[asterisk-users] func SHARED, how to use?
Hello! Could somebody tell me how to use SHARED function? I want to get RTCP stats from SIP , but current channel is DAHDI. How can I get SIP channel? -- _ -- 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] Dialplan extension pattern matching for '/' character
Hi Friends, LOCAL/*89/9875784578 I want to match above dialstring into dialplan context. How can i match dialplan extension pattern matching for *89/9875784578 with including '/' character. Thanks in advance. -- Best Regards, Rajnikant Vanza -- _ -- 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] getting error chan_sip.c: Failed to grab lock, trying again..
I understand the point. However, till the time I upgrade, I need to figure out how to stop this. Also, I checked the bug ID 6181. but could not find something like a version in which this is closed. So unable to decide as to which version I need to go to. Regards --- On Mon, 9/20/10, dotnetdub dotnet...@gmail.com wrote: From: dotnetdub dotnet...@gmail.com Subject: Re: [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again.. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, September 20, 2010, 8:03 AM On 20 September 2010 05:33, dashy dude dashy_v2...@yahoo.com wrote: Hi, I tried disabling cdr_addon_mysql.so. Still error comes let's say once a day or so. Is there anything else I can do about? rgds --- On Thu, 9/9/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: From: Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Subject: Re: [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again.. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, September 9, 2010, 2:17 AM Hi! I am running asterisk ver 1.2.4 and have faced this error: Try a downgrade to Asterisk 0.7.1 ;- Philipp Do you honestly think that you are going to get support here for this version of Asterisk? Upgrade to something from the last year or so.. 1.4.xx branch is very stable. -Inline Attachment Follows- -- _ -- 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] Not able to join conference
I was wondering what happened if YOU put that number in. Does it put everyone in to the same conference? That would, at least, prove that the MeetMe app was working as it should (unless you've tried this already). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 20 September 2010 14:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Not able to join conference it's going to put you in conf no 500 without prompting you to enter a conference number I guess, but i don't it's going to solve my issue. actually I'm atill wondering is there a way to debug just Meetme app output or the only way is turn the whole debug thing on? On Mon, Sep 20, 2010 at 4:11 AM, Andrew Thomas a...@datavox.co.uk wrote: What happens if you put in a 'room' number? Eg: exten = 8080,3,MeetMe(500|MDci) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 17 September 2010 14:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not able to join conference Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the conference and meet with the people already in, here is the lines corresponding to conf in the dialplan, that would be a big help if you guys can help diagnose the issue. exten = 8080,1,Answer exten = 8080,2,Wait,1 exten = 8080,3,MeetMe(|MDci) 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 -- Abdullah -- _ -- 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] Not able to join conference
On Tue, Sep 21, 2010 at 8:33 AM, Andrew Thomas a...@datavox.co.uk wrote: I was wondering what happened if YOU put that number in. Does it put everyone in to the same conference? That would, at least, prove that the MeetMe app was working as it should (unless you've tried this already). yes, as i said, it will place all caller in conf no 500. and it's not supposed to work like that, for the meetme app, it's working fine except this issue and i cannot even reproduce the issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 20 September 2010 14:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Not able to join conference it's going to put you in conf no 500 without prompting you to enter a conference number I guess, but i don't it's going to solve my issue. actually I'm atill wondering is there a way to debug just Meetme app output or the only way is turn the whole debug thing on? On Mon, Sep 20, 2010 at 4:11 AM, Andrew Thomas a...@datavox.co.uk wrote: What happens if you put in a 'room' number? Eg: exten = 8080,3,MeetMe(500|MDci) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 17 September 2010 14:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not able to join conference Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the conference and meet with the people already in, here is the lines corresponding to conf in the dialplan, that would be a big help if you guys can help diagnose the issue. exten = 8080,1,Answer exten = 8080,2,Wait,1 exten = 8080,3,MeetMe(|MDci) 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 -- Abdullah -- _ -- 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 -- Abdullah -- _ -- 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] 3rd party app store
On 09/21/2010 03:41 AM, Rod Montgomery wrote: [/snip] Does anyone reading this have an opinion on whether commercial listings for complementary products and services should appear directly on Asterisk.org? Just my two cents - but I prefer that organisations keep a clear line between open-source/not-for-profit and commercial operations or intentions. You get a not-very-pleasant smell about it when the two start to intermingle to the point where you can't tell where one ends and the other begins. Sebastian -- _ -- 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] Digest Username/auth name mismatch
On 09/21/2010 04:26 AM, t. k wrote: Hi Thanks for help. I will try to help. But others might know more. What SIP client are you using - a softphone, a hardphone? It looks like the client is sending the full at 192.168.0.1 instead of just as the username. Sebastian That's right.hardphone is sending at 192.168.0.1 for Proprietary specification. ※Digest usrname can't change with SIP Client. so I would like to solve this hardphone issue with asterisk. Isn't there any way to configure the username in the hardphone to be just ? Sebastian -- _ -- 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] getting error chan_sip.c: Failed to grab lock, trying again..
