Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?
We've seen this issue in the past on and off with a variety of random users. We were never able to pinpoint the issue. We'll look into it again and see if we can find anything new. I've created a JIRA issue to track the issue for the moment. https://issues.asterisk.org/jira/browse/ASTERISK-27049 Any further comments or information should go on that issue. Thanks! On Mon, Jun 12, 2017 at 8:12 AM, Sebastian Gutierrez <scg...@gmail.com> wrote: > same here. > > >> >> >> >> On 12 June 2017 at 08:07, Olivier <oza.4...@gmail.com> <oza.4...@gmail.com> >> wrote: >> >> Hello, >> >> I'm a faithful reader of this mailing list, for several years now. >> >> Lately, I'm receiving emails asking me to re-enable my list >> subscription due >> to "excessive bouncing". >> >> What does this exactly mean and why am I receiving this ? >> Beside re-enabling my subscription, what can I do to improve things ? >> >> Regards >> >> >> -- Rusty Newton Digium, Inc. | Systems Support Administrator445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] this is so wonderful!
Removed from the list. On Sun, Mar 19, 2017 at 11:54 AM, kenc <k...@vipmarketing.org> wrote: > Dear friend! > > > > There is something really wonderful I wanted to show you, I hope you'll love > that stuff) Please take a look read more -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing RTP frame size
On Thu, Apr 7, 2016 at 11:04 AM, Jan Blom <jan.b...@peopleinteractive.se> wrote: > > Is this supposed to work? Any suggestions for workarounds? I believe so. That sounds odd. Hard to know without seeing the packet trace of the call. Which SIP channel driver are you using? I think you are safe to go ahead and file an issue report. Please include the sip.conf/pjsip.conf plus a packet capture and Asterisk debug log (be sure to get the DEBUG channel turned on in logger.conf) with correlating SIP trace. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Patched Res_Musiconhold.So module
On Sat, Nov 21, 2015 at 2:15 PM, <topdog2...@gmail.com> wrote: > Good day Asterisk users, > If this is the wrong place to post this, my apologies. > However, I'm trying to see where I can get a patch for the > res_musiconhold.so module. > I have an issue where if someone is placed on hold, or is placed in a > queue, after any announcement is played in the queue, or if someone is > put on hold, the call is resumed, then is put back on hold, if the > same music is still playing, then that same music I have is started > over from the beginning and doesn't continue from where it left off. > I saw somewhere, where you have to patch, or obtain, a newer version > of res_musiconhold.so. > I don't seem to have a newer version of he above module, nor can I > obtain a newer version. I also canot get into the asterisk issue > tracker, as it requires a client certificate for which I do not have. > I'm running Asterisk 1.8.28.2; on Linux, CentOS 6.7 32-bit. > Any help that can be provided would be appreciated. Hi! A few things: I think the fix and patch you are looking for is here: https://issues.asterisk.org/jira/browse/ASTERISK-24019 You are running an EOL verison of Asterisk[1] that left security maintenance on 2015-10-21 which means it could quickly become a security risk (or may already be) to run that version. Not only that, you are not up to date on the 1.8 branch which means your older version could already have security issues or critical bugs where fixes exist for them. In fact if you update to the latest 1.8 at least - it should contain the fix you are looking for. In regards to the issue tracker you do not need a special certificate. As is mentioned on the front page of the tracker [2] you need to create an account on signup.asterisk.org. Hope all that helps! Thanks. [1]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions [2]: https://issues.asterisk.org/jira/secure/Dashboard.jspa -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Error while compiling asterisk asterisk-1.8.32.3
On Sat, Nov 21, 2015 at 1:51 AM, Jayesh Labade <jayesh.lab...@gmail.com> wrote: > Hi, > > I encountered following error while compiling asterisk-1.8.32.3. I am > using Debian 8(Jessie) 64 bit version. > > make[1]: *** [chan_dahdi.so] Error 1 > Makefile:351: recipe for target 'channels' failed > make: *** [channels] Error 2 > > Detailed error attached in log file. I'm not sure what is going on there but I wanted to mention that Asterisk 1.8 is completely EOL, there will be no further fixes, even security fixes. For new installations you should use Asterisk 13 which is the most recent LTS. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Looking for a patched res_musiconhold.so Module for 32-bit Linux/Astersk
On Sat, Nov 21, 2015 at 11:29 AM, Daniel Chavez <topdog2...@gmail.com> wrote: > Good day Asterisk users, > If this is the wrong place to post this, my apologies. > However, I'm trying to see where I can get a patch for the res_musiconhold.so > module. > I have an issue where if someone is placed on hold, or is placed in a queue, > after any announcement is played in the queue, or if someone is put on hold, > the call is resumed, then is put back on hold, if the same music is still > playing, then that same music I have is started over from the beginning and > doesn't continue from where it left off. > I saw somewhere, where you have to patch, or obtain, a newer version of > res_musiconhold.so. > I don't seem to have a newer version of he above module, nor can I obtain a > newer version. I also canot get into the asterisk issue tracker, as it > requires a client certificate for which I do not have. > I'm running Asterisk 1.8.28.2; on Linux, CentOS 6.7 32-bit. > Any help that can be provided would be appreciated. Looks like you posted twice in the same day. I responded to your other thread. Try not to duplicate posts. Thanks! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] issue with bridgeConference
On Mon, Nov 2, 2015 at 3:16 PM, hadi <almarzuki2...@hotmail.com> wrote:
> I have configure bridgeConference. But im having some issue. I want to give
> the ability to the user when dialing from the Conference to hangup the call
> by sending dtmf tones without being hangup from the Conference. For example
> if the user call some person and that person not answering, the user has the
> ability to hangup the call by sending *9 and return back the Conference, and
> start calling again.
>
> Here is my dial plan:-
>
> exten => 200,1,Dial(SIP/200,,Hhg)
> exten => 200,n,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Hangup
> exten => s-CONGESTION,1,Congestion
> exten => s-CANCEL,1, Busy
> exten => s-BUSY,1,Busy
> exten => s-CHANUNAVAIL,1,Playback(switchoff)
> exten => s-CHANUNAVAIL,n,Read(number,,,sn)
> exten => s-CHANUNAVAIL,n,GotoIf($["${number}" = "9"]?106)
> exten => s-CHANUNAVAIL,106,SoftHangup(${EXTEN})
I suppose by bridgeConference you mean ConfBridge?
If you require assistance you'll need to describe more than what you
*want to do*. You'll need to describe the issue you are having.
Include dialplan and logs to demonstrate the issue.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] PJSIP with registratrion to DNS SRV records fail with PJLIB_UTIL_EDNSNOANSWERREC
On Wed, Nov 4, 2015 at 9:19 AM, Daniel Tryba <dan...@tryba.nl> wrote: > > I finally thought it might be a good time to start looking at the pjsip > implementation in Asterisk 13. But trying to register to a sip cluster > that uses SRV records fails randomly with: > > [Nov 4 15:50:59] WARNING[31330]: pjsip:0 : tsx0x7f075c006 Failed to > send Request msg REGISTER/cseq=17800 (tdta0x7f075c0058f0)! err=320047 > (No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC)) > [Nov 4 15:50:59] WARNING[31330]: res_pjsip_outbound_registration.c:735 > schedule_retry: No response received from 'sip:sip.itco.nl' on > registration attempt to 'sip:tr...@sip.itco.nl', retrying in '60' > > [Nov 4 15:51:59] WARNING[31330]: pjsip:0 : tsx0x7f075c006 Failed to > send Request msg REGISTER/cseq=17801 (tdta0x7f075c0058f0)! err=320047 > (No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC)) > [Nov 4 15:51:59] WARNING[31330]: res_pjsip_outbound_registration.c:735 > schedule_retry: No response received from 'sip:sip.itco.nl' on > registration attempt to 'sip:tr...@sip.itco.nl', retrying in '60' > > At 15:52:59 the register succeeds somehow. > > Attached is a pcap of the DNS request and the responses (capture filter: > port 53 or port 5060 or port 5061). Unlike the warning says the > responses are there. > > Does anybody have a hint of what is going on/what I do wrong? > > pjsip.conf: > [transport-udp] > type=transport > protocol=udp > bind=0.0.0.0 > > [transport-tcp] > type=transport > protocol=tcp > bind=0.0.0.0 > > [tryba] > type=endpoint > transport=transport-udp > context=tryba > disallow=all > allow=alaw > outbound_auth=tryba_auth > force_rport=yes > direct_media=no > ice_support=yes > auth=tryba_auth > > [tryba_auth] > type=auth > auth_type=userpass > password=** > username=tryba > > [tryba_register] > transport=transport-udp > type=registration > server_uri=sip:sip.itco.nl > client_uri=sip:tr...@sip.itco.nl > contact_user=tryba > outbound_auth=tryba_auth > expiration=180 > For those wandering web-searching souls: https://issues.asterisk.org/jira/browse/ASTERISK-25528 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] How to encode plus sign in REGEX function in dialplan?
On Thu, Nov 5, 2015 at 12:49 AM, Recursive <li...@binarus.de> wrote:
> Dear all,
>
> I have made a fairly complex dialplan where I am using the REGEX function in
> many places. This works so far, but I wasn't able to solve the following
> problem. What I would like to do is the following (please note that this is
> normal regex syntax and obviously not what the REGEX function expects, but I
> hope it shows the idea):
>
> same => n(A1), GotoIf($[${REGEX("^\+49.*" ${EXTEN})}]?:A2)
>
> This line should make Asterisk jump to label A2 if the extension begins with
> +49. Since the plus sign is a special char in regexes, I have escaped it with
> \ as usual. But that does not work; the pattern is not matched and the goto
> is not executed when the extension begins with +49.
