[asterisk-users] [SIP/H.264] Codec negotiation problem ?
Hi, I've a problem configuring my Asterisk. What I try to reach is to interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP) with 1 constraint I can't change : every RTP flow needs to pass THROUGH Asterisk, and are NOT nated What I observe : - a call made from a SIP Phone registred in Asterisk to Tandberg works (voice and video bidirectionnal) - a call made from to Tandberg to the SIP phone doesn't work (voice bidirectionnal, voice only received by the SIP phone, no incomming video for Tandberg) I think the problem may come from codec negotiotation : - when call is made from the SIP phone, it uses code 99 for H.264 codec, as Asterisk. Tdb reply SIP:Ok with the same number for H.264 - when call is made from Tbd, it uses code 98 for H.264 codec. Asterisk then send the Invite with 99 as codec number I use the version 1.6.2.6 of Asterisk Is this kind of configuration supposed to work ? I know passing video media through Asterisk may not be optimal, but I really need it, even if I have to patch Asterisk Thanks for your help SDP send by Tandberg : -- v=0 o=tandberg 1 5 IN IP4 192.168.50.10 s=- c=IN IP4 192.168.50.10 b=CT:1920 t=0 0 m=audio 48260 RTP/AVP 100 101 9 8 0 102 b=TIAS:64000 a=rtpmap:100 G7221/16000 a=fmtp:100 bitrate=32000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:102 telephone-event/8000 a=fmtp:102 0-15 a=sendrecv m=video 48262 RTP/AVP 97 98 99 34 31 c=IN IP4 192.168.50.10 b=TIAS:192 a=rtpmap:97 H264-RCDO/9 a=fmtp:97 profile-level-id=008016;max- mbps=42000;max-fs=3600;max-smbps=323500 a=rtpmap:98 H264/9 a=fmtp:98 profile-level-id=428016;max-mbps=35000;max-fs=3600;max-smbps=323500 a=rtpmap:99 H263-1998/9 a=fmtp:99 custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo a=rtpmap:34 H263/9 a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200 a=rtpmap:31 H261/9 a=fmtp:31 cif=1;qcif=1;maxbr=19200 a=rtcp-fb:* nack pli a=sendrecv a=content:main a=label:11 a=answer:full m=application 5078 UDP/BFCP * c=IN IP4 192.168.50.10 a=floorctrl:c-s a=confid:1 a=floorid:2 mstrm:12 a=userid:1 a=setup:passive a=connection:new m=video 48264 RTP/AVP 99 34 31 c=IN IP4 192.168.50.10 b=TIAS:192 a=rtpmap:99 H263-1998/9 a=fmtp:99 custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo a=rtpmap:34 H263/9 a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200 a=rtpmap:31 H261/9 a=fmtp:31 cif=1;qcif=1;maxbr=19200 a=rtcp-fb:* nack pli a=sendrecv a=content:slides a=label:12 SDP send by Asterisk v=0 o=root 1077353049 1077353049 IN IP4 192.168.13.100 s=Asterisk PBX 1.6.2.6 c=IN IP4 192.168.13.100 b=CT:384 t=0 0 m=audio 14604 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 17962 RTP/AVP 99 a=rtpmap:99 H264/9 a=sendrecv Here is my sip.conf [general] port = 5060 bindaddr = 0.0.0.0 disallow = all realm = testRealm allow = ulaw allow = h264 videosupport=yes canreinvite = no calleridupdate = info usercallerid = no context = default [toTandberg] host=192.168.50.53 type=friend qualify=yes qualifyreq=1 -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe VS. Conference
hi, group there are two module can used for meeting. MeetMe and Conference(which is a plugin) My question is : which is better for large conference(maybe above 100 people in a meeting)? -- Thanks Regards Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Prepay Limited Calls.
Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really want to install another software to make this or modify all my settup. I'm wonder if someone is using something simple to limmit calls. Anyway if someone is using some other programs/software/scripts and another settup/method please let me know how is yours. I want to check few methods to realize that limmit. Thank you for help guys. -- _ -- 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 track a call result originated from originate AMI command
BTW, using the most common Asterisk distros out there that happen to sport a very complex dialplan, we see a lot of lost events, so that tracking calls on the basis of AMI observation alone becomes practically impossible. :-( l. 2010/8/8 Nasir Iqbal na...@ictinnovations.com Hi, Confusing! you are not alone here. Actually there is no unified development approach exist in Asterisk, every module, application introduce a new way to handle same things!! And the monitoring is most difficult part! you have to write different parsing algos to get each bit of information, and unfortunately you have to rewrite most of your code for every new release! And regarding your question, I recommend you to use AGI for monitoring here is some tips for you - in originate command use extension as destination. - create failed extension in same context. - you can include some variables in originate command which can be used later in dialplan. - use AGI scripts in destination and failed extensions to get and save call status in database. Regards -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Prepay Limited Calls.
Hi, It is simple to use max_limit perameter in dial command. Regards, Faisal Hanif On 8/9/2010 2:01 PM, Catalin S. wrote: Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really want to install another software to make this or modify all my settup. I'm wonder if someone is using something simple to limmit calls. Anyway if someone is using some other programs/software/scripts and another settup/method please let me know how is yours. I want to check few methods to realize that limmit. Thank you for help guys. -- _ -- 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] Prepay Limited Calls.
Hi Use Set(TIMEOUT(absolute)=XYZ) in your dialplan or timeout parameter in Dial and Originate commands. Get maximum available seconds from your db for calling peer and use it as timeout. But after every call you have to deduct used time from you db for calling peer. Regards On Mon, Aug 9, 2010 at 2:01 PM, Catalin S. jonsonpla...@gmail.com wrote: Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really want to install another software to make this or modify all my settup. I'm wonder if someone is using something simple to limmit calls. Anyway if someone is using some other programs/software/scripts and another settup/method please let me know how is yours. I want to check few methods to realize that limmit. Thank you for help guys. -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.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
Re: [asterisk-users] Codec Conversion
Hello Miguel molina, I did what you asked, but still the voice is too bad Regards On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina mmol...@millenium.com.cowrote: El 05/08/10 14:50, Tim Nelson escribió: - michel freiha mich...@gmail.com mich...@gmail.com wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi issue on sangoma A200
Hi max, Have look on my blog regarding this. http://ashikalim.blogspot.com/2010/08/config-asterisk-india-pstn-lines_09.html Thanks, Ashik On Sat, Aug 7, 2010 at 11:15 AM, Max Alex max.aster...@gmail.com wrote: Hi All, I have Sangoma A200 Card installed on my system, I have centos 5.5 with 64 bit, Here are the description for asterisk and dahdi. Asterisk 1.6..2.9 Dahdi: 2.3.0.1 I have two issues with dahdi 1) I am not getting full callerid on my phones from sangoma card to asterisk users. if i am connecting analog phone directly then i am getting callerid properly. I am in india and using Airtel Connection, I have set variables in chan_dahdi.conf as well for callerid but the not getting full digits in callerid, it is coming with 8 digits only. 2) Another issue is when I am hanging up the phone from inbound or outbound from the dahdi channel, it takes 5-6 seconds to dropping the call. Here are the confguration file for chan_dahdi.conf - ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2010-07-30 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes callerid=asreceived hanguponpolarityswitch=yes answeronpolarityswitch=yes ;cidstart=ring cidstart=polarity_IN ;cidsignalling=dtmf cidsignalling=dtmf hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no useincomingcalleridondahditransfer=yes ;callerid=asreceived ;Sangoma AFT-A200 [slot:4 bus:2 span:1] wanpipe1 context=from-internal group=1 echocancel=yes callerid=asreceived signalling = fxo_ks channel = 1 context=from-internal group=1 echocancel=yes callerid=asreceived signalling = fxo_ks channel = 2 context=from-zaptel group=0 echocancel=yes callerid=asreceived signalling = fxs_ks channel = 3 context=from-zaptel group=0 echocancel=yes callerid=asreceived signalling = fxs_ks channel = 4 --- Please hemp me for this issues. Thanks, Max Alex Voip Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] op_div: non-numeric argument
Ladies, Gentlemen
We are experiencing an unusual problem in our asterisk 1.4.34.. We are
attempting to determine if channels are in use before paging to them.
