best regards Jakob

May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: -- Executing [079…….@SIP-PHONE-1341818641386d4b8327576:1] NoOp("SIP/100-00000000", "outgoing call to provider: swisscom") in new stack
May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: -- Executing [079…….@SIP-PHONE-1341818641386d4b8327576:2] Set("SIP/100-00000000", "ORIGEXTENSION=079…….") in new stack
May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: -- Executing [079…….@SIP-PHONE-1341818641386d4b8327576:3] NoOp("SIP/100-00000000", "original extension = 079…….") in new stack
May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: -- Executing [079…….@SIP-PHONE-1341818641386d4b8327576:4] Dial("SIP/100-00000000", "DAHDI/g1/0794003384,,To") in new stack
May 14 10:09:13 asterisk[1320]: WARNING[1348]: app_dial.c:1547 in dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: == Everyone is busy/congested at this time (1:0/1/0)
May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: -- Executing [079…….@SIP-PHONE-1341818641386d4b8327576:5] Hangup("SIP/100-00000000", "") in new stack
May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: == Spawn extension (SIP-PHONE-1341818641386d4b8327576, 079……., 5) exited non-zero on 'SIP/100-00000000'
May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: -- Executing [h@SIP-PHONE-1341818641386d4b8327576:1] NoOp("SIP/100-00000000", "outgoing call to provider: swisscom") in new stack
May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: -- Executing [h@SIP-PHONE-1341818641386d4b8327576:2] Set("SIP/100-00000000", "ORIGEXTENSION=h") in new stack
May 14 10:09:13 asterisk[1320]: VERBOSE[1348]: -- Executing [h@SIP-PHONE-1341818641386d4b8327576:3] NoOp("SIP/100-00000000", "original extension = h") in new stack
May 14 10:10:22 asterisk[1320]: WARNING[1317]: asterisk.c:2903 in canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are no
May 14 10:10:35 asterisk[1320]: VERBOSE[1349]: -- Starting simple switch on 'DAHDI/1-1'
May 14 10:10:37 asterisk[1320]: NOTICE[1349]: callerid.c:665 in callerid_feed: Unknown IE 48
May 14 10:10:37 asterisk[1320]: VERBOSE[1349]: -- Executing [s@ANALOG-PROVIDER-18150250734be4635fc2802-incoming:1] NoOp("DAHDI/1-1", "incoming call from provider: swisscom to s") in new stack

[peppie]

I confirm

exactly the same problem here

[Michael]

I haven't run into this myself. Can you reliably recreate it?

[jstrebel]

Michael, it happen up to now every time I started akskozia. Tried 4 times and it happened 4 times. Let me know what kind of debugging you need.
Jakob

[peppie]

Hello Michael,

I also tried this a couple of times and it always happens after reboot, also after power loss
Outbound calls trough my sip provider are working, I just cannot make outbound calls trough analog card, not until called inbound with my cell phone. after that everyting works perfect. I make outbound calls with my analog phone only, there is no ip phone installed.
I use the generic image on a wyse thin client and wrote the image with physdiskwrite, hopefully this information is of any use.

regards Patrick

May 30 08:37:20 asterisk[1385]: VERBOSE[2069]: -- Starting simple switch on 'DAHDI/1-1'
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: -- Executing [0631004964@ANALOG-PHONE-3038494704bfc10dea47bb:1] NoOp("DAHDI/1-1", "outgoing call to provider: ziggo") in new stack
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: -- Executing [0631004964@ANALOG-PHONE-3038494704bfc10dea47bb:2] Set("DAHDI/1-1", "ORIGEXTENSION=0631004964") in new stack
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: -- Executing [0631004964@ANALOG-PHONE-3038494704bfc10dea47bb:3] NoOp("DAHDI/1-1", "original extension = 0631004964") in new stack
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: -- Executing [0631004964@ANALOG-PHONE-3038494704bfc10dea47bb:4] Dial("DAHDI/1-1", "DAHDI/g2/0631004964,,To") in new stack
May 30 08:37:29 asterisk[1385]: WARNING[2069]: app_dial.c:1547 in dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: == Everyone is busy/congested at this time (1:0/1/0)
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: -- Executing [0631004964@ANALOG-PHONE-3038494704bfc10dea47bb:5] NoOp("DAHDI/1-1", "DIALSTATUS = CONGESTION") in new stack
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: -- Executing [0631004964@ANALOG-PHONE-3038494704bfc10dea47bb:6] Goto("DAHDI/1-1", "ANALOG-PROVIDER-14373634304bfc11e93694b-CONGESTION,1") in new stack
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: -- Goto (ANALOG-PHONE-3038494704bfc10dea47bb,ANALOG-PROVIDER-14373634304bfc11e93694b-CONGESTION,1)
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: -- Executing [ANALOG-PROVIDER-14373634304bfc11e93694b-CONGESTION@ANALOG-PHONE-3038494704bfc10dea47bb:1] NoOp("DAHDI/1-1", "Failover needed, now using : voipdiscount") in new stack
May 30 08:37:29 asterisk[1385]: VERBOSE[2069]: -- Executing [ANALOG-PROVIDER-14373634304bfc11e93694b-CONGESTION@ANALOG-PHONE-3038494704bfc10dea47bb:2] Goto("DAHDI/1-1", "SIP-PROVIDER-19884083824bfc124373c75,0631004964,1") in new stack

[DrCain]

Same here. I have to call into the pbx, and it works again.
I'm using an OpenVOX TDM400p

[Michael]

I will contact OpenVox to see if they have seen similar issues. I was able to reproduce this problem but haven't had a chance to dive into it. At least I have a reference system now to work from. If I need any further debugging information, I'll post again. Thanks for letting me know about this. It's a really nasty bug.

[Michael]

I've found the issue, thanks to Jos at Deciso. There is an issue with the analog driver in the 2.0.0 release. This will be fixed in 2.0.1:
 - https://issues.asterisk.org/view.php?id=14577

[Michael]

I'm going to prepare a developer snapshot with this fix. Who can test it and what platform do you need?

[jstrebel]

Michael.
I run it on PCengines ALIX2d1 (embedded plattform) I like to test it.
Jakob

[Michael]

Here it is, thanks for volunteering!
 - http://downloads.askozia.com/tmp/askozia-pbx-embedded-x86-i486-uclibc-r1567.img

All of the changes in this image are:
 - updated German translation
 - updated Danish translation
 - updated Polish translation
 - fixed bug when changing the administrator username
 - option to add new isdn phone account only present if an appropriate port is available
 - reboots not required as often after General Setup page changes
 - translation completion percentages now displayed next to webgui languages
 - fixed outgoing calls to Analog providers after reboots

[andy]

All of the changes in this image are:

- VirtualFax?

[Michael]

Virtual Fax is in trunk, not the 2.0 branch. The 2.0 branch was created when 2.0.0 was released and will be used to maintain 2.0.1, 2.0.2, 2.0.x releases. The trunk of AskoziaPBX is now being used to work on the upcoming 2.1 releases.

[peppie]

Michael

I can test the snaphot as well, i use wyse thin client, generic image

Patrick

[jstrebel]

Michael, it seems the Problem "outgoing calls after reboot" is solved. (10 reboots no problem)
PCengines ALIX2 Image embedded.
Thanks jakob