>
> What I already have tried:
>
> 1) same => n(A1), GotoIf($[${REGEX("^\\+49.*" ${EXTEN})}]?:A2)
>
> 2) same => n(A1), GotoIf($[${REGEX("^\\\+49.*" ${EXTEN})}]?:A2)
>
> 3) same => n(A1), GotoIf($[${REGEX("^+49.*" ${EXTEN})}]?:A2)
>
> 4) same => n, Set(REPAT=^+49.*)
>same => n(A1), GotoIf($[${REGEX(${REPAT} ${EXTEN})}]?:A2)
>
> 5) same => n, Set(REPAT="^+49.*")
>same => n(A1), GotoIf($[${REGEX(${REPAT} ${EXTEN})}]?:A2)
>
> 6) same => n, Set(REPAT=^+49.*)
>same => n(A1), GotoIf($[${REGEX("${REPAT}" ${EXTEN})}]?:A2)
>
> 7) same => n, Set(REPAT="^+49.*")
>same => n(A1), GotoIf($[${REGEX("${REPAT}" ${EXTEN})}]?:A2)
>
> Neither of these worked.
>
> Actually, the REGEX function is not able to handle normal regular
> expressions. To make things worse, there doesn't seem to be any
> documentation. Could anybody please point me to documentation or tell me how
> write that very simple pattern?
>
> Thank you very much,
>
> Recursive
>
> P.S. This happens in Asterisk 13.6.0 - I haven't tested with other versions.
The documentation for that function is available at the CLI "core show
function REGEX" and is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_REGEX
It should be able to handle typical regular expression. I don't see
anything wrong with what you are doing. Please file a bug at
issues.asterisk.org/jira. Do include a debug log on the issue
captured when Asterisk attempts to execute these extensions.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Thanks,
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] important message
On Thu, Oct 22, 2015 at 2:19 PM, <brettl...@nemeroff.com> wrote: > Hello! > > > > New message, please read http://grillonwheelsnyc.com/told.php > > > > brettl...@nemeroff.com > > > -- > _ > -- 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 This user has been removed from the list. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Asterisk sip.conf insecure=port, invite - doesn't work
On Tue, Sep 29, 2015 at 8:24 AM, Jarek Jarzebowski <jarek.jarzebow...@gmail.com> wrote: > Hi all. > > I have asterisk with sip registered accounts (realtime). > Moreover I have SIP trunk defined as type=peer in sip.conf. > > When call is incoming from SIP trunk with CLID of one of sip friend defined > in MySQL sippeers table asterisk refuses INVITE as not authorized. > > I tried to use insecure=port,invite options under SIP trunk definition in > sip.conf but this not solves the problen. > > Could you point me what could be the solution? Is only a single SIP trunk behaving this way? Or does this happen with all of your trunks? What happens if you replicate the configuration in sip.conf instead of your database? Does it work there? Probably best to pastebin an Asterisk log including "sip set debug on" output so that we can see what is going on. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Fwd: ferie estive
On Mon, Aug 24, 2015 at 9:40 PM, Pete Mundy p...@fiberphone.co.nz wrote: Any chance the list admins could unsubscribe Mr Anzaldi until he gets his broken auto-responder fixed? He has been unsubscribed and alerted to the issue. Thanks! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] simultaneous use of chan_sip/chan_pjsip
On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka cerv...@fpf.slu.cz wrote: hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc You can use both.. you will want to make sure your bind addresses and ports don't conflict. Why not use chan_pjsip for all SIP connectivity? -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Recording INCOMING calls
On Thu, Jul 16, 2015 at 3:37 AM, Luca Bertoncello lucab...@lucabert.de wrote: Hi list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon = *3 then, in my dialplan: exten = 1,n,Dial(SIP/004935,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... Perhaps the incoming calls are routed through different dialplan and in that Dial you do not have the proper options? The dialplan you posted appears to be for dialing an explicit outbound number. In the console I can't see anything, too. You can't see *anything*? You may want to read up on logging configuration to make sure you have relevant logging channels going to your log file or console. https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration https://wiki.asterisk.org/wiki/display/AST/Logging Once you have verbose output going to a log, make sure it is turned up to 5 and then post the call output to the list. With that we'll be able to see what is happening. -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Recording INCOMING calls
On Fri, Jul 17, 2015 at 9:09 AM, Luca Bertoncello lucab...@lucabert.de wrote: Rusty Newton rnew...@digium.com schrieb: Perhaps the incoming calls are routed through different dialplan and in that Dial you do not have the proper options? The dialplan you posted appears to be for dialing an explicit outbound number. YES!! That was the problem! I just added xX to the previous Dial and all work! Awesome. Be aware that using both x and X will allow both parties on the call to start recording.. that means regardless of whether this is an inbound or outbound call that both the calling and the called party will be able to initiate recording (assuming they know the code, or press it accidentally). -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Asterisk 11 and pulse
On Wed, Jun 24, 2015 at 7:03 AM, Jerry Geis ge...@pagestation.com wrote: I am looking for some great instructions on using asterisk with pulse. I'm using centos 7 and pulse as a user and not having much luck. I have changed all permissions for the asterisk directories. set asterisk.conf user and group to be my user that is running. No go. Anyone done this? I'm not sure that your question is clear. You'll probably want to be more specific. What is pulse? You mention as a user, are you talking about voicepulse.com ? What are you trying to do with pulse? What problem are you running into? -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Custom header when busy
On Wed, Jul 1, 2015 at 4:46 AM, r...@yandex.ru wrote: Hi, all Is there someway ability to insert custom Header to SIP 486 message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment. I only know of the SIPAddHeader application which lets you add headers when used before Dial, so I don't think you can do this currently. -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Asterisk Tech/Eng Positions Open In Dallas TX
On Fri, Jun 19, 2015 at 4:42 PM, JR Richardson jmr.richard...@gmail.com wrote: We have a couple of positions open, please contact me off-list if interested. http://www.ntegratedsolutions.com/voice-engineer-dallas/ These are full time positions in Dallas, no telecommuters please. Thanks. JR Howdy, please don't cross-post onto the asterisk-users list with job postings. They are allowed on asterisk-biz, but not on asterisk-users. http://www.asterisk.org/community/discuss -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] SIP LDAP authentication
On Sat, Jun 20, 2015 at 4:19 PM, Owais Ahmad millennium@gmail.com wrote: Hello, Is there a definitive guide on how SIP peers could be authenticated using LDAP in asterisk 11 and up? It seems https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver is not updated as there are mis-matched parameters in the configuration samples and ldap schema files. This is because I get: *Command: *sip show peer 1000 load *Output: **ERROR*: res_config_ldap.c:1389 update_ldap: Couldn't modify 'name'='1000', dn:cn=1000,ou=x,dc=,dc= because Invalid syntax Any hints? In Chapter 18 of Asterisk the Definitive Guide there is a section on LDAP integration that might be helpful. If you find any errors in wiki documentation please comment on the wiki page or else file a bug report to let us know. -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Problem asterisk voicemail message records
On Mon, Jun 8, 2015 at 9:56 AM, Igor Potjevlesch igor.potjevle...@gmail.com wrote: Hello! I've got a little problem with Asterisk (11.14.1), the voicemessages are kinda limited to 40 seconds (average) aproximately; because when a message reach this long I got a cut in the file (*.wav) after I got this message: WARNING[15035][C-21ef]: format_wav_gsm.c:418 wav_read: Short read (20) (Resource temporarily unavailable)! Does anyone got this problem, any idea of what is happening? Thanks I don't see any similar issues in the issue tracker (that are still open or recently closed). You might gather a bit more debug (see https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information). Then pastebin that debug here. At least so that we can see the context for that message, that is what is going on right before and after it. If you can include level 5 DEBUG logger messages then that would be helpful. -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Find out or log negotiated codec for SIP channel?