This works correctly, as in it pages the phone.. however, we see the error
message below on the console... after googling, we discovered limited
information regarding the issue...
-- Executing [npanxx7...@from-pstn:1] Set(SIP/L2Net-SS-00db,
TIMEOUT(absolute)=60) in new stack
-- Channel will hangup at 2010-08-09 12:11:27 UTC.
-- Executing [npanxx7...@from-pstn:2] Page(SIP/L2Net-SS-00db,
Local/7...@page) in new stack
-- Executing [7...@page:1] Macro(Local/7...@page-9da0,2,
page|SIP/7299) in new stack
-- Executing [...@macro-page:1] ChanIsAvail(Local/7...@page-9da0,2,
SIP/7299|js) in new stack
[Aug 9 08:10:27] WARNING[29209]: ast_expr2.y:901 op_div: non-numeric
argument
-- Executing [...@macro-page:2] GotoIf(Local/7...@page-9da0,2,
0?fail:autoanswer) in new stack
-- Goto (macro-page,s,3)
-- Executing [...@macro-page:3] Set(Local/7...@page-9da0,2,
_ALERT_INFO=RA) in new stack
-- Executing [...@macro-page:4] SIPAddHeader(Local/7...@page-9da0,2,
Call-Info: sip:XXX.XXX.XXX.XXX;answer-after=0) in new stack
-- Executing [...@macro-page:5] NoOp(Local/7...@page-9da0,2, ) in new
stack
-- Executing [...@macro-page:6] Dial(Local/7...@page-9da0,2,
SIP/7299||) in new stack
-- Called 7299
-- Called 7...@page
-- SIP/ITSP-SS-00db Playing 'beep' (language 'en')
-- Got SIP response 486 Busy Here back from XXX.XXX.XXX.XXX
-- SIP/7299-00dd is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [...@macro-page:7] Hangup(Local/7...@page-9da0,2, ) in
new stack
== Spawn extension (macro-page, s, 7) exited non-zero on
'Local/7...@page-9da0,2' in macro 'page'
== Spawn extension (page, 7299, 1) exited non-zero on
'Local/7...@page-9da0,2'
-- Created MeetMe conference 1023 for conference '2051129475d'
-- Hungup 'DAHDI/pseudo-1451849221'
== Spawn extension (from-pstn, NPANXX7298, 2) exited non-zero on
'SIP/L2Net-SS-00db'
sipy*CLI core show version
*Pertinent DialPlan Logic...*
[from-pstn]
exten = NPANXX7299,1,Set(TIMEOUT(absolute)=60)
exten = NPANXX7299,2,Page(Local/7...@page)
[macro-page];
exten = s,1,ChanIsAvail(${ARG1}|js)
exten = s,n,GoToIf($[${AVAILORIGCHAN} = ]?fail:autoanswer) ; SUSPECTED
ISSUE
exten = s,n(autoanswer),Set(_ALERT_INFO=RA)
exten = s,n,SIPAddHeader(Call-Info: sip:XXX.XXX.XXX.XXX\;answer-after=0)
exten = s,n,Dial(${ARG1}||)
exten = s,n(fail),Hangup
Thanks in advance for any insight that you may be able to provide...
