[asterisk-users] Astricom 2014 presentations
Hi Will the presentations made at Astricom 2014 be made public as recorded videos ? thanks Bogdan -- _ -- 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 12 - zombie processes
Hello Mathew and everyone, We are still having reboots on our asterisk servers. The latest 12.6.1 release doesn't fix the issue. We have the core files of the latest reboots and also debug taken during the reboot. We would like to open an issue. What kind of information you need for the issue? Thanks, Yaron. On Tue, Oct 28, 2014 at 5:10 PM, Yaron Nachum nachum.ya...@gmail.com wrote: Mathew, When I run 'ps -ef|grep asterisk' the following processes are displayed: root 6861 1 0 Aug27 ?00:00:00 /bin/sh /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C /ivr/app/asterisk/etc/asterisk/asterisk.conf asterisk 8062 6861 3 Oct27 ?00:44:56 /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c root 20776 2200 0 11:20 pts/200:00:33 tail -f asterisk.log asterisk 23076 8062 0 17:01 ?00:00:00 [asterisk] defunct asterisk 23897 8062 0 17:03 ?00:00:00 [asterisk] defunct also when I run top the same amount of zombie processes are displayed: Tasks: 185 total, 1 running, 182 sleeping, 0 stopped, 2 zombie Regarding the AGI - we are using AGI in order to run php scripts for external logic. I have printed the PIDs of the php scripts and none of them are related to the PID's of those zombie processes. Do you have any idea how to find out what are these processes? Yaron. On Tue, Oct 28, 2014 at 4:53 PM, Matthew Jordan mjor...@digium.com wrote: On Tue, Oct 28, 2014 at 9:44 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello Mathew, In the following tutorial it says that channel are marked with ZOMBIE flag. From your response I assume it has no connection to my problem. https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Bridging+Project Regrading the zombie processes issue we are having. We have debug taken from the server during such process is invoked. If you want I can attache it. A zombie channel has nothing to do with a process. It was an artefact of an internal process known as a masquerade. While masquerades do sometimes still occur in Asterisk 12+, they are far less frequent and are no longer externally visible. Why do you think you have zombie processes? Asterisk does use a large number of threads, but generally rarely forks processes unless you are using something like original AGI. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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 -- _ -- 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] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.
Hello Experts, Could anybody pl help resolve my query? Thanks Regards, Subbaiah Nachiappan From: NACHIAPPAN, SUBBAIAH (SUBBAIAH) Sent: Tuesday, October 28, 2014 6:04 PM To: 'asterisk-users@lists.digium.com' Subject: RE: Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface. Hello Folks, Forgot to mention the software Versions which I am using: Asterisk: 1.8 Free PBX: 2.11 Asterisk NOW: 5.211.65 Thanks Regards, Subbaiah Nachiappan From: NACHIAPPAN, SUBBAIAH (SUBBAIAH) Sent: Tuesday, October 28, 2014 5:52 PM To: 'asterisk-users@lists.digium.com' Subject: Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface. Hello, I am new to Asterisk forum :). I have a requirement of terminating 3G Mobile originated calls (coming through 3G-MSC) to EPBX Phones via Asterisk PBX. Setup: Mobile Mobile Switching Center ( 3G)-SIP interface---Asterisk PBX---SIP Phone. I wanted to know if I require SIP licenses to integrate 3G MSC with my Asterisk server. I know there is a wealth of information in wiki link, but I am unable to locate the required configuration document which will help me in integrating MSC with Asterisk EPBX via SIP interface. Thanks in Advance!!! Thanks Regards, Subbaiah Nachiappan -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.commailto:asterisk-users-requ...@lists.digium.com Sent: Tuesday, October 28, 2014 5:41 PM To: NACHIAPPAN, SUBBAIAH (SUBBAIAH) Subject: Welcome to the asterisk-users mailing list -- _ -- 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] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.