On Thu, Jun 4, 2015 at 5:58 AM, Tony Mountifield t...@softins.co.uk wrote: Hi, despite some searching I haven't found an answer to this question: Is there a way I can see in the log, or find out in the dialplan, what codec has been negotiated for a SIP channel? If possible, I'd like to do this in both Asterisk 11 and in an old 1.2 system. What I'm specifically trying to do is to determine historically the usage of the G.729 licences installed in a system, but an answer to the more general question would be useful. Thanks Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org I believe core show channel channel name or sip show channel channel name may help you. From a log standpoint - there should be information on the codecs negotiated when you enable the DEBUG logger channel. You probably want to read through the logging documentation on the wiki. https://wiki.asterisk.org/wiki/display/AST/Logging https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Thanks, -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Fritzbox 7490
On Mon, Jun 8, 2015 at 12:00 PM, Christian christia...@runbox.com wrote: Hi, Sorry if off topic, but is anyone here on this list using it? I am currently searching for a good router for my home network wich supports SIP. Many thanks! You'll probably get more answers in a generic VOIP/SIP forum, but I'd recommend asking a more specific question. Almost any router will support SIP if you simply open the proper ports and setup the right forwarding. What kind of support you specifically looking for? Some routers have SIP ALG, but it isn't completely necessary and can sometimes be a hassle. I had to turn it off on my home ASUS RT-N56U as there was no configuration for it (it was even undocumented! yay!). -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] PJSIP Sends BYE with Wrong IP
On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard kct...@gmail.com wrote: Hello - I am trying to decide if I have stumbled across a bug in PJSIP or I am just missing something. My Asterisk has two interfaces, an internal eth0 and an external eth1. In pjsip.conf, I define the following transports: [trusted] type=transport protocol=udp bind=10.xx.yy.zz:5060 [untrusted] type=transport protocol=udp bind=12.4.aa.bb:5060 My internal endpoints use transport=internal and external endpoints use transport=external. I guess that's obvious. You show transports trusted and untrusted, you don't show any transports named internal and external... so that is confusing. Everything works fine, most of the time. INVITEs, 1XX, 2XX are sent to the right interface using the right source IP. But, when Asterisk tries to send a BYE to any internal endpoint, it sends using the external IP, but it is sent of the correct internal interface eth0. Only the IP layer is incorrect. The SIP layer has the correct IP in the Via header. From what I can tell, only BYE is affected. I didn't have this problem with chan_sip. Am I just missing some configuration? This sounds like improper configuration, or a bug. If you can pastebin a full (sanitized) pjsip.conf as well as an Asterisk log with verbose turned up[1], plus a SIP packet trace then we can take a look at it. [1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] PJSIP Sends BYE with Wrong IP
On Thu, Apr 2, 2015 at 11:07 AM, Trey Hilyard kct...@gmail.com wrote: I actually got the issue resolved by upgrading to 13.3.rc-1, since this is just my development system. I assume that the problem was resolved between the two releases. Sweet, glad to hear! -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] issue with inbound route
On Thu, Feb 26, 2015 at 10:34 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud i use elastix The best place to ask a question about Elastix configuration is the Elastix forums, http://forum.elastix.org/. The log output you show isn't enough to indicate the issue from what I can see. -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Asterisk does not listed to port 5060
On Mon, Feb 23, 2015 at 5:51 AM, Raj Roy Ghandhi roy.gan...@gmail.com wrote: Hi Friends, I encountered a strange issue. I am running Asterisk 11.8.1 on Cent OS with no firewall running. It has 3 NIC interfaces. in my sip.conf I have allowguest=yes bindaddr=0.0.0.0 udpbindaddr = 0.0.0.0 But my Asterisk instance does not pick the call at all. When I check the listening apps using lsof -i I get asterisk 3046 asterisk7u IPv4 1191172 0t0 TCP *:5038 (LISTEN) asterisk 3046 asterisk 10u IPv4 1191186 0t0 UDP *:sip asterisk 3046 asterisk 11u IPv4 1191187 0t0 TCP *:sip (LISTEN) asterisk 3046 asterisk 13u IPv4 1191196 0t0 UDP *:iax asterisk 3046 asterisk 15u IPv4 1191199 0t0 UDP *:commplex-main asterisk 3046 asterisk 16u IPv4 1191201 0t0 UDP *:4520 asterisk 3046 asterisk 19u IPv4 1191232 0t0 TCP localhost:5038-localhost:43353 (ESTABLISHED) But I van see the SIP Invite that comes into server and I can ngrep it as I believe UDP ports don't provide the state in lsof. Asterisk is listening here: asterisk 3046 asterisk 10u IPv4 1191186 0t0 UDP *:sip My system shows similar output for lsof and it works fine. Have you tried using the Asterisk CLI with sip set debug on to see if Asterisk shows any SIP packets? You might consider collecting a debug log with sip set debug on output : https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Once you have that, provide a pastebin link to the output and someone may be able to help you out. -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Connect Asterisk to WiFi
On Tue, Dec 23, 2014 at 4:17 PM, Joseph syscon...@gmail.com wrote: Are there any adapters that would allow me to connect asterisk to wifi or we are not there yet? I have Digium adapter S101i that was discontinued but similar device that would connect to wifi network and a cell phone would be handy. All you need is a Wi-Fi interface for your computer, or a wired Ethernet adapter for Wi-Fi. Such as any of these on amazon: http://www.amazon.com/b?node=13983791 Asterisk doesn't care whether it is wired or wireless, it is just talking via IP over your network interface. The s101i was an IAX to Analog adapter that also communicated with Asterisk over IP. Connecting a cell phone directly to Asterisk is a whole different story. Do you mean an actual cellular device such as a GSM cell phone? Or do you mean a phone that connects to a 802.11 Wi-Fi signal? -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Connect Asterisk to WiFi
On Tue, Dec 23, 2014 at 6:34 PM, Rusty Newton rnew...@digium.com wrote: On Tue, Dec 23, 2014 at 4:17 PM, Joseph syscon...@gmail.com wrote: Are there any adapters that would allow me to connect asterisk to wifi or we are not there yet? I have Digium adapter S101i that was discontinued but similar device that would connect to wifi network and a cell phone would be handy. All you need is a Wi-Fi interface for your computer, or a wired Ethernet adapter for Wi-Fi. Such as any of these on amazon: http://www.amazon.com/b?node=13983791 Asterisk doesn't care whether it is wired or wireless, it is just talking via IP over your network interface. The s101i was an IAX to Analog adapter that also communicated with Asterisk over IP. Connecting a cell phone directly to Asterisk is a whole different story. Do you mean an actual cellular device such as a GSM cell phone? Or do you mean a phone that connects to a 802.11 Wi-Fi signal? Just realized I linked to USB to Wi-Fi adapters, but that works too. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Connect Asterisk to WiFi
On Tue, Dec 23, 2014 at 6:51 PM, Joseph syscon...@gmail.com wrote: Most cell phone don't have a USB port but you are correct, maybe I just need IAX2 soft-phone like: Zoiper - it works on most of the platforms. I think Zoiper registers directly with Asterisk IAX2 (if configured) as an extension, isn't it? If your cellphone is capable of a Wi-Fi connection and you can find a softphone such as zoiper that will run on it, yes it could talk directly to Asterisk via IAX2 protocol. A softphone could also talk to Asterisk over other protocols such as SIP. The configuration will depend on whether you are trying to connect over a local wireless LAN, or from the Internet back to your server. Here is a tutorial that features zoiper: https://wiki.asterisk.org/wiki/display/AST/Hello+World The tutorial assumes you are running Zoiper on the PC, but setting it up should be similar. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] 11.5.0: blindxfer problems
On Wed, Dec 17, 2014 at 1:09 PM, sean darcy seandar...@gmail.com wrote:
I've got a confbridge set up which works if dialed locally:
-- Executing [266@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [266@internal:2] SendDTMF(DAHDI/1-1, 1) in new stack
-- Executing [266@internal:3] ConfBridge(DAHDI/1-1, 1) in new stack
-- DAHDI/1-1 Playing 'conf-onlyperson.ulaw' (language 'en')
...
extensions.conf:
[globals]
...
GOTO_ON_BLINDXFR=internal,266,1
features.conf:
[featuremap]
blindxfer = #1
But:
-- Executing [s@DialOut:14] Dial(DAHDI/1-1,
motif//+12345678...@voice.google.com,,rTt) in new stack
-- Called motif//+12345678...@voice.google.com
-- Motif/+12345678...@voice.google.com-688c is proceeding passing it to
DAHDI/1-1
-- Motif/+12345678...@voice.google.com-688c answered DAHDI/1-1
-- Started music on hold, class 'default', on
Motif/+1234567...@voice.google.com-688c
-- DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 17 09:46:59] WARNING[19083][C-00be]: features.c:2550
builtin_blindtransfer: No digits dialed.
-- DAHDI/1-1 Playing 'pbx-invalid.ulaw' (language 'en')
I'm expecting the blind transfer to GOTO internal,266,1.
If I input 266 at the transfer dial tone, the blind transfer occurs.
Do I have this set up incorrectly?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
${GOTO_ON_BLINDXFR} - Transfer to the specified
context/extension/priority after a blind transfer (use ^ characters in
place of | to separate context/extension/priority when setting this
variable from the dialplan)
Try using ^ characters as it mentions there.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] c option doesn't work if used with q option in meetme
On Fri, Dec 12, 2014 at 4:34 AM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, Asterisk version 11.13.1 I'm trying use realtime meetme application with c and q option. If both options are used together in meetme table under 'opts' field, c option (Announce user(s) count on joining a conference.) doesn't work i.e. system doesn't play user counting to caller. Is it bug or some configuration problem. This was fixed a while back: https://issues.asterisk.org/jira/browse/ASTERISK-17053 If it is an issue again then it is a regression. I'm not sure if realtime would make a difference. You probably want to file it as a bug. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines As the guidelines mention, be sure to include Asterisk logs with debug showing the issue. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Corrupt MixMonitor recordings - .gsm format
On Fri, Dec 12, 2014 at 12:31 AM, Stefan Viljoen viljo...@verishare.co.za wrote: snip The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. What changed? It is unlikely that Asterisk suddenly changed without outside intervention, unless there was some slow leak and a resource limitation was reached. Your troubleshooting below would have at least temporarily resolved that issue. snip Only one site has started producing corrupt .gsm files since last week. I've already replaced that server with a brand new one, reinstalled the operating system and Asterisk, problem still persists. If it is only the one site, and replacing Asterisk AND the entire machine with OS did not fix the issue, then I'd think something changed at the site outside your Asterisk server. I would look into what is different between VOIP traffic you are receiving on that site and other sites. Though, I really wouldn't know what to look for in this case. The only recent MixMonitor audio corruption issue I found reported was https://issues.asterisk.org/jira/browse/ASTERISK-24507 and it doesn't match what you are describing here, unless maybe you only do blind transfers on this site, and no other sites and you only started having the blind transfers 7 months into usage. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Dynamic Call parking
On Thu, Jul 3, 2014 at 7:49 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, I know now after some testing that there is no dynamic call parking. Also explains why you find no example when searching the internet : no one has a working example. Sorry to bump an old thread, but I wanted to make sure a link to this relevant issue was here for the archives: https://issues.asterisk.org/jira/browse/ASTERISK-24596 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] park()-command always parks on default 701
On Tue, Nov 25, 2014 at 8:27 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, I have the following in my dialplan : exten = callpark,n,Set(PARKINGDYNPOS=200-210) exten = callpark,n,Set(PARKINGDYNCONTEXT=parked_001) exten = callpark,n,Park(2s,parkinglot_001) I see on the CLI : [Nov 25 15:08:47] -- Executing [callpark@pbx-routing:10] Set(SIP/SipT01-000b, PARKINGDYNPOS=200-210) in new stack [Nov 25 15:08:47] -- Executing [callpark@pbx-routing:11] Set(SIP/SipT01-000b, PARKINGDYNCONTEXT=parked_001) in new stack [Nov 25 15:08:47] -- Executing [callpark@pbx-routing:12] Park(SIP/SipT01-000b, 5s,parkinglot_001) in new stack [Nov 25 15:08:47] == Parked SIP/SipT01-000b on 701 (lot parkinglot_001). Will timeout back to extension [pbx-routing] s, 1 in 50 seconds [Nov 25 15:08:47] -- Added extension '701' priority 1 to parked_77 Why does Asterisk park on 701 ? Why not on 200 ? I believe because you haven't set the PARKINGDYNEXTEN channel variable or defined a PARKINGDYNCONTEXT (template parking lot) with a parkext option defined. Therefore it defaults back to the parkext of the default lot. It is all sort of unclear at the moment unfortunately - https://issues.asterisk.org/jira/browse/ASTERISK-24596 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] google voice
On Mon, Nov 17, 2014 at 6:37 PM, George Wu aihu...@gmail.com wrote: anybody know the motif driver if the integration with google voice still work or not? What's the best way for the interop with google voice? https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google is the relevant documentation on the wiki. I haven't tried it myself in a long while, however Google was supposed to end XMPP support for GV back in May. I've heard mixed reports from community members. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Como unir webrtc con asterisk???