(sensitive information masked for obvious reasons)
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Re: [asterisk-users] Monitor asterisk
maybe, can use the sneplivre for this... www.sneplivre.com.br detail is that in portuguese, this can be translated easily (i think) Renato dos Santos ren...@opens.com.br OpenS Tecnologia Ltda Rua Padre Marcelino Champagnat, 236 Jardim Atlântico - Florianópolis - SC - Brasil +55 (48) 3954-8000 http://www.opens.com.br shazaum.wordpress.com On 9 August 2010 00:30, Nasir Iqbal na...@ictinnovations.com wrote: I agree with you and suggest you to use CLI command via AMI, for example Command core show channels I prefer CLI commands when they are available, as they return an aggregate response as compared to AMI you do not need to filter, identity, and group multiple responses / events to get result of a single command! Regards On Sun, Aug 8, 2010 at 11:28 PM, Richard Zulu richard.z...@time.co.ugwrote: Thanks Nasri, I don't want to only be able to use the CLI because I need the Helpdesk and application support Unit to be able to monitor, and they are not all the techy with CLI and stuff.. On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal na...@ictinnovations.comwrote: Hi following asterisk cli commands can help show channels, show uptime and show sysinfo here is an example asterisk -x core show sysinfo On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu richard.z...@time.co.ugwrote: Hey guys, I have my asterisk box running without a gui. I now need to monitor usage, calls, traffic of voice calls on this asterisk server. I cannot now install a gui because the configs will be wiped out, how can i go about monitoring all the above? -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.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 -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.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 -- _ -- 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] Monitor asterisk
Hallo Keane, I truly have a nagios server, up and running 24/7 -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] redirect based on incoming number
How does one redirect calls based on incoming number or caller ID or the lack thereof? current I have for number 123-4567 that it redirects all 800 , 877 and 866 numbers to Voicemail directly. If the primary area code is 352 then accept this and pass it to extension exten = 1234567/_352XXX,4,Dial(SIP/,240) exten = 1234567/_800XXX,4,Voicemail(5...@default,b) exten = 1234567/_866XXX,4,Voicemail(5...@default,b) exten = 1234567/_877XXX,4,Voicemail(5...@default,b) exten = 1234567/1800XXX,4,Voicemail(5...@default,b) exten = 1234567/1866XXX,4,Voicemail(5...@default,b) exten = 1234567/1877XXX,4,Voicemail(5...@default,b) exten = 1234567/+1800XXX,4,Voicemail(5...@default,b) exten = 1234567/+1866XXX,4,Voicemail(5...@default,b) exten = 1234567/+1877XXX,4,Voicemail(5...@default,b) exten = 1234567/_*1866876.,4,Voicemail(5...@default,b) exten = 1234567/_+18668762996,4,Voicemail(5...@default,b) Any help will be greatly appriecated Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scilence problem on running call
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kisho...@techroutes.com Subject: [asterisk-users] Scilence problem on running call Importance: High Dear All, I am getting scilence for 2-3 second in running calls on E1 CAS in Asterisk .. anybody help me ...what is the problem.. Regards, Kishor Kumar My vote is going to be for either DAHDI negotiation or a general network delay. Is the silence at the start of the call? It is also possible that you need to adjust your RXgain. -- _ -- 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] Codec Conversion
El 09/08/10 05:30, michel freiha escribió: Hello Miguel molina, I did what you asked, but still the voice is too bad Regards On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina mmol...@millenium.com.co mailto:mmol...@millenium.com.co wrote: El 05/08/10 14:50, Tim Nelson escribió: - michel freiha mich...@gmail.com mailto:mich...@gmail.com wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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 Hi, I didn't ask nothing... but as Tim said you are encouraged to change the iLBC codec to other (could be GSM) and do some tests. Play with several codecs and see which one fits your needs or whether this is not a codec or transcoding issue. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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 track a call result originated from originate AMI command
On Mon, Aug 9, 2010 at 12:08 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:
BTW, using the most common Asterisk distros out there that happen to sport a
very complex dialplan, we see a lot of lost events, so that tracking calls
on the basis of AMI observation alone becomes practically impossible.
:-(
l.