On 29/10/14 12:59 pm, A J Stiles wrote: Imagine what would have happened to the human race if Ugg the Caveman decided not to share the secret of making fire with everyone freely, but instead went around demanding shiny beads with menaces from anyone who just wanted to keep themselves warm . That's the best analogy I've heard in favour of open development for a long time, and something that non-techs can understand. I thank you sir :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 12 - zombie processes
On Wed, Oct 29, 2014 at 6:59 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello Mathew and everyone, We are still having reboots on our asterisk servers. The latest 12.6.1 release doesn't fix the issue. We have the core files of the latest reboots and also debug taken during the reboot. We would like to open an issue. What kind of information you need for the issue? Please do open an issue on issues.asterisk.org. Instructions for generating a backtrace are on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace Make sure you have DONT_OPTIMIZE and, preferably, BETTER_BACKTRACES selected in menuselect. Depending on the nature of the crash, you may be asked for more information, but we won't know until we see the backtrace. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] Astricom 2014 presentations
On 10/29/2014 05:50 AM, Bogdan Cristea wrote: Hi Will the presentations made at Astricom 2014 be made public as recorded videos ? thanks Bogdan I'll second the request for that, and also ask if the sessions on Kamailio will be similarly available. Cheers, 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] Astricom 2014 presentations
I can confirm that all the videos from AstriCon 2014 will be available at www.AstriCon.net within about 3 weeks. On 29 Oct 2014 16:33, Jeff LaCoursiere j...@jeff.net wrote: On 10/29/2014 05:50 AM, Bogdan Cristea wrote: Hi Will the presentations made at Astricom 2014 be made public as recorded videos ? thanks Bogdan I'll second the request for that, and also ask if the sessions on Kamailio will be similarly available. Cheers, 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 -- _ -- 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] My Asterisk can not send fax via T.38
Hello, I am a newer on asterisk. when I tried to send fax, I can not get success. After doing a lot of reseach, I decide to ask my question at here. Instead of using old fax machine, I want my system can send fax via t.38. Could anyone give me some idea to correct my configuration? Thanks a lot! Here is the detail: my system: *ubuntu 14.04 + asterisk 11.7* I use apt-get to install the ubuntu default version of asterisk 11.7. for easy debug, I just change a few param value with the default version sip.conf t38pt_udptl = yes,redundancy,maxdatagram=400 faxdetect = yes extension.conf [sendFAX] exten = s,1,VERBOSE(sending fax...) exten = s,n,Set(FAXOPT(headerinfo)=Fax from a Demo test) exten = s,n,SendFAX(/tmp/demo.tiff,f) ;I get demo.tiff file from $ gs -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=letter -sOutputFile=dest src exten = s,n,VERBOSE(ok!) exten = s,n,Hangup I use AMI to originate a fax call. In the CLI, everything looks well. I didn't get any error message. When I use wireshark to check the detail of this comunication, I found Asterisk used G711 instead of using T.38 which is expected. however, at the receiver end, I didn't receive the fax, and I just got a error Dcn No Dis After a research, I got this: T.30 Fax Signaling Messages In a Voip fax call, T.38 packets are preceded and succeeded by T.30 fax signaling messages. These messages include: 1. DIS: Digital Identification Signal indicating terminating fax capabilities (for example, data rate) 2. DCS: Digital Command Signal indicating transmission mode that will be used by originating fax (for example, transfer rate) 3. TCF: Training Check Sequences (sent for 1.5 seconds) 4. CFR: Confirmation To Receive indicating the receiving fax is ready to receive the document 5. MPS: MultiPage Signal (sent after each page if more than one page is sent) 6. MCF: Message Confirmation indicating the page was received 7. EOP: End Of Procedure message indicating there are no more pages to be sent 8. DCN: Disconnect message Additional optional messages: 1.CSI: Called Subscriber Identification 2.TSI: Transmitting Subscriber Identification But I am still confused with what Dcn No Dis means what's wrong with my asterisk system. I am sure of these: 1. the receiver is working well. 2. My ISP provider is fully support fax termination both in g711 and t.38 3. My testing server is not behind any firewall. The demo.jpg http://goo.gl/ix8JzY is my wireshark's screenshot. form 19 to 1841, all traffic are RTP package. The t.38 png http://goo.gl/lpD7Z5 diagram illustrates is a typical fax call. -- _ -- 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 12 - zombie processes
On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Mathew, When I run 'ps -ef|grep asterisk' the following processes are displayed: root 6861 1 0 Aug27 ?00:00:00 /bin/sh /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C /ivr/app/asterisk/etc/asterisk/asterisk.conf asterisk 8062 6861 3 Oct27 ?00:44:56 /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c root 20776 2200 0 11:20 pts/200:00:33 tail -f asterisk.log asterisk 23076 8062 0 17:01 ?00:00:00 [asterisk] defunct asterisk 23897 8062 0 17:03 ?00:00:00 [asterisk] defunct also when I run top the same amount of zombie processes are displayed: Tasks: 185 total, 1 running, 182 sleeping, 0 stopped, 2 zombie Regarding the AGI - we are using AGI in order to run php scripts for external logic. I have printed the PIDs of the php scripts and none of them are related to the PID's of those zombie processes. Do you have any idea how to find out what are these processes? Yaron. Are you doing anything like: # asterisk -rx 'core show channels' via an external process? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 12 Dialplan
