[asterisk-users] DAHDI congestion problem
I am unable to dial out over a Wildcard TDM400P. This was working previously, so must have messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX 2.5.2.2. When I dial, I see: -- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1, DAHDI/g0/9239220,300,) in new stack == Everyone is busy/congested at this time (1:0/1/0) It looks like the group is valid: dahdi show channels group g0 Chan Extension Context Language MOH InterpretBlocked State 4from-zaptel en default In Service The only Warning or Error I see is when asterisk first starts a new call. logger.c: -- Starting simple switch on 'DAHDI/1-1' [Sep 27 15:55:50] WARNING[4199] chan_dahdi.c: Unable to enable echo cancellation on channel 1 (No such device) On my TDM400P card, channel 1 is my analog phone, 2 my fax, and 4 the POTS line. More config files etc below. Any ideas? Thanks, Andy /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 10 22:20:05 2009 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) fxols=1 #echocanceller=mg2,1 fxols=2 #echocanceller=mg2,2 # channel 3, WCTDM/4/2, no module. fxsks=4 echocanceller=mg2,4 # Global data loadzone= us defaultzone = us dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) dahdi show channel 4 Channel: 4I File Descriptor: 20 Span: 1*CLI Extension: Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0I Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: yes Busy Count: 6 Busy Pattern: 0,0 TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI congestion problem
Andy Howell wrote: I am unable to dial out over a Wildcard TDM400P. This was working previously, so must have messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX 2.5.2.2. When I dial, I see: -- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1, DAHDI/g0/9239220,300,) in new stack == Everyone is busy/congested at this time (1:0/1/0) It looks like the group is valid: dahdi show channels group g0 Chan Extension Context Language MOH InterpretBlocked State 4from-zaptel en default In Service The only Warning or Error I see is when asterisk first starts a new call. logger.c: -- Starting simple switch on 'DAHDI/1-1' [Sep 27 15:55:50] WARNING[4199] chan_dahdi.c: Unable to enable echo cancellation on channel 1 (No such device) On my TDM400P card, channel 1 is my analog phone, 2 my fax, and 4 the POTS line. More config files etc below. Any ideas? Thanks, Andy /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 10 22:20:05 2009 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) fxols=1 #echocanceller=mg2,1 fxols=2 #echocanceller=mg2,2 # channel 3, WCTDM/4/2, no module. fxsks=4 echocanceller=mg2,4 # Global data loadzone= us defaultzone = us dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) dahdi show channel 4 Channel: 4I File Descriptor: 20 Span: 1*CLI Extension: Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0I Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: yes Busy Count: 6 Busy Pattern: 0,0 TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry to reply to my own post. I found that if I receive a call, I can then make outgoing calls until I reboot again. There must be something on my config that doesn't fully initialize the card. Once I've received a call, I can restart asterisk and it still works. Weird. Regards, Andy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P stops answering
I also see this from time to time. Running: Asterisk CVS-Nv1-0-9-09/02/05-14:07:10 built by [EMAIL PROTECTED] on a i686 running Linux Setup a little cron job that, in the wee hours, does a: service asterisk restart and then * will start answering the lines again. Not ideal, but.. Chris, Thanks. I saw another post mentioning that the top slot of the Dell Optiplex was not reliable to interupts. I kind of doubt that since there are just working off the same riser. This a low profile desk-top model. I switched it anyway. We'll see. If this does not do it, then I'll try the updating the zaptel code as Kevin suggested. There is also a utility called zttool. Looks like it should show me if there are missed interupts. Thanks, Andy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P stops answering