Failed to grab lock, is usually used when referencing that it can't lock the port (in this case the sip port you use), because the port is used by another app/service. Just a tip. And by the way, *DO* upgrade. On Tue, Sep 21, 2010 at 12:08 PM, dashy dude dashy_v2...@yahoo.com wrote: I understand the point. However, till the time I upgrade, I need to figure out how to stop this. Also, I checked the bug ID 6181. but could not find something like a version in which this is closed. So unable to decide as to which version I need to go to. Regards --- On *Mon, 9/20/10, dotnetdub dotnet...@gmail.com* wrote: From: dotnetdub dotnet...@gmail.com Subject: Re: [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again.. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, September 20, 2010, 8:03 AM On 20 September 2010 05:33, dashy dude dashy_v2...@yahoo.comhttp://mc/compose?to=dashy_v2...@yahoo.com wrote: Hi, I tried disabling cdr_addon_mysql.so. Still error comes let's say once a day or so. Is there anything else I can do about? rgds --- On Thu, 9/9/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.dehttp://mc/compose?to=klitz...@pool.informatik.rwth-aachen.de wrote: From: Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.dehttp://mc/compose?to=klitz...@pool.informatik.rwth-aachen.de Subject: Re: [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again.. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comhttp://mc/compose?to=asterisk-us...@lists.digium.com Date: Thursday, September 9, 2010, 2:17 AM Hi! I am running asterisk ver 1.2.4 and have faced this error: Try a downgrade to Asterisk 0.7.1 ;- Philipp Do you honestly think that you are going to get support here for this version of Asterisk? Upgrade to something from the last year or so.. 1.4.xx branch is very stable. -Inline Attachment Follows- -- _ -- 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 -- -- Ondrej Škopek -- _ -- 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] 3rd party app store
I wholly disagree. Open-source does not imply not-for-profit at all. Look at Red Hat. They sell open source software, by way of selling support, and access to stable repositories for updates. So this line does not need to exist. If the line does exist, then I agree it should be well defined. Thanks, Lyle J. McKarns --- Network Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax: 1 207 725 8552 Nexus Management, Inc.│ Registered Office: 4 Industrial Parkway, Suite 101, Brunswick, Maine. 04011│Company No. 19891257D, Registered in Maine│ A member of the Nexus Management Plc group of companies -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Sent: Tuesday, September 21, 2010 8:58 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 3rd party app store On 09/21/2010 03:41 AM, Rod Montgomery wrote: [/snip] Does anyone reading this have an opinion on whether commercial listings for complementary products and services should appear directly on Asterisk.org? Just my two cents - but I prefer that organisations keep a clear line between open-source/not-for-profit and commercial operations or intentions. You get a not-very-pleasant smell about it when the two start to intermingle to the point where you can't tell where one ends and the other begins. Sebastian -- _ -- 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] Unexplained message in 1.6.2
Every time I start Asterisk or do a simple reload I see this message: Cannot open maximum file descriptor 32767 at boot? No such file or directory Does anybody have some idea of what can it be? It did not happen in version 1.4. Philip -- _ -- 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] Unexplained message in 1.6.2
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR Sent: Tuesday, September 21, 2010 8:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unexplained message in 1.6.2 Every time I start Asterisk or do a simple reload I see this message: Cannot open maximum file descriptor 32767 at boot? No such file or directory Does anybody have some idea of what can it be? It did not happen in version 1.4. Philip Tilghman did a post on this for 1.4 SVN - http://www.mail-archive.com/svn-comm...@lists.digium.com/msg51758.html My C is iffy, but I think it means Asterisk tried to open too many files on startup. -- _ -- 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] AGI Delimiter in 1.6
On 16 September 2010 22:23, Barry Miller asterisk-us...@notanet.net wrote: For an interim fix, setting res_agi=1.4 in the [compat] section of asterisk.conf should work. See UPGRADE-1.6.txt . I have tried this but it still complains about the pipe not being a comma. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] AGI Delimiter in 1.6
On 09/21/2010 04:22 PM, Jon Farmer wrote: On 16 September 2010 22:23, Barry Millerasterisk-us...@notanet.net wrote: For an interim fix, setting res_agi=1.4 in the [compat] section of asterisk.conf should work. See UPGRADE-1.6.txt . I have tried this but it still complains about the pipe not being a comma. Regards Jon Hello, in asterisk 1.4 this works : exten = s,n,Queue(queuenametimeout,test.agi^VAR) in asterisk 1.6 this works : exten = s,n,Queue(queuenametimeout,test.agi,VAR) So you need . 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
Re: [asterisk-users] func SHARED, how to use?
