Re: [asterisk-users] undefined symbol: cap_set_proc on several modules after installation from source
On Thu, May 12, 2011 at 09:44:31PM -0400, Jose P. Espinal wrote: Hello Folks, What could be producing the following warnings on console, after an installation from source (Asterisk 1.4.41): [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'res_musiconhold.so': /usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol: cap_set_proc This one should come from libcap. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 realtime tables.
I was looking for MySQL table structures for ARA (Asterisk 1.8.X). I found one for SIP friends on, https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure But it seems that it is not as per the Asterisk 1.8 SIP options. i.e. it contains 'call-limit' which is deprecated in 1.8 and not the 'callcounter' as one of the fields. Pardon my ignorance, but are 'cid_number','trunkname','fullname' from given link sip parameters to be set? sip.conf has no such entries. I also looked at .../contrib/realtime/mysql, and didn't find 'callcounter' in sipfriends.sql. I also couldn't find tables for queue and queue members in it. Yes, I can update or add latest options(fields) in sipfriends.sql. I can even create structures for queue and queue members.I just wanted to come across the latest table definitions so that I don't spend time on reinventing the wheels. Will it not be a good option to have latest and all '.sql's for ARA in contrib/realtime/mysql? I appreciate the kind of work Digium has done in the form of Asterisk. I also acknowledge the kind of (indirect :)) help I have got from Asterisk users' group. Thanks a lot. [SATISH] -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Light indicator managed by Asterisk
Probably using XML - which is phone dependant. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 12 May 2011 21:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk On 05/12/2011 07:12 PM, Carlos Chavez wrote: On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. BLF lights can be manipulated with Hints and the DEVSTATE function to set custom device states. This way you can have a BLF light react to any event you want. Hello, I must say that I have succeeded in working with DEVSTATE to get a BLF-light in several colors. Which works great for what I want. Thank you for the feedback. Do you think it is also possible to get info displayed on the screen of the IP-phone ? Any idea how that would work ? Something tells me that this will depend on the brand/type of IP-phone. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Light indicator managed by Asterisk
Cor-wrong (sort of). There is a backport of DevState/Device_State for 1.4 https://issues.asterisk.org/view.php?id=15818 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: 12 May 2011 20:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk Correct. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, May 12, 2011 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk Eric Wieling wrote: pbx*CLI core show application minivmmwi Core show application minivmmwi core show function DEVICE_STATE Both of these must be a 1.6.x or newer, I have neither under 1.4 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Light indicator managed by Asterisk
Andrew Thomas wrote: Cor-wrong (sort of). There is a backport of DevState/Device_State for 1.4 https://issues.asterisk.org/view.php?id=15818 Very cool! I'll have to review this weekend. Thank you! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 + google voice
Im running v1.8.2.3 and not have no had this issue you speak of? I saw it once or twice, but otherwise, it works. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeremy Kister Sent: Thursday, May 12, 2011 11:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 1.8 + google voice On 5/12/2011 11:08 PM, Jeremy Kister wrote: [May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to establish the call anyway I found the problem, and I am sending in a bug report :) if anyone is interested, the issue is 19286 (i'll be completing it shortly) -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php? id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list
[asterisk-users] DAHDI Error
Hi, Sometimes calls on Asterisk fail to connect to DAHDI channels and giving below error: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) There are 8 E1 connected on server and only 15-20 simultaneous calls. All channels and E1 are showing in service without any alarms. Could anyone please let me know why this is happening? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Error
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Friday, May 13, 2011 9:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI Error Hi, Sometimes calls on Asterisk fail to connect to DAHDI channels and giving below error: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) There are 8 E1 connected on server and only 15-20 simultaneous calls. All channels and E1 are showing in service without any alarms. Could anyone please let me know why this is happening? The message is likely coming from the telco or from the destination number. It is a common issue. I usually put something in my dialplan to retry all calls that receive an unexpected hangup cause to work around the telco seemingly randomly sending back odd hangup causes. You should not retry ALL calls, only ones with unexpected hangup causes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] undefined symbol: cap_set_proc on several modules after installation from source