2014-11-12 16:24 GMT-06:00 Dario Estupinan darioestupi...@soygenial.co: tengo la siguiente pagina pero no se como seguir despues del punto 22 http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html gracias! You haven't described your problem and I'm relying mostly on Google translate, so I'm not sure what to tell you. There is another tutorial available at: https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 If you have an issue specific to FreePBX, you might try asking about it on the FreePBX community forums: http://community.freepbx.org/ Thanks, -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Erratic calls through NAT-ed server
On Thu, Nov 13, 2014 at 8:23 AM, Norman Laidla norman.lai...@telegrupp.ee wrote: Morning, We recently pushed our Asterisk video bridge into a DMZ and since then, local calls have been unreliable to say the least. While offsite calls work nicely, calls on our internal server usually fail to ring the far end. Two test calls that were made 4 minutes apart yielded different results: one rang the far end, the other kept trying to transmit the Invite. The configuration didn't change at all between the two calls. I've been going over the debug logs, but haven't noticed any possible reasons why one call failed. It's the same all the way to the part where the far end is called. The endpoints use different ports for UDP signaling and Asterisk is set to expect UDP packets from those ports. The RTP port range is the same between the ends (at least where it's configurable), Asterisk and the firewall. All ports that we're using have been opened in the firewall and incoming UDP traffic is routed to Asterisk. In Asterisk settings, localnet is defined as the LAN that both endpoints are on, externip is the public address of the server. Directrtpsetup and directmedia are both set to no and nat is set to yes. So, what could be causing this issue? If out of multiple calls, some work and some don't - you either have found a bug or something is really changing between the calls. That is assuming the failing/working behavior does not fit an obvious pattern (e.g. unique to a particular dialed remote party). If you pastebin two Asterisk logs that show the working and failing calls then someone may be able to look through them and spot an issue. Be sure the Asterisk logs show VERBOSE and DEBUG channels at level 5 or above. See: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information You might also mention the exaction version of Asterisk you are using and which channel driver (though it sounds like chan_sip based on the options described). -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] E1 - Cisco - Asterisk and vice verso
On Wed, Nov 12, 2014 at 10:02 AM, Dmitriy Sirant l...@skoda.com.ua wrote: Hi, Change my Dynastar E1 gateway to Cisco with E1 module, but can't make easiest dialplan. All my routing i made on asterisk, so i need that cisco all calls from E1 send via sip to Asterisk and all calls came from Asterisk by sip send to E1. From E1 to Asterisk already work, but can't understand how send all from Asterisk SIP to E1 ? Can you help ? If you are taking SIP calls into Asterisk and want to send them out E1, you need an Asterisk-compatible E1 board, such as: http://www.digium.com/en/products/telephony-cards/digital/single-span Your question isn't very specific, so if you are having problems with dialplan you will need to elaborate on Asterisk version, hardware drivers and Asterisk channel drivers in use, etc. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] howto cancel simultaneous calls - dial(sip/phone1sip/phone2)
On Fri, Oct 10, 2014 at 8:14 AM, Marek Cervenka cerv...@fpf.slu.cz wrote: hi. i have dialplan with 2 simultaneous calls - dial(sip/phone1sip/phone2). when i cancel call on phone1 (push reject button), the call is still ringing on phone2 can i cancel call on both phones from one place(one phone)? thanks The calls to each destination are separate call legs (channels). Rejecting the call from one phone will only cancel that one channel. This seems like something that should be easy to do, but I can't think of an easy way to do it at the moment. After talking with some others, I don't think there is a *simple* way to do it. If you are using external scripts with Asterisk APIs such as AMI or ARI, then you can probably accomplish what you want using those interfaces. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] conversation record prematurely
On Thu, Sep 18, 2014 at 7:14 PM, Joseph syscon...@gmail.com wrote: How do I find out/verify if b option is used with MixMonitor? In the help text for an application you'll find the available arguments and options. On the Asterisk CLI you can view an application's help text with core show application application name. You might be interested in checking out a wiki page (on this same subject) I added recently.[1] In addition, the help text for applications, functions and more gets added onto the wiki by scripts.[2] [1]: https://wiki.asterisk.org/wiki/display/AST/CLI+Syntax+and+Help+Commands [2]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Command+Reference -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] SIPAddHeader from a realtime databse
On Mon, Sep 22, 2014 at 9:43 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Guys I'm using asterisk 1.8.23.1 When I add a SIP Header from inside the extensions.conf (SIPAddHeader(Alert-Info:http://www.notused.com\;info=alert-internal\;x-line-id=0) ) it works fine. When I try to do the same thing from within a database table, all of the string apart from x-line-id=0 gets ignored. I've tried escaping the semicolon and not escaping it and the result is always the same, just the last part of the full string is expressed. Some of the ways that I have tried to enter the string are below: appdata='Alert-Info:http://www.notused.com\\;info=alert-internal\\;x-line-id=0' appdata='Alert-Info:http://www.notused.com;info=alert-internal;x-line-id=0' appdata='Alert-Info:http://www.notused.com;info=alert-internal;x-line-id=0' Does anyone know the correct format to store this in a DB table for it to be expressed correctly? I'm using MySQL. There is an existing report filed here: https://issues.asterisk.org/jira/browse/ASTERISK-19254 You can try Walter's suggestion on the issue and report back whether it works or not. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] conversation record prematurely
On Thu, Sep 18, 2014 at 3:16 PM, Joseph syscon...@gmail.com wrote:
I have following line in a context:
...
exten =
_587NXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten = _587NXX,n,MixMonitor(${recordfilename},b)
...
It records the conversation but it ends prematurely, after 10min. Why?
Where is the setting to records until a user hangup the handset.
Without further information the only reason I could see would be the
'b' option in use for MixMonitor. If the channels were no longer
bridged it would stop recording. That is according to the
documentation.. which every once in a while is wrong. Other than that,
it should record as long as the channel is bridged.
Can you pastebin a log showing that particular call?[1]
[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Record call ends in 10min
On Thu, Sep 18, 2014 at 5:00 PM, Joseph syscon...@gmail.com wrote: On 09/18/14 15:43, Joseph wrote: I don't have any maxduration set in any config file. maxduration: maximum recording duration in seconds. If missing or 0, there is no maximum. Joseph, please don't start new threads on a duplicate topic so quickly. It can create noise in the list and cause confusion when some people reply to your second thread vs your first. I've replied to your first thread. Re: maxduration, I believe that is an option for Record and not MixMonitor, but I could be wrong. Thanks, -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] ASTERISK AND CHAT MESSAGES
On Thu, Sep 11, 2014 at 10:46 AM, mohamed keita keitamohamed1...@gmail.com wrote: Hello. I have properly configured my asterisk server with some SIP accounts that work very well. Now when I want to send a chat message from my SIP client(Zoiper), I receive the following error message: WARNING [2578]: chan_sip.c: 14511 receive_message: Received Message to sip: @. transport = UDP from sip: @. transport = UDP; tag = b03dfe52, dropped it. .. Thank you in advance for your help. Mohamed, It is difficult to investigate an issue such as this without at least a few details about the software environment and what you are trying to accomplish. There isn't typically much we can do with just a single WARNING message excerpted from a log. For those that may be able to help you, you may want to provide at a minimum: * The exact version of Asterisk you are using. * The steps you took to produce the issue. * What you expected to happen and what actually happened. * What version of Zoiper you are using. * The destination of the SIP text message you are trying to send it to. (i.e. is this phone to phone, or phone to Asterisk, etc). Thanks, -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] chan_sip.c:23647 handle_request_invite: Failed to authenticate device
On Thu, Sep 11, 2014 at 10:14 AM, Deepak Bhatia dee...@voxomos.com wrote: Hi, Why are we getting message in the asterisk [Sep 10 12:55:23] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: Failed to authenticate device 601sip:601@111.118.185.107; tag=2f498fbd [Sep 10 12:55:24] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: Failed to authenticate device 601sip:601@111.118.185.107;tag=209a8aa9 Regards Deepak Bhatia Deepak, This is commonly a wrong password on the client-side. That is, the device attempting to call Asterisk is failing an authentication challenge for one reason or another. For anyone to help you, you would probably need to post your sip.conf configuration (full) and screenshots of your soft/hard client configuration. Obviously you would want to use a fake password and sanitize IP addresses if necessary. Thanks, -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Ast to Ast TLS trunk
On Wed, Sep 10, 2014 at 2:14 PM, Rizwan H Qureshi rizwanhas...@gmail.com wrote: Hi Everyone, How can I create a TLS based sip trunk between two asterisk servers. I have been trying to do it but i dont know how to defined the client certificate on the asterisk server. Has anyone tried this? There is a tutorial for secure calling with TLS and SRTP here: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Understanding local channels
On Mon, Aug 25, 2014 at 11:33 AM, Patrick Laimbock patr...@laimbock.com wrote: On 25-08-14 17:06, Mitch Claborn wrote: Can someone point me to a good tutorial / explanation of local channels? I've been using them without really understanding what is going on, and we all know how dangerous that is! I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but I'm just not quite getting it. How about the info on the Asterisk wiki: https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels On the left side there's a menu with examples and modifiers. HTH, Patrick It may also help to check out the section on Channels: https://wiki.asterisk.org/wiki/display/AST/Channels Before going into the Local Channel config section:https://wiki.asterisk.org/wiki/display/AST/Local+Channel If you can think of a way we can improve the documentation on Local Channels, let us know. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] help