Hi! Could somebody tell me how to use SHARED function? http://www.voip-info.org/wiki/index.php?page=Asterisk+func+shared I want to get RTCP stats from SIP, but current channel is DAHDI. How can I get SIP channel? If you have one DADHI and one SIP channel bridged together, then only for the SIP channel you will be able to retrieve rtcp data. Depending on wether that SIP channel is the first (local) or the second (outbound or remote) call leg you will need to follow the approach described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+rtcp Quote: Use the M option of Dial() if you would like to get the codec (audionativeformat) of the remote call leg/channel and similar data. Those are not available anymore during the hangup phase (h extension), however you can store them directly in the CDR system, or use the SHARED function to export them to the local call leg/channel. Philipp -- _ -- 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] Unexplained message in 1.6.2
On Tuesday 21 September 2010 08:42:04 CDR wrote: Every time I start Asterisk or do a simple reload I see this message: Cannot open maximum file descriptor 32767 at boot? No such file or directory Does anybody have some idea of what can it be? It did not happen in version 1.4. Philip Essentially what this is saying is that you've raised your per-process file descriptor limit higher than your booted kernel will allow in a single process. This should almost never happen. See the value here: bash% sysctl fs.file-max -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] 3rd party app store
Personally, I would like to see less commercial marketing on http://asterisk.org. I count 5 separate marketing ads on the download page alone. This is just my opinion. The level of commercialism on the Asterisk.org download page does not bother me at all. Seems eminently fair for Digium to advertise their free (!) entry points for Switchvox and FFA. Asterisk training support - I have no problem with those either. The support and training are pay-for products, but are a big help to the community also. My $0.02. Cassius Smith -- _ -- 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] random hangups on RBS T1
Hi, I have an asterisk 1.4.35 server with a Digium TE410P (1st gen) four port T1 card. Only one RBS T1 plugged into it right now. I have been getting complaints about random hangups. Endpoints are all remote, but I track very closely the latency (by graphing the output of sip show peers) which normally shows me when a peer is having connectivity issues. Several that I have investigated this morning have no such issues (latency less than 20ms and steady). I have several servers with Sangoma A104d cards, and the Sangoma driver has a debug mode that lets me see the RBS bit transitions. I have used this in the past to prove that the T1 provider is actually triggering the hangup from their side. Does any such debug mode exist for the Digium cards? I would like to dig into this, because if I can prove the carrier is at fault I will have hard data to bring to a PUC meeting next month :) Any suggestions? Thanks, j -- _ -- 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] Unexplained message in 1.6.2
Every time I start Asterisk or do a simple reload I see this message: “Cannot open maximum file descriptor 32767 at boot? No such file or directory”. It only works if I set 1024 in asterisk.conf maxfiles However, my sysctl fs.file-max fs.file-max = 65535 and my ulimits are ulimit -a core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 40 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 40 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 1056768 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited -- _ -- 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] random hangups on RBS T1
On 09/21/2010 10:48 AM, Jeff LaCoursiere wrote: I have several servers with Sangoma A104d cards, and the Sangoma driver has a debug mode that lets me see the RBS bit transitions. I have used this in the past to prove that the T1 provider is actually triggering the hangup from their side. Does any such debug mode exist for the Digium cards? I would like to dig into this, because if I can prove the carrier is at fault I will have hard data to bring to a PUC meeting next month :) Any suggestions? Jeff, you can monitor the state of the RBS bits via 'dahdi_tool'. But in case you need a running log (and I haven't tested this) I added a patch to https://issues.asterisk.org/view.php?id=18025 if you want to try that out. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] random hangups on RBS T1