I am happy to report that https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Applications+REST +API has the answer to my dilemma. It seems an app has to subscribe to channel events before it can receive the events like ChannelVarset... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Tuesday, October 28, 2014 2:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 12 Dialplan Tried this: wscat -c ws://myhost.mydomain.net:8090/ari/events?api_key=secret:secretapp=hell o-world It is only showing the stasis related events. I am interested in AMI events, specifically Varset. Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Monday, October 27, 2014 7:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 12 Dialplan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Monday, October 27, 2014 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 12 Dialplan On Mon, Oct 27, 2014 at 2:40 PM, Murthy Gandikota mgandik...@nts.net wrote: I am unable to detect the Manager_Setvar event using ARI. Can you please let me know, in ARI lingo, the curl or javascript code to detect the AMI Manager_Setvar event for myvar in the following dialplan: [default] exten = 1000,1,NoOp() same = n,Answer() same = n,set(myvar=test) same = n,Stasis(hello-world) same = n,Hangup() Thanks Perhaps it would be easier if you provided some information about the ARI application you've written. Have you connected a WebSocket? Are you receiving other ARI events? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org I am using ari4java to capture stasis events like StasisStart, StatisEnd, etc. However, I am unable to capture the Varset event as explained before. In particular the myvar variable is not associated with any app It is perhaps a channel variable. -- _ -- 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] Astricom 2014 presentations
On 10/29/2014 05:50 AM, Bogdan Cristea wrote: Hi Will the presentations made at Astricom 2014 be made public as recorded videos ? thanks Bogdan I'll second the request for that, and also ask if the sessions on Kamailio will be similarly available. Cheers, j That would be awesome if they chose to do this. Bryant -- _ -- 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] OT: script to remove leading and trailing silence
Anybody care to share a script or snippet of what they use to remove leading and trailing silence from customer recorded files? I've fiddled with sox to remove the leading; reverse the file; remove the now leading; and reverse the file again, but I'm not really happy with my results. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 12 Dialplan
On Wed, Oct 29, 2014 at 1:21 PM, Murthy Gandikota mgandik...@nts.net wrote: I am happy to report that https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Applications+REST+API has the answer to my dilemma. It seems an app has to subscribe to channel events before it can receive the events like ChannelVarset... That's correct. You are only implicitly subscribed to channels that are in the Stasis application your websocket is for (in your case, 'hello-world'). Otherwise, you have to subscribe to various event sources through the applications resource. The Introduction to ARI and Channels page on the wiki has more on this here: https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Channels#IntroductiontoARIandChannels-ChannelsinaStasisApplication -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)
On Oct 28, 2014, at 5:03 PM, Ben Langfeld b...@langfeld.me wrote: On 28 October 2014 19:47, Derek Andrew derek.and...@usask.ca wrote: What is the alternative to the dial plan? Is everyone talking about getting rid of the statements like: exten = s,1, what is the alternative? Remote applications based on APIs like ARI. This is the start of the discussion, and please remember that nothing has been decided or even presented as a robust plan yet. This is brain-storming. We’re not at the start of the “discussion” to deprecate the dial plan. The start of the “discussion” began when some developers decided to try standing Asterisk on its head by adding “asynchronous AGI.” Evidently, that was good so then they continued the “discussion” by adding ARI/Stasis. Now the “discussion” is in full career as ARI/Stasis has metastasized beyond its original scope to encompass all of Asterisk. None of said “discussion” ever happened on the lists nor was the broader Asterisk community involved as far as I can determine. A parallel “discussion” was started by a shill at AstiCon this year to begin to get the “vast unwashed” onboard with ARI/Stasis, that is, so that Matt could come back from AstiCon claiming that the broader Asterisk community is in agreement that ARI/Stasis is the future of Asterisk and that the dial plan can be deprecated. The inevitable result of these parallel paths is a completely predictable train wreck when the developers designing features that users don’t want crash into users who have been using Asterisk as originally designed. Additionally, note that the original proposal was to deprecate AMI/AGI in favour of ARI once it is feature complete with those protocols; an entirely lesser change than the removal of the dialplan in its entirety. So you're saying that deprecating the dial plan is not on the table? How then do you explain statements like this: Leif: we're in a transition, moving from dialplan model to external control model. Probably need external application to be built for us to move completely away from AMI/AGI.” or this Paul: take away apps, and whatever is in the core is what we should care about.” -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- 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 13 : SILK codec ?
Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? sean -- _ -- 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 13 : SILK codec ?
On Wed, Oct 29, 2014 at 5:16 PM, sean darcy seandar...@gmail.com wrote: Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? codec_silk for Asterisk 12 will most likely not work in Asterisk 13. A number of performance improvements in the media handling in Asterisk required some codec compatibility changes. I would expect said modules to be available in the next few weeks. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