On Fri, Aug 22, 2014 at 6:48 AM, chandapure shiva chandapure.shiv...@gmail.com wrote: Dear all, I was going through sip.conf file and i am not able to understand the working and how to test the functionality of below fields. 1.tcpauthlimit 2.tcpauthtimeout any inputs regarding this will appreciated, thanks in advance Do you have a specific question? What do you mean How to test the functionality of below fields? Here is the documentation on those options from the sip.conf sample file: ;tcpauthtimeout = 30; tcpauthtimeout specifies the maximum number ; of seconds a client has to authenticate. If ; the client does not authenticate beofre this ; timeout expires, the client will be ; disconnected. (default: 30 seconds) ;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of ; unauthenticated sessions that will be allowed ; to connect at any given time. (default: 100) -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Can't hangup channel from CLI
On Fri, Aug 22, 2014 at 6:00 PM, Steve Edwards asterisk@sedwards.com wrote: Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting Asterisk from a Tekelec T9000. I'm accumulating stuck channels. snip I haven't identified what callers are doing to reproduce the error reliably yet. Any clues or suggestions? You might see if they are getting stuck due to a deadlock: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace?src=search#GettingaBacktrace-GettingInformationForADeadlock If you get the traces required, you could open an issue on the bug tracker. If commands like core show channel channel and sip show channel channel work then you'll want to attach that data as well. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Certified Asterisk 11.6 Menuselect
On Mon, Jul 21, 2014 at 9:36 AM, Ryan Wagoner rswago...@gmail.com wrote: Has there been a change in the way certified Asterisk is being packaged? Starting with certified Asterisk 11.6 has all the extended options are checked by default in menuslect? Certified Asterisk 11.2 does not have them checked and neither does certified Asterisk 1.8.15? Thanks for taking note. I've filed an issue here https://issues.asterisk.org/jira/browse/ASTERISK-24104 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Native architecture never available in menuselect
On Mon, Jul 21, 2014 at 12:36 AM, CDR vene...@gmail.com wrote: I want to compile Asterisk always for the native architecture of the machine, and I find that it is never available. It says Depends on: native_arch(E) Can use: N/A Conflicts with: N/A Support Level: core This is Fedora 20 gcc (GCC) 4.8.3 20140624 (Red Hat 4.8.3-1) many thanks Philip I've rarely seen a machine that it isn't available on. The exception for me was a virtualbox machine in one particular case. You may get more help if you describe more about the CPU/architecture that the machine uses. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] CDR(dst) not set in AEL macro
On Fri, Jul 11, 2014 at 8:38 AM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi I'm using a macro to dial in a AEL dialplan. The problem is the macro do not set the field CDR(dst), showing only ~~s~~. I tried various configurations, but without solutions. Please don't post new duplicate threads in such a short time span. You posted this already yesterday. You'll have to be patient and wait for someone to respond. Thanks, -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 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] Pickup problem
On Wed, Jul 9, 2014 at 3:55 PM, Massimo Nuvoli mass...@archivio.it wrote: I found a very strange proble whit two asterisk servers in the same network. Scenario Asterisk A with extensions 5XX Asterisk B with extensions 2XX There is NO link between the two asterisks. Call from 501 to 503, 503 ringing Call from 201 to 203, 203 ringing The 202 extension comand a pickup (i dont manage this Asterisk, i think with the Pickup command). The 202 answer the 501 call and not the 201. snip Your description of the issue doesn't make any sense. You seem to be describing an extension on one Asterisk system using call pickup to pickup a call on another Asterisk system with no connection between the two. There is nowhere near enough information here to tell what is going on. Can you post an Asterisk full log showing the extension performing a call pickup? With that information someone may be able to help you. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Need a developer to write me a patch
On Thu, Jul 10, 2014 at 9:08 AM, CDR vene...@gmail.com wrote: I cannot wait for the regular bug-patch process to play out. I am considering hiring a developer to fix bug 24015, and of course submit the patch for the bug. The issue is simple, the app Transfer does not transfer when using PJSIP.. I called Digium and they said that they do not do this kind of work. The Asterisk users list is not intended for this sort of post. You can post on the asterisk-biz list if you are posting a job offering or looking for consultants. Otherwise bug bounties are accommodated on the dev list: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] PJSIP Include not working
On Thu, Jun 26, 2014 at 12:30 PM, CDR vene...@gmail.com wrote: I did what we use to dim that is add a line to pjsip.conf like #include /etc/asterisk/pjpeers.conf but the file is not loaded. Am I doing something wrong this functionality is disabled? Nope, this should work fine. You might have to use quotes with the full path. I cannot remember this second. Try #include /etc/asterisk/pjpeers.conf Also it is funny to use peers in the file name since there is no peers concept in res_pjsip. :) -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Changing recorded file storage directory.
On Fri, Jun 27, 2014 at 3:12 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 26 June 2014 15:42, Anurag Rana anuragrana31...@gmail.com wrote: Hi All, In asterisk, default directory to store the call-recording files is /var/spool/asterisk/monitor. Can we change this directory? How? Hi You can specify the full path when doing the Monitor or MixMonitor application. You can change the spool directory in your asterisk.conf but this will move all the directories that normally live under /var/spool/asterisk For reference, you can also find this information in the Asterisk documentation at https://wiki.asterisk.org/wiki/display/AST/Directory+and+File+Structure -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] OPTIONS Request without username - Forbidden
On Wed, Jun 25, 2014 at 9:30 AM, Rafael Visser visser.raf...@gmail.com wrote: Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is 403 Forbidden. Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Is it wright? How can i instruct FREEPBX to send the username in the option request? It may be worth asking on the FreePBX forums at http://community.freepbx.org/ as the Asterisk users who use FreePBX are generally monitoring that community. Many people here won't be able to answer your question *within the context* of FreePBX configuration. Your question is also not clear. You should ask the provider specifically which header and where in what URI they want to see the username in. If this wasn't FreePBX I'd tell you to just try setting the callerid and fromuser options for the corresponding SIP peer. I don't want to pretend to know FreePBX, so I still recommend you go ask on their forum to get better assistance. Good luck! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Popup URL for outgoing calls.
On Sat, Jun 21, 2014 at 5:57 AM, Inventions resea...@businesstz.com wrote: Can anyone tell me how to implement a popup URL native asterisk when making outbound call? For example, a user (A Part) dial from a softphone number 07112233, when a call is received (or even before) by B-Part, a CRM pops up with information for user 07112233 on A-Part computer. More less like incoming url popup on a queue. No one is going to do the work for you. You'll have to do the research. A good place to start is probably the sections in the Asterisk Definitive Guide on Asterisk Gateway Interface and Asterisk Manager Interface http://shop.oreilly.com/product/0636920025894.do -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Request for spandsp paid support
On Tue, Jun 17, 2014 at 8:26 AM, Ramesh Hegde rame...@gmail.com wrote: Hello Does anyone know if there is anyone/any company which does paid support for spandsp? We are looking for such a company/individual who will support spandsp based on on specific Service level agrements Regards Ramesh Hi! Please use the asterisk-biz mailing list for commercial/paid support/job posting type discussions. http://lists.digium.com/mailman/listinfo/asterisk-biz -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] quickstart
On Tue, Jun 17, 2014 at 2:04 AM, thufir hawat.thu...@gmail.com wrote: I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. Once you install asterisk: https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk Try following: https://wiki.asterisk.org/wiki/display/AST/Hello+World Simply use a hard phone instead of a soft-phone. Then go from there on to two phones. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] maxsecs not working
On Fri, May 30, 2014 at 7:42 AM, David Cunningham dcunning...@voisonics.com wrote: Hi Rusty, We found the problem - a configuration error. Thank you for the response. I'm glad you found the issue! No problem. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] voicemail with odbc
On Thu, May 29, 2014 at 3:33 AM, ProNek pro...@gmail.com wrote: Hi, I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not understand database functionality on asterisk fully. The most suspected area is func_odbc. I already googled but not luck. Your guide is warmly welcomed snip You already started another mailing list thread on this topic a few hours before this. Please don't do that in the future. If you are going to post again, just post to the thread you already started instead of starting a new one. Did you double-check your database table carefully against the required schema? https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] maxsecs not working
On Thu, May 22, 2014 at 6:22 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, We have servers running Asterisk 1.8.20.1 and 11.7.0, and on both setting maxsecs in voicemail.conf doesn't seem to have any effect. A voicemail keeps recording after the specified time, and when the caller hangs up the voicemail is saved in the mailbox. Are we doing something really silly? snip Nope that configuration looks fine and it works on my systems as expected in the latest of those branches. Using your configuration I tried changing the maxsecs value and it appears to be respected. If you can reproduce the issue and provide debug to demonstrate, then you might file a bug report. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] How to enable DTLS