On Tue, 21 Sep 2010, Shaun Ruffell wrote: On 09/21/2010 10:48 AM, Jeff LaCoursiere wrote: I have several servers with Sangoma A104d cards, and the Sangoma driver has a debug mode that lets me see the RBS bit transitions. I have used this in the past to prove that the T1 provider is actually triggering the hangup from their side. Does any such debug mode exist for the Digium cards? I would like to dig into this, because if I can prove the carrier is at fault I will have hard data to bring to a PUC meeting next month :) Any suggestions? Jeff, you can monitor the state of the RBS bits via 'dahdi_tool'. But in case you need a running log (and I haven't tested this) I added a patch to https://issues.asterisk.org/view.php?id=18025 if you want to try that out. Cheers, Shaun Fantastic! I will give the patch a shot. Thanks, j -- _ -- 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] Polycom dhcp boot
Did you get this to work? If not, shoot me an email. We use the Polycom's, and I can send you our config file. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of colin mcdermott Sent: Friday, September 10, 2010 7:36 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom dhcp boot Hi all I have a few Polycom 331's but after following allot of advice I can't get them to provision from a dhcp boot server. We have a sonicwall router in place. I can press setup and set the FTP boot server to my * box. From there th phones boot fine. But I cannot get them to autoprovision. I have tried dhcp option 66=ipaddres. 66=FTP://PlcmSpIp:plcms...@192.168.1.1/ u ahve also tried options 129, 150, 160, etc. I realise that this is not an asterisk issue. But does anyone have any experience on this (particularly using sonicwall routers for Dhcp)? -- _ -- 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 incoming message. Checked by AVG - www.avg.com Version: 9.0.851 / Virus Database: 271.1.1/3132 - Release Date: 09/13/10 02:35:00 -- _ -- 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] digits in chan_dahdi
Hello I use Asterisk with FXS extensions in chan_dahdi and I'm having trouble detecting the digits in dahdi. I dial 12345678, but only '16 'is received by the asterisk. The following appears in the logs: [Sep 21 18:11:44] DTMF [8536] channel.c: DTMF end '1 'received on DAHDI/10-1, duration 0 ms [Sep 21 18:11:44] DTMF [8536] channel.c: DTMF end accepted without begin '1 'on DAHDI/10-1 [Sep 21 18:11:44] DTMF [8536] channel.c: DTMF end passthrough '1 'on DAHDI/10-1 [Sep 21 18:11:45] DTMF [8536] channel.c: DTMF end '6 'received on DAHDI/10-1, duration 0 ms [Sep 21 18:11:45] DTMF [8536] channel.c: DTMF end accepted without begin '6 'on DAHDI/10-1 [Sep 21 18:11:45] DTMF [8536] channel.c: DTMF end passthrough '6 'on DAHDI/10-1 [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Set [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Set [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: gotoif [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Set [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Set [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: SetMusicOnHold [Sep 21 18:11:48] DEBUG [8536] app_macro.c: Executed application: Goto [Sep 21 18:11:48] DEBUG [8536] chan_dahdi.c: Took DAHDI/10-1 off hook I use the headset Zox TS19. I tried changing the value of toneduration = 100 but did not work. Anyone know how I can solve this problem? thank you very much. Marcus Vinícius. -- _ -- 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] digits in chan_dahdi
I dial 12345678, but only '16 'is received by the asterisk. You may want to try relaxdtmf=yes in chan_dahdi.conf. That fixed a similar problem for me. -- _ -- 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] Mixing ISDN and R2 in the same card...
I have a project where I need to connect two E1 links from different providers. One will be PRI ISDN (Telefonica) and the other MFC/R2 (Telmex). There should not be any problem supporting both types of link on a single TE220B card but my concern is more about who will be the primary timing source. Since the links come from different companies how do I choose the clock source? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- 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] Bug with Realtime?