Tzafrir Cohen wrote: [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'res_musiconhold.so': /usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol: cap_set_proc Could this be related to having used 'strip' on the binaries? Note: I have previously compiled/installed Asterisk 1.4.X on a fresh install of Slackware 13.X and never faced this before. -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Error
On 13 May 2011 14:06, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Friday, May 13, 2011 9:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI Error Hi, Sometimes calls on Asterisk fail to connect to DAHDI channels and giving below error: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) There are 8 E1 connected on server and only 15-20 simultaneous calls. All channels and E1 are showing in service without any alarms. Could anyone please let me know why this is happening? The message is likely coming from the telco or from the destination number. It is a common issue. I usually put something in my dialplan to retry all calls that receive an unexpected hangup cause to work around the telco seemingly randomly sending back odd hangup causes. You should not retry ALL calls, only ones with unexpected hangup causes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can’t dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can’t dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
[asterisk-users] Unknown Agent Status on DAHDI
Hi Guys: I am very new in Asterisk Queue, so may be i'm doing wrong somewhere. I have Asterisk 1.8.3.3 and Dahdi 2.4.1.2. I defined some agent's on Asterisk Queue, and the problem is that the agent is allways on UNKNOWN status, so Asterisk can dial to the agent even if the agent is allready busy. No matter if the agent is dynamic, realtime or static. I tried with sip channels and there where no problems, the problem is only with dahdi. Do you have any tips for this issue?. Sorry if i am the wrong list. thanks in advance. rv -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Backport of DEVICE_STATE to 1.4
Hi, Here http://www.voip-info.org/wiki/view/Asterisk+func+device_State you can find a link to download a backported for Asterisk 1.4 version of DEVICE_STATE function. (Elsewhere, you can find reference to another backported function DEVSTATE which seems to behave the same as DEVICE_STATE). As I would like to prepare as much as possible, my dialplan to 1.6 and beyond, I would prefer to use DEVICE_STATE if possible. Anyway, a quick inside this fucn_devstate.c file shows that some (all ?) Log or Error messages are still refering to DEVSTATE. My question is which is the best source to get DEVICE_STATE function for Asterisk 1.4 ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outbound calls via google voice not answered by toll free numbers with ivrs
Hi All, I'm using Asterisk 1.8.2 with outbound calls using Google Voice. I've been having issues calling several toll free numbers where the call 'is ringing' but never transitions to 'answered'. These are toll free numbers which are typically answered by an ivrs where you enter eg. a conference bridge number. I searched google and the closest reported issues I found are - https://issues.asterisk.org/view.php?id=18319 (on 1.6.x) and https://issues.asterisk.org/view.php?id=5266 (where the ibm support number listed does not work for my setup either) The number in the second ticket can be used as a test case - 800-426-7378 - and I'm hoping someone has run into this before. I have already tried both 'auto' and 'rfc2833' settings for dtmfmode and can provide any additional details about my setup. Thanks in advance! -Gaurav -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ --
Re: [asterisk-users] DAHDI Error
I didn't understand very well.. So you cant dial on the first 24 channels? Did you take care on the jumper of the card?. There is something related to E1 (31 channels) or T1 (24 channels). And check the system.conf either. rv 2011/5/13 deeps backup backup.de...@gmail.com I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can’t dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can’t dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? On 13 May 2011 15:07, deeps backup backup.de...@gmail.com wrote: On 13 May 2011 14:06, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Friday, May 13, 2011 9:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI Error Hi, Sometimes calls on Asterisk fail to connect to DAHDI channels and giving below error: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) There are 8 E1 connected on server and only 15-20 simultaneous calls. All channels and E1 are showing in service without any alarms. Could anyone please let me know why this is happening? The message is likely coming from the telco or from the destination number. It is a common issue. I usually put something in my dialplan to retry all calls that receive an unexpected hangup cause to work around the telco seemingly randomly sending back odd hangup causes. You should not retry ALL calls, only ones with unexpected hangup causes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can’t dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can’t dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Error