On Mon, May 19, 2014 at 11:58 PM, bhavik patel bhavikpatel14...@gmail.com wrote: Hi All, Currently i am integrating webRTC demo. I have issue using firefox,someone suggest me to enable DTLS for webRTC working in firefox using Asterisk. I am using Asterisk 11.9.0. https://groups.google.com/forum/#!searchin/doubango/bhavik/doubango/Mv9u0YkNb90/55VElJ1TdY8J Can any one tell me how to enable DTLS ? I haven't setup a WebRTC environment with DTLS, but you can find a section in the sip.conf.sample file starting with ; DTLS-SRTP CONFIGURATION that has descriptions of all of the DTLS options. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Asterisk 11.9 with webRTC demo integration
On Sat, May 10, 2014 at 2:27 AM, bhavik patel bhavikpatel14...@gmail.com wrote: Hi All, snip For Outbound calls : when i am dialling 8002 - 8001 every time Chrome Browser asking for allow microphone. Is there any way to disable asking permission and allowing it by default ? when i allow microphone then SIpml5 phone showing like Not Allow. That is a question about Chrome, not about Asterisk. A quick Google search pulls up this information: https://support.google.com/chrome/answer/2693767?hl=en If you select Allow on a http URL your preference will not be remembered in future visits. If you select Allow on a https URL, your preference will be remembered in future visits. snip Here is the asterisk logs : http://pastebin.com/JZeDjyay For Incoming calls : When call come to browser,And allow microphone then Call rejected and asterisk showing like Got SIP response 603 Failed to get local SDP in asterisk CLI. But After some google i found new link https://code.google.com/p/sipml5/wiki/Downloads for SIPml-api.js and after replacing that JS File Calls are comming in browser even i am able to answer that calls,Also in browser it says In call but in asterisk CLI it keep showing ringing and other end showing like remote ringing . Not sure what is going on here. You can try following my tutorial for testing with the SIPML5 demo here: https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 , It also uses Asterisk 11 and chan_sip which matches what you are doing. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, May 7, 2014 at 2:46 AM, Administrator TOOTAI ad...@tootai.net wrote: I got it: if the filename is given in totality it's working (as you do it). It's the #include /path to directorie/*.conf which is not taking in account (here *.conf description) That still works for me as well. I switched to Asterisk 11.9.0 built fresh from a tarball with default compilation options. I contructed a basic sip.conf, and added this line to the end: #include /etc/asterisk/sip_includes/*.conf I then edited /etc/asterisk/sip_includes/sip_included.conf to add a SIP peer. Started Asterisk. Then edited only the sip_included.conf file to change the peer name. Connected to Asterisk console, performed 'sip show peers', 'sip reload', 'sip show peers'. Everything worked as expected: here is a pastebin: http://pastebin.com/XGKKu4x9 That is, when using a wildcard in the file path in an include inside sip.conf, Asterisk correctly detects a change in the included conf file upon a sip reload. You might try reproducing the issue on a fresh install, on your non-production system to see if you can narrow down where the difference is. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, May 7, 2014 at 10:14 AM, Administrator TOOTAI ad...@tootai.net wrote: Le 07/05/2014 16:50, Rusty Newton a écrit : I contructed a basic sip.conf, and added this line to the end: #include /etc/asterisk/sip_includes/*.conf Here is the point. Modify it the way explained in previous message, like #include sip_includes/*.conf You should face the problem. And if you run it twice in a raw, it will do nothing the second time. Unfortunately, no. I went ahead and tried this as well. I still get working behavior even when using #include sip_includes/*.conf -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, May 7, 2014 at 12:59 PM, Administrator TOOTAI ad...@tootai.net wrote: Please try the includes *exactly* as I have them in sip.conf (same directories name and subdirectories) knowing that local is in /etc/asterisk I used your identical config to narrow it down. I re-opened https://issues.asterisk.org/jira/browse/ASTERISK-23683 and edited the Summary and Description fields, as well as linked it an issue where the fix for that issue *may* have introduced the problem you found. Thanks! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI ad...@tootai.net wrote: snip As explained in one on my previous message, it's a bug, easily reproducible: take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like this (what is important is the #include): snip NOTICE[3346]: app_queue.c:6811 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. despite the fact that modification was done in a .conf file. I took this example as with module reload app_queue the above message appears. For sip, iax, voicemail, aso there is no message, just SIP reload or ... To make asterisk take the modification in account, you have to open /etc/asterisk/[sip|iax|voicemail|queue|..].conf and save it without making any change. After this the command will be execute. It you run it a second time in a raw, you will see that the false behavior appears again till you again open/save the original file. Hi! I tried to reproduce using your description here and could not reproduce the issue. I tried with both sip.conf and queues.conf. Making a change in an included .conf file, but NOT the parent .conf file and then reloading that module from the CLI results in: centosclean*CLI module reload app_queue.so -- Reloading module 'app_queue.so' (True Call Queueing) [May 6 17:51:39] NOTICE[16211]: app_queue.c:7765 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. == Parsing '/etc/asterisk/queues.conf': Found == Parsing '/etc/asterisk/queue_include_1.conf': Found == Parsing '/tmp/queue_include_2.conf': Found I get the same behavior with sip.conf, it appears to work fine, whether I'm making only changes in the parent .conf or the included children. I even tried with two different included files in each sip.conf and queues.conf, one in /tmp and one in /etc/asterisk. Same working behavior. I used SVN-branch-11-r413305, so you might want to test there. However I'm still confused as to how you are seeing the behavior you are seeing. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Other Allison prompts?
On Fri, May 2, 2014 at 6:48 AM, Tony Mountifield t...@softins.co.uk wrote: I was wondering whether there are any other collections of Asterisk- compatible prompts recorded by Allison, that people might have kindly made available for free download. I found mention of some in an article at http://www.venturevoip.com/news.php?rssid=2690 but the links referred to on www.asterisk.org appear no longer to exist. These sounds were integrated into the 1.4.12 version of our Extra sounds release. I already have the core- (good quality) and extra- (poor quality) sets of standard prompts. On a related note, the extra- set appears to have been converted from the old GSM format. Are there any plans to have them re-recorded in good quality? Not at this time, you are welcome to contact Allison and ask her if she would be willing to do that. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] asterisk's internal database
On Tue, Apr 29, 2014 at 1:31 AM, binary dreamer dreamer.bin...@gmail.com wrote: i would like to read information from a file (txt) There are a few applications and functions that may help you out. In Asterisk 10 or before try the ReadFile application. Otherwise in 11 or beyond I believe you want the FILE function. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ReadFile https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FILE You could also use the SHELL function to execute a command on the system and capture output https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SHELL Then you might also look into Asterisk Gateway Interface for more complex tasks and control. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_AGI http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/AGI.html If you are learning Asterisk; the book I linked above contains some great information on AGI and is written by some of our notable community members. The latest edition is available here: http://shop.oreilly.com/product/0636920025894.do -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Asterisk call forward for T1 incoming calls
On Fri, Apr 25, 2014 at 9:19 AM, Al lists asteris...@gmail.com wrote: Is there a way to divert incoming calls on DAHDI T1 channels so telco gets the diversion and send the call to new number and releasing the channel? I'm no PRI expert, but I do remember from my time working with the T1 interface cards that two B-channel transfers did something like this. Digium has documentation on that here: http://kb.digium.com/articles/Configuration/Two-B-Channel-Transfers If that doesn't help, and you have a Digium card; you might call Digium tech support to ask about it. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] asterisk's internal database
On Thu, Apr 24, 2014 at 6:34 AM, binary dreamer dreamer.bin...@gmail.com wrote: hello everyone. I am running plain asterisk and I am using asterisk's internal database for: -phonebook -blacklist numbers instead of having to update the database of new entry or delete an entry, is it possible to have it in an external file such as txt? so every new entry/deletion will take place there. Are you wanting to swap out Asterisk's internal database with a different data storage interface? If so, that isn't possible as far as I know. If you are wanting to just read information from a file into Asterisk variables.. there may be other ways to do what you want. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] WebRTC and JsSIP
On Wed, Apr 16, 2014 at 1:35 PM, Consultor VOIP v...@axys.com.br wrote: Hi ! My name is Gerald and I am working with WEBRTC and JsSIP. I configure my Asterisk 11.7.0 to work wit WEBRTC. Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at the Asterisk, but when we try to make a call they send a 488 response and finish it. here is the part of the SIP DEBUG We can't do much with part of your debug. You'll want to post a pastebin link to your full SIP trace, and be sure that it includes at least VERBOSE messages turned up to 5.[1] Work on WebRTC support is on-going, so you'll want to test in the very latest Asterisk version in your branch (11 or above). That means you need to be on 11.9.0-rc2[2] at this moment. [1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information [2]: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.9.0-rc2.tar.gz -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] CLI command to see if SRTP is active?
On Fri, Mar 28, 2014 at 7:39 PM, Patrick Laimbock patr...@laimbock.com wrote: Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. I don't have any encrypted calls up in front of me at this second to provide an example, however if you are just wanting to verify SRTP is active for a call you will see SRTP related messages on the CLI if you turn up DEBUG message verbosity and have it going to the console, or else possibly with output from rtp set debug on. As for the show channels type commands, it may say something about encryption rather than SRTP directly. I'll take a look later if I get a chance. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] CLI command to see if SRTP is active?