I checked the bug reports and all I could find was similar issues with the Asterisk 1.6 which (according to the reports) have been resolved. I couldnt find anyone talking about 1.4 so I created a new issue and someone closed the case and added this note:- This does not appear to be a bug, but rather a support issue. Please use the asterisk-users mailing list for such issues. The problem looks like your device has not re-registered after your 'sip reload' which means it does not exist in memory, and thus causes Asterisk to not know where to send the call. Your device needs to re-register after a 'sip reload' in order for Asterisk to know where to send the call. I really think that sip reload shouldn't purge all the realtime peer registrations. It should treat the realtime peers the same way as the hardcoded peers. As i've said, the hardcoded peers don't lose registration when I issue a SIP RELOAD. Asterisk should be flexible enough to allow modification of the sip.conf file without losing all the realtime registrations. Does anyone have a comment on the subject? Am I expecting too much? I'm open to feedback. Thanks Dan -- _ -- 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] Res: digits in chan_dahdi
Hello, thanks for the reply. I tried relaxdtmf = yes but has not worked. If I type very slowly digits are recognized normally. But if I dial a number and enter the redial button, the digits are recognized in the asterisk. It appears that: [Sep 21 19:20:24] DEBUG [4751] chan_dahdi.c: waitfordigit returned 0 ... tks Marcus Vinicius De: Richard Kenner ken...@gnat.com Para: asterisk-users@lists.digium.com Enviadas: Terça-feira, 21 de Setembro de 2010 18:48:54 Assunto: Re: [asterisk-users] digits in chan_dahdi I dial 12345678, but only '16 'is received by the asterisk. You may want to try relaxdtmf=yes in chan_dahdi.conf. That fixed a similar problem for me. -- _ -- 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] Res: digits in chan_dahdi
I tried relaxdtmf = yes but has not worked. If I type very slowly digits are recognized normally. Then indeed it won't make a difference. If that were your problem, it likely wouldn't work at any speed. -- _ -- 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] Bug with Realtime?
On Tue, 2010-09-21 at 19:04 -0400, Dan Journo wrote: I checked the bug reports and all I could find was similar issues with the Asterisk 1.6 which (according to the reports) have been resolved. I couldnt find anyone talking about 1.4 so I created a new issue and someone closed the case and added this note:- This does not appear to be a bug, but rather a support issue. Please use the asterisk-users mailing list for such issues. The problem looks like your device has not re-registered after your 'sip reload' which means it does not exist in memory, and thus causes Asterisk to not know where to send the call. Your device needs to re-register after a 'sip reload' in order for Asterisk to know where to send the call. I really think that sip reload shouldn't purge all the realtime peer registrations. It should treat the realtime peers the same way as the hardcoded peers. As i've said, the hardcoded peers don't lose registration when I issue a SIP RELOAD. Asterisk should be flexible enough to allow modification of the sip.conf file without losing all the realtime registrations. Does anyone have a comment on the subject? Am I expecting too much? I'm open to feedback. I use realtime on 1.4 and 1.6 servers but always with rtcachefriends=yes in sip.conf so I can use things like sip show peers. My experience is that when I issue a sip reload all registrations disappear but the moment a call comes in for a peer it uses the last IP where it was registered to send the call. I guess this is the equivalent do doing sip show peer XXX load. The peer does not need to register again to get calls. I think this works because the internal database has the last IP of the peer even after a sip reload. If you do a database show you will see something like: /SIP/Registry/ 192.168.2.215:5060:3600::sip:x...@192.168.2.215:5060;transport=udp -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- 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: digits in chan_dahdi
On 09/21/2010 06:09 PM, Marcus Vinicius wrote: I tried relaxdtmf = yes but has not worked. If I type very slowly digits are recognized normally. But if I dial a number and enter the redial button, the digits are recognized in the asterisk. It appears that: [Sep 21 19:20:24] DEBUG [4751] chan_dahdi.c: waitfordigit returned 0 ... I don't know if this will help, but I saw something similar recently where the rxgain and txgain in chan_dahdi.conf were set too high and interfered with the DTMF detection algorithms. Resetting them to 0.0 restored expected DTMF detection behaviour. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Bug with Realtime?