Show us a pri debug of a problem call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Friday, May 13, 2011 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI Error I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can't dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can't dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? On 13 May 2011 15:07, deeps backup backup.de...@gmail.com wrote: On 13 May 2011 14:06, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Friday, May 13, 2011 9:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI Error Hi, Sometimes calls on Asterisk fail to connect to DAHDI channels and giving below error: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) There are 8 E1 connected on server and only 15-20 simultaneous calls. All channels and E1 are showing in service without any alarms. Could anyone please let me know why this is happening? The message is likely coming from the telco or from the destination number. It is a common issue. I usually put something in my dialplan to retry all calls that receive an unexpected hangup cause to work around the telco seemingly randomly sending back odd hangup causes. You should not retry ALL calls, only ones with unexpected hangup causes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can't dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can't dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so
Sorry for top post I'm responding from my blackberry I haven't tried with timerfd but with timer pthread 1.8 is very unstable I think I have seen a post to the list from kevin fleming that the same is for timerfd that there is a nasty bug which they haven't found the reason for yet -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 13 May 2011 15:17:24 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
+1 /@ \ \ ___ \ (__O) \ (@) \ (@) \ (__o)_\ \\ On Tue, May 10, 2011 at 4:23 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: I'll keep this brief because I don't want to come across like any more of an a$$ than I absolutely have to, especially since I know I've blown my stack before. Gentlemen (and Ladies, if you're out there), If someone gives you advice on this list, and ESPECIALLY if they give you advice offlist, have the courtesy to (AT THE LEAST) to let them know when/if you get your question answered or your problem solved. As many people point out, on community supported mailing lists and forums around the world, these user lists are comprised of people who are giving their time freely to help others learn about the software the list is about. Sometimes those lists are about software that is quite useful in a commercial setting, perhaps even very much in demand, like Asterisk. Now, you should always appreciate when you get assistance from people on user lists, but when you're asking for help on a list like this one, (where I'd say 80% of the participants on the list are professionals who earn their living by selling their knowledge of how to install, configure, and maintain a server application like Asterisk) it would be extremely appreciated if you show some courtesy to the individual(s) who assisted you for free. I've had several individuals contact me offlist (without being given permission first, which is first and foremost bad form) and ask for my assistance with configuring a feature, troubleshooting an issue, and once I got an email that said something along the lines of: I saw a post on the list where you said you could accomplish insertNiftyFeatureThatDidNotPreviouslyExistHere Tell me how to do it I'm sure many of you have been the recipient of more than your fair share of emails offlist asking for help, and I'm sure a great number of you try to offer assistance. What is bothering me is the fact there seems to be a new trend forming, wherein I don't get a repsonse from the person I tried to help, even when I can feel confident in saying that I know I gave them the piece of information they needed in order to answer their question and accomplish the goal of making Asterisk perform the way they wanted. Has anyone else noticed this trend? Those of you who are making the requests, is there a reason why you don't feel the need to be courteous and at least say, Hey that advice worked, everything's working now? Next time you ask for help, especially when it's offlist (and even MORE SO when you're contacting someone you weren't invited to contact offlist), I want you to remember that the person you're contacting usually gets paid for their time as an Asterisk professional, and that they're helping you for free. Hell, if you want to get down to brass tacks about it, thatr person who is taking the time to try and help you is increasing his or her own professional competition.. that's all...nothing super rude, but I had to get that one out there I get annoyed when I answer about 12-13 questions (all in separate emails, mind you) from someone, and then I never get even find out if I was successful in helping them -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Error