On Mon, Mar 31, 2014 at 1:26 PM, Rusty Newton rnew...@digium.com wrote: On Fri, Mar 28, 2014 at 7:39 PM, Patrick Laimbock patr...@laimbock.com wrote: Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. I don't have any encrypted calls up in front of me at this second to provide an example, however if you are just wanting to verify SRTP is active for a call you will see SRTP related messages on the CLI if you turn up DEBUG message verbosity and have it going to the console, or else possibly with output from rtp set debug on. As for the show channels type commands, it may say something about encryption rather than SRTP directly. I'll take a look later if I get a chance. I see your issue on the tracker now, posting the link here for the sake of those who read the archives https://issues.asterisk.org/jira/browse/ASTERISK-23564 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Default extension
On Wed, Mar 26, 2014 at 1:14 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hello, When I get a SIP INVITE as follows: INVITE sip:s@10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18 To: sip:02XX@IP:5060 Contact: sip:1053212@IP:5060 Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 252 Asterisk considers that the extension is 's'. (The Register) How to make the extension number that is shown in the 'To' ?? What version of Asterisk are you using? It would help to show how you are performing the dial in dialplan or otherwise. If you are dialing a user/peer present in sip.conf or a database then show that configuration as well. Based on that someone could make a suggestion. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Asterisk 11.8.0 and 11.8.1
On Tue, Mar 25, 2014 at 10:13 AM, Jerry Geis ge...@pagestation.com wrote: OK back in the office so I can get some files in my confbridge.conf file [MessageNetConfUserMuted] type=user quiet=yes startmuted=yes announce_only_user=no announce_user_count_all=no announce_join_leave=no This is from the console so 410 and 411 are joining as MUTED. snip This all worked fine 11.0 - 11.7. I have only encountered problems with 11.8+ dropping back to 11.7 works again. So how do I found out if its something I have wrong or was this introduced in 11.8+ Hey Jerry, when something like this occurs suddenly between minor release versions, you can always check issues.asterisk.org/jira to see if it has been reported. A search for the words muted and confbridge, then ordering the results by creation date will show this issue: https://issues.asterisk.org/jira/browse/ASTERISK-23461 Which looks like the same issue that you are having. If you click on the source tab you can see the commits it was fixed in. Looks like it was after 11.8.1, so you'll have to wait until the next release, or grab 11 from SVN. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] calls processed value definition
On Mon, Mar 24, 2014 at 8:45 AM, Mitch Claborn mitch...@claborn.net wrote: The core show channels verbose command shows a calls processed value. Mine is currently 1928273. Exactly what does this figure represent? How is a call defined in this context? The simple answer is, that count is not useful, as a call is very ambiguous and Asterisk doesn't really have a definition for a call. The count is only useful to a developer who knows exactly what that count is tracking. The long answer is to read the comments from Matt Jordan on this issue: https://issues.asterisk.org/jira/browse/ASTERISK-21384 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Any help Address 0xfffffffe out of bounds in app_confbridge.casterisk-11.5.1 using confbridge.conf
On Thu, Mar 13, 2014 at 4:18 AM, hkc323 hkc...@gmail.com wrote:
Address 0xfffe out of bounds why and how to
solve.MyConfbridgeCount(conferencenumber,variablename )return total number
of user in conference given by conferencenumber otherwise zero.At runtime
using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call
function count_exec(struct ast_channel *chan, const char *data).But at
compile time char * data cause core dumped. Asterisk-11.5.1 Centos6
app_confbrige.c confbridge.conf
=
Task: Using Dailplan user want to retrive no of user in conference
'6050' = 1. Verbose(3,testMyConfbridgeCount) [pbx_config] 2.
MyConfbridgeCount(4000,count) [pbx_config] 3. verbose(3,== ${count} )
[pbx_config]
Please discontinue spamming the users list with your posts.
Not receiving an answer to your question is not a reason to repeatedly
post (four posts now in the past few days?)
I've already responded to your original post and asking you to post on
the issue tracker and follow the issue guidelines to provide the
information needed to investigate the crash.
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
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Re: [asterisk-users] strange records in cdr
On Thu, Mar 13, 2014 at 7:58 AM, Игорь Гайсин igor.gaj...@tts.tv wrote: snip mysql select calldate, clid,src,dst,dcontext,channel,lastapp,duration,billsec,disposition,amaflags from cdr where calldate like '2014-03-13 09:56:04'; +-+-+-+-+---++-+--+-+-+--+ | calldate| clid| src | dst | dcontext | channel | lastapp | duration | billsec | disposition | amaflags | +-+-+-+-+---++-+--+-+-+--+ | 2014-03-13 09:56:04 | 100 100 | 100 | s | from-sip-external |SIP/ip-00065fd2 | Answer |0 | 0 | ANSWERED|3 | +-+-+-+-+---++-+--+-+-+--+ What is clid 100 100? Why it came from? No this source into log. That is the Caller ID information for that channel. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] asterisk11.5.1 module not load why ? any help
On Fri, Mar 7, 2014 at 12:38 AM, hkc323 hkc...@gmail.com wrote: === Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. For the developers who can interpret an Asterisk backtrace, they often need more information than just the backtrace itself. If you can reproduce the crash on the latest version of the 11 branch then you'll want to follow the guidelines here https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines and post the issue on the bug tracker. Be sure to provide instructions and configuration that would allow us to reproduce the issue. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Asterisk 11, CEL and ConfBridge recordings
On Fri, Mar 7, 2014 at 7:21 AM, Jairo jairomolin...@gmail.com wrote:
Thank you very much Rusty.
It really works. Even if ${MyCustomFileName} gets a different value when the
second participant enters the conference, the filename remains the name
defined when the first participant enters (because he started the
conference).
Another thing, if I need to know the time a conference ended should I use
CEL or is there another better approach?
If you can get it from CEL, there is that, otherwise you can track
when you receive the AMI event ConfbridgeEnd
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_ConfbridgeEnd
That is all I got from poking around the docs. :)
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direct: +1 256 428 6200
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Re: [asterisk-users] Asterisk 11, CEL and ConfBridge recordings
On Wed, Mar 5, 2014 at 1:30 PM, Jairo jairomolin...@gmail.com wrote:
Dear friends,
Need to know filenames of conference recordings in Asterisk 11.
Besides directory scanning the recordings could use CEL:
Filter MySQL rows with eventtype equal CHAN_START and channame like
ConfBridgeRecorder and then get the eventtime field and convert to timestamp
to complete filename(s).
Would you suggest any other approaches?
You might set the record file path yourself through the CONFBRIDGE
function, for example, in dialplan:
...stuff up here to build a unique file name into MyCustomFileName...
exten = 1,n,Set(CONFBRIDGE(user,record_file)=${MyCustomFileName}.wav)
Then of course you now know the file name so you could do whatever you
wanted with it afterwards.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CONFBRIDGE
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direct: +1 256 428 6200
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Re: [asterisk-users] Cancel a ringing SIP call when the other party disconnect
On Fri, Feb 21, 2014 at 10:55 AM, Ruddy Gbaguidi plugwo...@micnes.com wrote: Hi, Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged). But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with the qualify. Is there any setup or asterisk configuration I need to enable to have A close its call ? Note: when A is already talking with B, the call is hanged up on rtp timeout. But not during the Ringing phase. I'm not sure it is possible to configure Asterisk to hang up during the ringing phase when a peer/endpoint becomes unreachable. I don't see an option or parameter for that behavior. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Asterisk NAT
On Fri, Feb 21, 2014 at 11:13 AM, Gholamreza Sabery gr.sab...@gmail.com wrote: Dear Mr. Newton Thank you for your response. I red the wiki and sip.conf sample file and I know about directmedia option. Actually these options are for times that you know about your connected networks (you know which clients are behind NAT and which are not). But my configuration is different. I have an A2Billing From my understanding and the documentation, the intent with directmedia=nonat is that it will act like directmedia=yes if the peer is detected as *not* being behind NAT, and directmedia=no if the peer is detected as being behind NAT. This implies that the administrator would not know ahead of time what is needed, otherwise seemingly you would just use yes or no. However I'm still not sure that will be helpful for your particular scenario. users. Here I want Asterisk to automatically detect when two users are behind the same NAT and redirect their traffic inside that NAT; this way the load of RTP traffic on Asterisk server will be reduced. I don't know that this is possible with any simple Asterisk configuration. Good luck! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Dynamically setting from domain when calling friends
On Thu, Feb 20, 2014 at 3:45 AM, Torbjörn Abrahamsson torbjorn.abrahams...@gmail.com wrote: I tested SIPFROMDOMAIN, and it worked. Important thing to note is that I needed to have at least one underscore at the beginning of the variable, as your example did, it needs to be inherited at least one level. I don't really see way this should be needed, shouldn't Dial be able see it in the channel that executes the application? Maybe this should be noted on the wiki as well, to avoid this kind of confusion? Glad to hear it worked for you! Information on variable inheritance is already on the wiki. Here https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance+Basics and here https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance Those two pages and their sub-pages have some overlap and may need to be consolidated. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Asterisk NAT
On Tue, Feb 18, 2014 at 10:53 PM, Gholamreza Sabery gr.sab...@gmail.com wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? Hi! As many others mentioned, if you don't get an answer, first go googling then try the #asterisk IRC channel, or maybe the forums at forums.asterisk.org. I noticed your first post today and was going to answer it there, before I saw this new post as well... To attempt answering your question... I believe so. The NAT section of the sip.conf sample contains a lot of helpful options, including: ;directmedia=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being ; sent is known to not be behind a NAT (as the RTP core can ; determine it based on the apparent IP address the media ; arrives from). That is for chan_sip in Asterisk 11, and should also be available in Asterisk 1.8 I've not used a config with this option before, but it sounds like the intent is what you may need. A link to the sample file (that is also included with your source files) http://svnview.digium.com/svn/asterisk/branches/11/configs/sip.conf.sample?view=markup -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Asterisk NAT
On Wed, Feb 19, 2014 at 2:55 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Wednesday 19 Feb 2014, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? No answer on the list probably just means the question was answered before; so your best bet is to search the mailing list archives and the wiki at http://voip-info.org Eventually, you will have been yomping around in Tech Land for long enough to graduate from ignorant tourist to seasoned traveller -- and then you get to ignore noob questions yourself. Or set yourself up as a tour guide, if you feel that way inclined :) It is worth nothing that the official Asterisk wiki is at http://wiki.asterisk.org. If there is something missing from there, feel free to let me or someone in #asterisk-dev know and we'll make sure things get updated. One thing I do have on my to-do list is a NAT guide. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Asterisk as a client: can I get the remote SIP server to ignore rport?
On Wed, Feb 19, 2014 at 11:53 AM, Markus unive...@truemetal.org wrote: Hi list, I have a fresh install of Asterisk 12.0.0 and I'm going to use it only as a client. I'm trying to SIP REGISTER with a remote SIP provider. The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of the VM performs static NAT from the RFC IP address to a dedicated public IP address, however, they are rewriting ports at will. That's the problem. Here's an excerpt from tcpdump: snip I'm thinking the answer is no, but is there any option how I can get the remote SIP provider to answer me on port 5060? Without having them to change anything in their config. http://www.ietf.org/rfc/rfc3581.txt To force RFC3581 support for outbound REGISTER messages, you can set nat=force_rport in the general section of your sip.conf. (This also forces RFC3581 compliance for inbound messages, for any peers that inherit this general option) In my testing this results in the outbound REGISTER setting rport like Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK493d3405;rport If the far end supports RFC3581 it should respond back to the port from what it received requests. That should be 5060 if they receive the message from 5060. However.. if your VM/Network provider is rewriting things, then they could potentially remove your rport value, send it out over a different port, or do any number of crazy things. Depending on what they have going on in their network, receiving it back at 5060 is no guarantee it'll get back to your Asterisk VM. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Dynamically setting from domain when calling friends
On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
torbjorn.abrahams...@gmail.com wrote:
I have a problem where I would like to be able to send an arbitrary SIP
domain when sending a call to a registered friend. By default the from
domain is set to the IP of the Asterisk server, but I would like to set it
to something else.