I use realtime on 1.4 and 1.6 servers but always with rtcachefriends=yes in sip.conf I already use that and it doesnt seem to re-register when a call comes in. Only when the registration period expires, or the peer dials out. -- _ -- 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] AGI Delimiter in 1.6
Hi, I fixed it in the end by adding the sip headers I was interested in as extra x headers in the openser config. Then just capturing these in the asterisk dialplan as variables. Simples. Regards Jon On 21 Sep 2010 16:03, Jonas Kellens jonas.kell...@telenet.be wrote: On 09/21/2010 04:22 PM, Jon Farmer wrote: On 16 September 2010 22:23, Barry Millerasterisk-us...@notanet.net wrote: For an interim fix, setting res_agi=1.4 in the [compat] section of asterisk.conf should work. See UPGRADE-1.6.txt . I have tried this but it still complains about the pipe not being a comma. Regards Jon Hello, in asterisk 1.4 this works : exten = s,n,Queue(queuenametimeout,test.agi^VAR) in asterisk 1.6 this works : exten = s,n,Queue(queuenametimeout,test.agi,VAR) So you need . 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 -- _ -- 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] Solving the CDR mess of attended transfers
On Tue, Sep 7, 2010 at 5:36 PM, Fabiano Carlos Heringer b...@grupoheringer.com.br wrote: Em 07/09/2010 17:15, Miguel Molina escreveu: El 07/09/10 14:49, Fabiano Carlos Heringer escribió: Is there a way to solve the mess on CDR caused by CDR Transfer? anyway, by paid support, no paid, or another way... Im going crazy about this. My boss is really furious because he don´t understand nothing on the CDR. I tried the 1.6.2.11, Asterisk 1.8 beta, and everything still the same. Any solution? Thanks! Hi Some quick measures: 1. Enable unanswered=yes on cdr.conf and try to see if it helps you with the CDR. 2. Try using CEL (Channel Event Logging) in 1.8-beta and try to see if that helps in a definite way. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Hi, will make this change on my cdr.conf About CEL on asterisk 1.8 i tried some test on my test server, he really logs each event on log, but i did not understood how he will work on a user view (most simple). It´s possible to log this events on a database such mysql? Thanks! Sorry for the delay, things have been busy here. Yes, there are problems with the existing CDR interface, mostly historical, because as Asterisk grew, the CDR system became obsolete. There were attempts to make it work, but structurally and architecturally, it was just not going to work. CEL was my answer, built on the channel event goodness that Russell. It's now in 1.8; but it lacks a converter to CDRs. You *could* just use the string of events coming out of CEL, but... I'd love to see your SQL statements to pull things together! I had begun writing a CEL-CDR converter, but got laid off before I could finish it. It makes a good start toward a finished package. Long ago (what, almost 2 years now?) I proposed two methods of generating CDR's. One was 'simple', the other 'Complex, or Leg Based. Since then, I refined the doc to just 'Simple', and outlined with some examples how it would/should work. The doc still needs to be cleaned up, but you may make sense of it. The trouble with CDRs is that no two shops can agree on a CDR standard that involves transfers, parks, etc. Beyond the start, answer, and end times, and some fundamental data about the call (source, dest, responsible party, etc.) There isn't much unity about what timepoints need to be represented, etc. And I'd seen so few implementations, I couldn't judge a good way to generalize the CDR converter. So, I challenge everyone to look at my simple CDR definition, and see it would possible for you to adapt it (perhaps via a mapping from it to your desired conflagration/configuration. To look at the doc, do svn co http://svn.digium.com/svn/team/murf/asterisk-RFCs and look at the document in there (I have a few different formats, the .docx is the source). It's been in flux. Just the first few examples are accurate. Let me know what you think. murf -- Steve Murphy ParseTree Corp -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Hi Everyone, I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs. So, the tunnel is established nicely and everyone can ping others. sip show peers shows the local subnet of the SIP Phones registered (192.168.100.0/24 ). But there is the old bad one-way audio. Calls also drop after few seconds. In the SIP debug I can see that asterisk uses it's external public IP address to communicate to endpoints that are known to it as the 192.168.100.0/24 endpoints and the endpoints identify themselves with the OpenVPN tunnel IP address scheme in one part of the sip handshake. How can this be fixed? After all, with the OpenVPN this should all look like an internal network to Asterisk. I have added my comments followed by # to lines below that are problematic. --- SIP read from UDP:192.168.100.5:5060 ---#This line is good as it uses the local DHCP supplied network address scheme INVITE sip:2...@172.16.0.1:5060 SIP/2.0 #This line is BAD. Why are we inviting Ext. 203 with it's OpenVPN IP while it's on the same network of 192.168.50.0/24 as 202? Via: SIP/2.0/UDP 192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6 Max-Forwards: 70 From: SIP Phone - Ext. 202 sip:2...@172.16.0.1:5060;tag=6d6f8c4226 #BAD line again. Should be SIP:2...@192.168.100.6 sip%3a...@192.168.100.6 To: 203 sip:2...@172.16.0.1:5060 #Bad again Call-ID: 43af67a634e06e75 CSeq: 32058 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: SIP Phone - Ext. 202 sip:2...@192.168.50.5:5060 ;transport=udp;+sip.instance=urn:uuid:--1000-8000-00085D25B72F Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.5.2.1500 Content-Type: application/sdp Content-Length: 594 Basically the phones should only send with FROM their local 192.168.100.0/24address and Asterisk should only send ANSWER and ACK back to 192.168.100.0/24 rather than sending it to 172.16.0.0/24 (which is the openvpn client ip). Once above is fixed, I think all the audio and call cut will go away. I hate to use a sip proxy in this situation since I already have an openvpn connection. Any feed back is appreciated. 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