I can dial 1-24 channels but not after that. There are 8 E1s. Box was working fine and carrying traffic on all E1s before. Just recently i noticed this problem has occurred. On 13 May 2011 16:30, Rafael Visser visser.raf...@gmail.com wrote: I didn't understand very well.. So you cant dial on the first 24 channels? Did you take care on the jumper of the card?. There is something related to E1 (31 channels) or T1 (24 channels). And check the system.conf either. rv 2011/5/13 deeps backup backup.de...@gmail.com I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can’t dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can’t dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? On 13 May 2011 15:07, deeps backup backup.de...@gmail.com wrote: On 13 May 2011 14:06, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Friday, May 13, 2011 9:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI Error Hi, Sometimes calls on Asterisk fail to connect to DAHDI channels and giving below error: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) There are 8 E1 connected on server and only 15-20 simultaneous calls. All channels and E1 are showing in service without any alarms. Could anyone please let me know why this is happening? The message is likely coming from the telco or from the destination number. It is a common issue. I usually put something in my dialplan to retry all calls that receive an unexpected hangup cause to work around the telco seemingly randomly sending back odd hangup causes. You should not retry ALL calls, only ones with unexpected hangup causes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can’t dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can’t dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
sangoma cards do not use dahdi... 13.5.2011 v 17:16, satish patel satish...@hotmail.com: Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ? campbx1*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 res_timing_dahdi.soDAHDI Timing Interface 0 3 modules loaded From: n...@njcolledge.net To: asterisk-users@lists.digium.com Date: Fri, 13 May 2011 15:11:19 + Subject: Re: [asterisk-users] 1.8 and prematuremedia problem At the asterisk CLI type “module show like timing” Whichever has a use-count 1 is the one you are using. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 13 May 2011 16:03 To: tbs...@gmail.com; asterisk-users Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not
Re: [asterisk-users] 1.8 and prematuremedia problem
You mean say i don't use res_timing_dahdi.so ? I guess this is just timing module nothing related to Card. _S From: tu...@canistec.com Date: Fri, 13 May 2011 18:30:52 +0200 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem sangoma cards do not use dahdi... 13.5.2011 v 17:16, satish patel satish...@hotmail.com: Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ? campbx1*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 res_timing_dahdi.soDAHDI Timing Interface 0 3 modules loaded From: n...@njcolledge.net To: asterisk-users@lists.digium.com Date: Fri, 13 May 2011 15:11:19 + Subject: Re: [asterisk-users] 1.8 and prematuremedia problem At the asterisk CLI type “module show like timing” Whichever has a use-count 1 is the one you are using. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 13 May 2011 16:03 To: tbs...@gmail.com; asterisk-users Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also
Re: [asterisk-users] Light indicator managed by Asterisk
On Thu, 2011-05-12 at 22:17 +0200, Jonas Kellens wrote: On 05/12/2011 07:12 PM, Carlos Chavez wrote: On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. BLF lights can be manipulated with Hints and the DEVSTATE function to set custom device states. This way you can have a BLF light react to any event you want. Hello, I must say that I have succeeded in working with DEVSTATE to get a BLF-light in several colors. Which works great for what I want. Thank you for the feedback. Do you think it is also possible to get info displayed on the screen of the IP-phone ? Any idea how that would work ? Something tells me that this will depend on the brand/type of IP-phone. Aastra phones have a very good XML browser. You can use an API to send messages and configuration to the phone. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6: Custom Name for Recordings file
Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6: Custom Name for Recordings file
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
See the fname_base information below.
pbx*CLI core show application monitor
-= Info about application 'Monitor' =-
[Synopsis]
Monitor a channel.
[Description]
Used to start monitoring a channel. The channel's input and output voice
packets are logged to files until the channel hangs up or monitoring is stopped
by the StopMonitor application.
By default, files are stored to /var/spool/asterisk/monitor/. Returns
'-1' if monitor files can't be opened or if the channel is already monitored,
otherwise '0'.
[Syntax]
Monitor([file_format[:urlbase]][,fname_base[,options]])
[Arguments]
file_format
optional, if not set, defaults to 'wav'
fname_base
if set, changes the filename used to the one specified.
options
m: when the recording ends mix the two leg files into one and delete
the two leg files. If the variable ${MONITOR_EXEC} is set, the application
referenced in it will be executed instead of soxmix/sox and the raw leg
files will NOT be deleted automatically. soxmix/sox or ${MONITOR_EXEC}
is handed 3 arguments, the two leg files and a target mixed file name
which is the same as the leg file names only without the in/out designator.
If ${MONITOR_EXEC_ARGS} is set, the contents will be passed on as
additional arguments to ${MONITOR_EXEC}. Both ${MONITOR_EXEC} and the
Mix flag can be set from the administrator interface.
b: Don't begin recording unless a call is bridged to another channel.
i: Skip recording of input stream (disables 'm' option).
o: Skip recording of output stream (disables 'm' option).
[See Also]
StopMonitor()
--
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Re: [asterisk-users] Asterisk 1.6: Custom Name for Recordings file
On Fri, May 13, 2011 at 11:07 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
See the fname_base information below.
pbx*CLI core show application monitor
-= Info about application 'Monitor' =-
[Synopsis]
Monitor a channel.