Kevin P. Fleming wrote: Andy Howell wrote: I have a weird problem in which my digium card stops answering. After running for a couple days, incoming calls are not seen. Running asterisk -r shows no incoming calls. Restarting Asterisk does not help. After a reboot it is fine. This problem was fixed in CVS (HEAD and v1-0) quite some time ago; what versions are you running? Its 1.0.9, as part of [EMAIL PROTECTED] 1.3 Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P stops answering
Leonardo Gomes Figueira wrote: Hi, Andy Howell wrote: I have a weird problem in which my digium card stops answering. After running for a couple days, incoming calls are not seen. Running asterisk -r shows no incoming calls. Restarting Asterisk does not help. After a reboot it is fine. Any ideas? Do you have APIC enabled on the BIOS/kernel ? Try to disable it on the BIOS or with noapic on the kernel. I found out this was the cause of this problem here on a VIA motherboard and it was fixed with noapic. I just don't know why... :) Leonardo, I have it APIC disabled. I thought that interupts might be the problem from reading the voip wiki. I had the card in another machine, where I first noticed the problem. After lots of messing around, I decided to go with a machine that others said worked well. I'm now running on a Dell Optiplex GX150 with 1Ghz CPU and 512MB of memory. The machine is dedicated to asterisk. At boot, the card is reported as: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) I suppose I could just reboot nightly. Trouble is, there is no way to detect that it is not working, other than trying to call in. Outgoing calls continue to work. Thanks, Andy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P stops answering
I have a weird problem in which my digium card stops answering. After running for a couple days, incoming calls are not seen. Running asterisk -r shows no incoming calls. Restarting Asterisk does not help. After a reboot it is fine. Any ideas? Thanks, Andy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] trouble using variables with included contexts
I'm trying to set a variable in one context and use it in another:
[c1]
MY_GROUP = SIP/200SIP/201
[c2]
include c1
exten s,1,NoOp(${MY_GROUP})
The noop prints out a blank string. In [c1], I've tried
exten s,1,SetVar(MY_GROUP=SIP/200SIP/201)
and
exten s,1,SetVar(_MY_GROUP=SIP/200SIP/201)
and
exten s,1,SetVar(__MY_GROUP=SIP/200SIP/201)
and
exten s,1,SetGlobalVar(MY_GROUP=SIP/200SIP/201)
Nothing set in [c1] works. If in [c2] I do:
exten s,1,SetVar(MY_GROUP=SIP/200SIP/201)
exten s,2,NoOp(${MY_GROUP})
Then that works. From reading
http://www.voip-info.org/tiki-index.php?page=Asterisk%20variables
it seems like what I'm trying to do should work.
I using Asterisk 1.09. Putting in the MY_GROUP = SIP... in [globals]
section works, but that gets overwritten by AMP, so its not an option.
Any ideas what I'm doing wrong?
Thanks.
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Re: [Asterisk-Users] Can't initiate a call with X-Lite.
Rich Adamson wrote: It is just sending a sip invite to [EMAIL PROTECTED] Does the X-Lite need to connect to via a proxy? No. You should work on configuring xlite to register with asterisk. Thanks. I can get it to work that way. What I was trying to simulate was an external user calling in. Sorry, I should have stated that. From your asterisk CLI, try sip debug to see the flow of packets to/from asterisk; sip no debug will shut it off. That is just what I needed. I found that asterisk is looking in the 'default' context for the extension, whereas our extensions are under [from-sip]. I've got a little more configuring to do. Thanks. Andy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't initiate a call with X-Lite.
Hello, I'm trying to place a call to asterisk using X-Lite. Asterisk is setup with some Grandstream phones. I can call from one grandstream extension to another. When I try to an extension with X-Lite, it comes back with Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk extension. It is just sending a sip invite to [EMAIL PROTECTED] Does the X-Lite need to connect to via a proxy? After several days of reading RFCs and looking at packet traces, I know a bit more about SIP, but not quite enough to make this work. Is there a way to get asterisk to say what its doing? I tried -vv etc, but the only messages are see are when I use one of my my Grandstream phones. On the wire, is see the same To: header from both the grandstream and the X-Lite soft phone. I don't understand why its found by one, and not the other. Thanks, Andy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