The case is that when a call from a foreign domain comes in to the Asterisk,
it will connect it to the callee (but with the domain changed). When the
callee wants to make a redial from call history, the domain will not be
correct.
I could probably do something with the fromdomain setting of the friend, but
I would like it to be dynamic, ie not having to update the friend definition
every time a different domain is used.
I understand that I would need to use outbound proxy in the client to
prevent it from dialing the domain directly.
Is this possible? Any alternatives?
I'm a little confused about what you want to do, however I'll throw
some information at you in hopes that it will help out.
I did a little research and found that you can set the outbound From
header domain and From header user through two channel variables:
SIPFROMDOMAIN, SIPFROMUSER
They are sparsely documented, but there is an example in extensions.conf
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial)
; check if we set the FREENUMDOMAIN global variable in [global]
same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
;if we did set it, then we'll use it for our outbound dialing
domain
It looks like they were added in 1.6.2.X of Asterisk, so if you are
using 1.8.X or above, you should have them.
On your inbound call, you could use the function SIP_HEADER[1] to
gather the domain and store it for later use when you want to set it
on the outbound call. Though I'm not sure how you could tell that the
call was a redial.
[1]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER
I'm assuming when your SIP client redials that it calls through
Asterisk and is not dialing the previously caller directly.
Hope any of that helps. *Goes off to document SIPFROMDOMAIN and
SIPFROMUSER on the wiki*
--
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Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Dynamically setting from domain when calling friends
Actually SIPFROMDOMAIN was documented here:
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
, but SIPFROMUSER was not. They are now both there! :)
On Wed, Feb 19, 2014 at 9:08 AM, Rusty Newton rnew...@digium.com wrote:
On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
torbjorn.abrahams...@gmail.com wrote:
I have a problem where I would like to be able to send an arbitrary SIP
domain when sending a call to a registered friend. By default the from
domain is set to the IP of the Asterisk server, but I would like to set it
to something else.
The case is that when a call from a foreign domain comes in to the Asterisk,
it will connect it to the callee (but with the domain changed). When the
callee wants to make a redial from call history, the domain will not be
correct.
I could probably do something with the fromdomain setting of the friend, but
I would like it to be dynamic, ie not having to update the friend definition
every time a different domain is used.
I understand that I would need to use outbound proxy in the client to
prevent it from dialing the domain directly.
Is this possible? Any alternatives?
I'm a little confused about what you want to do, however I'll throw
some information at you in hopes that it will help out.
I did a little research and found that you can set the outbound From
header domain and From header user through two channel variables:
SIPFROMDOMAIN, SIPFROMUSER
They are sparsely documented, but there is an example in extensions.conf
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial)
; check if we set the FREENUMDOMAIN global variable in [global]
same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
;if we did set it, then we'll use it for our outbound dialing
domain
It looks like they were added in 1.6.2.X of Asterisk, so if you are
using 1.8.X or above, you should have them.
On your inbound call, you could use the function SIP_HEADER[1] to
gather the domain and store it for later use when you want to set it
on the outbound call. Though I'm not sure how you could tell that the
call was a redial.
[1]:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER
I'm assuming when your SIP client redials that it calls through
Asterisk and is not dialing the previously caller directly.
Hope any of that helps. *Goes off to document SIPFROMDOMAIN and
SIPFROMUSER on the wiki*
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Dynamically setting from domain when calling friends
On Wed, Feb 19, 2014 at 12:12 PM, Torbjörn Abrahamsson torbjorn.abrahams...@gmail.com wrote: Thank you very much. I will try this! It seems to be what I'm looking for. I'm in most cases working with 1.2 asterisks, so I'm not up to date on newer features. My current project however needed a newer version. I tried some googleing, but I did not find these variables. Glad to help! Wow.. 1.2 ! Most are using 1.8 or 11 these days, so it is good to be aware of that when seeking help and Googeling. The 1.8 branch is the oldest supported version at the moment. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] type=peer vs type=user (depricated?)
On Wed, Jan 22, 2014 at 5:56 PM, Michelle Dupuis mdup...@ocg.ca wrote: I'm looking at setting type=peer vs type=user (in both IAX and SIP conf entries), and I found a comment attributed to digium (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that type=user is depricated and that we should only use type=peer Is that still correct? Will type=user be phased out, and should even new installs of older asterisk versions (eg: 1.6) use type=peer only? Are people still using type=user for phone sets? (and type=peer for upstream/trunks only) Howdy! This is always a confusing part of the chan_sip SIP channel driver. Rather than try to dig into any history, here is the current documentation (from sip.conf.sample in the Asterisk source of 1.8,11,12) that you should base your decision to use a particular type on: ; SIP entities have a 'type' which determines their roles within Asterisk. ; * For entities with 'type=peer': ; Peers handle both inbound and outbound calls and are matched by ip/port, so for ; The case of incoming calls from the peer, the IP address must match in order for ; The invitation to work. This means calls made from either direction won't work if ; The peer is unregistered while host=dynamic or if the host is otherise not set to ; the correct IP of the sender. ; * For entities with 'type=user': ; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't ; call them) and are matched by their authorization information (authname and secret). ; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting ; as long as the incoming SIP invite authorizes successfully. ; * For entities with 'type=friend': ; Asterisk will create the entity as both a friend and a peer. Asterisk will accept ; calls from friends like it would for users, requiring only that the authorization ; matches rather than the IP address. Since it is also a peer, a friend entity can ; be called as long as its IP is known to Asterisk. In the case of host=dynamic, ; this means it is necessary for the entity to register before Asterisk can call it. Most new work for SIP support in Asterisk is happening around res_pjsip[1][2]. I don't know that there is any plans to deprecate type=user going forward in chan_sip. Is that still correct? Will type=user be phased out, and should even new installs of older asterisk versions (eg: 1.6) use type=peer only? New installs of older Asterisk versions? That doesn't sound wise, seeing as the 1.6 branch doesn't have any support, even for security issues... A new install of Asterisk should be on a version of Asterisk supported by the developers.[3] Right now, that would be the latest of the 1.8,11, or 12 branches. That being said, 12 is rather new and has many significant changes that should be considered.[3] [1]: https://wiki.asterisk.org/wiki/display/AST/New+in+12 [2]: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip [3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Hope that helps, thanks! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] type=peer vs type=user (depricated?)
On Thu, Jan 23, 2014 at 7:01 PM, Rusty Newton rnew...@digium.com wrote: snip the 1.8,11, or 12 branches. That being said, 12 is rather new and has many significant changes that should be considered.[3] I meant to reference link [1] of course. :) -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Mailinglist Digium IP-phones : provisioning Digium D70
On Wed, Jan 22, 2014 at 9:11 AM, Jonas Kellens jonas.kell...@telenet.be wrote: snip So how do I get the Digium IP-phone to use the md5 digest authentication ?? For the benefit of the archives and those reading the list, but not the forums - this was answered here http://forums.digium.com/viewtopic.php?p=195944 It won't use MD5. It only uses Basic. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] MeetMe conference splitting
On Thu, Jan 23, 2014 at 8:09 AM, Igor Dvorzhak idm...@gmail.com wrote: snip How to move 2 of 3 users in the MeetMe conference to the newly created MeetMe conference? Dialplan example is welcome. Maybe something like an AMI redirect? https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Redirect https://wiki.asterisk.org/wiki/display/AST/AMI+Examples -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] Queue with linear strategy does not work
On Tue, Dec 10, 2013 at 10:14 PM, Thorben Jensen i...@thorben.dk wrote: I have a queue with linear strategy. When I add dynamic members it does NOT ring the members in the order they are added. I use the command AddQueueMember to add members but it seems to be random how it rings the members. Does it ring them in the same order for every call(even if it is not the expected order)? Does the order change up for each call even when no new members have been added? Can you provide a pastebin of a verbose log (see logger.conf) demonstrating the problem? -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] what is the possible cause of maximum pbx stack exceeded
I'm not a developer, but from comments in the code, it looks like that warning is generated when Asterisk dialplan processing exceeds a certain depth of includes. Seeing as it is possibly a dialplan related issue, and FreePBX is writing your dialplan, you may have the best odds of getting a relevant answer by asking on the FreePBX forums (and giving them access to a copy of your logs to examine) That's all I got! :) On Wed, Dec 4, 2013 at 3:27 AM, cov...@ccs.covici.com wrote: Hi. I am using asterisk 11 svn r401076M and I am getting this warning at times. I can't find much doing a google search, so anyone with any ideas? I have looked at the logs, but can find no particular pattern to indicate where this is happening and the system appears to be otherwise working, but I am still wondering if something is wrong. I am also using freepbx in case there are known issues there -- because some of these occur during their dialout trunk code. Any suggestions would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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] No of parked calls limit
I doubt there is a hard-coded limit for parked calls specifically, but I don't know the answer. If you ask in #asterisk-dev when devs are around, a developer familiar with the parking code could probably tell you. As you probably already know, you can limit calls in a variety of ways, one of the simplest and probably most general is the maxcalls option in asterisk.conf. On Tue, Oct 29, 2013 at 6:17 PM, Matt Hamilton mistral9...@hotmail.com wrote: Is there a limit to the number of parked calls Asterisk can handle? Thanks, Matt -- _ -- 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 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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