[Description]
Used to start monitoring a channel. The channel's input and output voice
packets are logged to files until the channel hangs up or monitoring is
stopped
by the StopMonitor application.
By default, files are stored to /var/spool/asterisk/monitor/. Returns
'-1' if monitor files can't be opened or if the channel is already
monitored,
otherwise '0'.
[Syntax]
Monitor([file_format[:urlbase]][,fname_base[,options]])
[Arguments]
file_format
optional, if not set, defaults to 'wav'
fname_base
if set, changes the filename used to the one specified.
options
m: when the recording ends mix the two leg files into one and delete
the two leg files. If the variable ${MONITOR_EXEC} is set, the
application
referenced in it will be executed instead of soxmix/sox and the raw leg
files will NOT be deleted automatically. soxmix/sox or ${MONITOR_EXEC}
is handed 3 arguments, the two leg files and a target mixed file name
which is the same as the leg file names only without the in/out
designator.
If ${MONITOR_EXEC_ARGS} is set, the contents will be passed on as
additional arguments to ${MONITOR_EXEC}. Both ${MONITOR_EXEC} and the
Mix flag can be set from the administrator interface.
b: Don't begin recording unless a call is bridged to another channel.
i: Skip recording of input stream (disables 'm' option).
o: Skip recording of output stream (disables 'm' option).
[See Also]
StopMonitor()
Thx Eric.
I read the link e1*CLI core show application monitor but I could not
follow what I should do to customize the file name of the recording.
I guess some changes to the dialplan is required ?
Thx
S
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk 1.6: Custom Name for Recordings file
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
On Fri, May 13, 2011 at 11:07 PM, Eric Wieling
ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
See the fname_base information below.
pbx*CLI core show application monitor
-= Info about application 'Monitor' =-
[Synopsis]
Monitor a channel.
[Description]
Used to start monitoring a channel. The channel's input
and output voice
packets are logged to files until the channel hangs up
or monitoring is stopped
by the StopMonitor application.
By default, files are stored to
/var/spool/asterisk/monitor/. Returns
'-1' if monitor files can't be opened or if the channel
is already monitored,
otherwise '0'.
[Syntax]
Monitor([file_format[:urlbase]][,fname_base[,options]])
[Arguments]
file_format
optional, if not set, defaults to 'wav'
fname_base
if set, changes the filename used to the one specified.
options
m: when the recording ends mix the two leg files
into one and delete
the two leg files. If the variable ${MONITOR_EXEC}
is set, the application
referenced in it will be executed instead of
soxmix/sox and the raw leg
files will NOT be deleted automatically. soxmix/sox
or ${MONITOR_EXEC}
is handed 3 arguments, the two leg files and a
target mixed file name
which is the same as the leg file names only without
the in/out designator.
If ${MONITOR_EXEC_ARGS} is set, the contents will be
passed on as
additional arguments to ${MONITOR_EXEC}. Both
${MONITOR_EXEC} and the
Mix flag can be set from the administrator interface.
b: Don't begin recording unless a call is bridged to
another channel.
i: Skip recording of input stream (disables 'm' option).
o: Skip recording of output stream (disables 'm' option).
[See Also]
StopMonitor()
Thx Eric.
I read the link e1*CLI core show application monitor but I
could not follow what I should do to customize the file name
of the recording.
I guess some changes to the dialplan is required ?
Re-read your message, and realized you are asking about a GUI for Asterisk.
Sorry, I can't help you with that.
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Re: [asterisk-users] Asterisk 1.6: Custom Name for Recordings file
On 5/13/11 10:57 AM, RSCL Mumbai wrote:
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
[snip..]
Thx Eric.
I read the link e1*CLI core show application monitor but I could not follow
what I should do to customize the file name of the recording.
I guess some changes to the dialplan is required ?
try something like:
Monitor(wav,${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${EXTEN})
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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[asterisk-users] Unusual message
Hi, Needed to test follow-me this evening on Asterisk 1.6.2.17 and received the following message: == Spawn extension (international-US, 0114407590XX, 5) exited non-zero on 'Local /0114407590XX@aXX-a62a;2' -- no live channels left. exiting. I have not seen that before. What does it mean ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription
Hi all, Anyone know how to make asterisk properly reply to options keep-alive? Or just force a 200 OK somehow? I recently took over a server and they have ~80 pap2 devices that send nat keep-alive and * always replies with 481 No subscription. It's more of an annoyance, I know but I like to keep my pcap's clean. S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler Sent: Friday, May 13, 2011 2:59 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription Hi all, Anyone know how to make asterisk properly reply to options keep-alive? Or just force a 200 OK somehow? I recently took over a server and they have ~80 pap2 devices that send nat keep-alive and * always replies with 481 No subscription. It's more of an annoyance, I know but I like to keep my pcap's clean. You should be able to turn NAT Keepalive off on the PAP2s. If you need a NAT Keepalive type of service, use the qualify=yes for those peers in Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription
On Fri, May 13, 2011 at 2:58 PM, Skyler skchopper...@gmail.com wrote: Hi all, Anyone know how to make asterisk properly reply to options keep-alive? Or just force a 200 OK somehow? I recently took over a server and they have ~80 pap2 devices that send nat keep-alive and * always replies with 481 No subscription. It’s more of an annoyance, I know but I like to keep my pcap’s clean. Which version of Asterisk? 1.8 should have this built-in. I made a patch for 1.6.2 which you can download at http://pastebin.com/Ls3m8t15 Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription
Really!? Wow, that would be so easy as it looks like qualify=yes is already enabled on each SIP channel. I'll give that a try/test first and report back. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, May 13, 2011 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler Sent: Friday, May 13, 2011 2:59 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription Hi all, Anyone know how to make asterisk properly reply to options keep-alive? Or just force a 200 OK somehow? I recently took over a server and they have ~80 pap2 devices that send nat keep-alive and * always replies with 481 No subscription. It's more of an annoyance, I know but I like to keep my pcap's clean. You should be able to turn NAT Keepalive off on the PAP2s. If you need a NAT Keepalive type of service, use the qualify=yes for those peers in Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1325 / Virus Database: 1500/3635 - Release Date: 05/13/11 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler Sent: Friday, May 13, 2011 3:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription Really!? Wow, that would be so easy as it looks like qualify=yes is already enabled on each SIP channel. I'll give that a try/test first and report back. The purpose of qualify= is not for NAT keep-alives, but it generates enough traffic to keep the NAT translations open anyway. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lead time for RPM's?
On 05/12/2011 02:46 PM, Jason Parker wrote: I'll make it a point to respond to this email when the new builds are available. These builds are now available. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime - ara180 - part-2
On Thu, 2011-05-12 at 21:30 +0200, bakko wrote: Hi, look if you have res_config_mysql.so module instaled on your asterisk. On CentOS /usr/lib/asterisk/modules Regards Tnx for your reply. It turned out, that mysql-support was in a different rpm (addons) As systems are never connected to the big-bad-external world, it took some time to do an upgrade and fetch the missing rpm. had some serious complaining l-users to tend to ;-) So i left the config unchanged, but noticed: kc3054*CLI sip show users Username Secret Accountcode Def.Context ACL ForcerPort j.witvliet geheim default No Yes 0277611 25b06d3a0b5ef73default No Yes kc3054*CLI kc3054*CLI sip show peers Name/username Host Dyn Forcerport ACL Port Status Realtime 277611 Unspecified) D N 0Unmonitored j.witvliet (Unspecified) D N 0Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] kc3054*CLI kc3054*CLI realtime mysql status general connected to asterisk@127.0.0.1, port 3306 with username voipadmin for 2 hours, 7 minutes. kc3054*CLI kc3054*CLI kc3054*CLI realtime mysql cache kc3054*CLI So, no more complaints at the cli or logfile, But the sip-entry from mysql 0031756 still does not show up. I hope that any suggestions/help will be educational to others Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lead time for RPM's?
BTW, is GTalk/Jabber a part of RPM now? -Vladimir On 5/13/2011 5:43 PM, Jason Parker wrote: On 05/12/2011 02:46 PM, Jason Parker wrote: I'll make it a point to respond to this email when the new builds are available. These builds are now available. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-cpu utilization 60 %
Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