You can filter AMI. If you know PERL, you can start with my script
that works with callbacks:
$callbacks{'Newstate'} = \newstate_callback;
$callbacks{'Dial'} = \dial_callback;
And create appropriate functions for storing desired values to the
database. We catch Dial, Answer, Ringing, Hangup events and store that
info to database with very accurate timestamps :-)
http://github.com/Motiejus/Asterisk-perl-AMI/blob/master/asterisk_ami.pl
Regards,
Motiejus Jakštys
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Re: [asterisk-users] MeetMe VS. Conference
On Mon, Aug 9, 2010 at 4:36 AM, Zhang Shukun bit...@gmail.com wrote: hi, group there are two module can used for meeting. MeetMe and Conference(which is a plugin) My question is : which is better for large conference(maybe above 100 people in a meeting)? There's at least one more choice, which is ConfBridge(), assuming you're running 1.6.2.* I personally haven't used anything except MeetMe, and I have no experience with that large of a conference. -- _ -- 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 response 500 Server Internal Error
Hi, I have problem in initiating an dial out call with SIP response 500 Server Internal Error The sip debug as == Using SIP RTP CoS mark 5 Audio is at 113.253.226.92 port 18284 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 203.80.89.139:5060: INVITE sip:27101...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.226.92:5060;branch=z9hG4bK5b563aea;rport Max-Forwards: 70 From: IAX-cklee sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as4cffc48a To: sip:27101...@s2hkbntel.net:5060 Contact: sip:3594410...@113.253.226.92 sip%3a3594410...@113.253.226.92 Call-ID: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net CSeq: 102 INVITE User-Agent: Asterisk Remote-Party-ID: IAX-cklee sip:6...@s2hkbntel.netsip%3a6...@s2hkbntel.net ;privacy=off;screen=yes Date: Mon, 09 Aug 2010 15:03:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 241 v=0 o=root 1216883305 1216883305 IN IP4 113.253.226.92 s=Asterisk PBX 1.6.2.10 c=IN IP4 113.253.226.92 t=0 0 m=audio 18284 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 27101...@hkbn2b --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 100 Trying t: sip:27101...@s2hkbntel.net:5060 f: IAX-cklee sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as4cffc48a i: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea Server: MCS5x00_3.0 k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 - --- (9 headers 0 lines) --- --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 500 Server Internal Error t: sip:27101...@s2hkbntel.net:5060;tag=301677433 f: IAX-cklee sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as4cffc48a i: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 - --- (8 headers 0 lines) --- -- Got SIP response 500 Server Internal Error back from 203.80.89.139 Transmitting (NAT) to 203.80.89.139:5060: ACK sip:27101...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.226.92:5060;branch=z9hG4bK5b563aea;rport Max-Forwards: 70 From: IAX-cklee sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as4cffc48a To: sip:27101...@s2hkbntel.net:5060;tag=301677433 Contact: sip:3594410...@113.253.226.92 sip%3a3594410...@113.253.226.92 Call-ID: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net CSeq: 102 ACK User-Agent: Asterisk Remote-Party-ID: IAX-cklee sip:6...@s2hkbntel.netsip%3a6...@s2hkbntel.net ;privacy=off;screen=yes Content-Length: 0 I have no idea how to make it work. CK -- _ -- 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 response 500 Server Internal Error
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Subject: Re: [asterisk-users] SIP response 500 Server Internal Error Hi, I have problem in initiating an dial out call with SIP response 500 Server Internal Error The sip debug as snip --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 500 Server Internal Error t: sip:27101...@s2hkbntel.net:5060;tag=301677433 f: IAX-cklee sip:3594410...@s2hkbntel.net mailto:sip%3a3594410...@s2hkbntel.net ;tag=as4cffc48a i: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 I have no idea how to make it work. CK It looks like your firewall is blocking 203.80.89.139? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting two calls with Originate
Hello list!! I want to connect an open call with an extension. I call in with a DID, them redirect to the extension using AGI. Can I use agi's originate to make the second call without dropping the first DID call? How would I go about this? I had something like this in mind: first answer the DID call, then with AGI: Action: login Username: Secret: Events: off Action: Originate Channel: OpenDIDCall Context: context-for-second-call Exten: number to be called Priority: 1 Callerid: CallerID Timeout: 30 and connect the 2 calls. I am not having much luck, am I going about this the wrong way? Thanks in advance for your replies. -- _ -- 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] Connecting two calls with Originate
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: [asterisk-users] Connecting two calls with Originate Hello list!! I want to connect an open call with an extension. I call in with a DID, them redirect to the extension using AGI. Can I use agi's originate to make the second call without dropping the first DID call? How would I go about this? snip I am not having much luck, am I going about this the wrong way? Thanks in advance for your replies. Assuming that you're not trying to dial back out on the same line, this should not be problematic. The AGI originate is not necessarily aware that it is working in tandem with an existing call. The Channelopendidcall is the wrong way part of this equation. For example, if the call comes in on DAHDI/1-1, you can't use DAHDI/1-1 to open a second call whilst it is active; you can make a call on DAHDI/1-2 and join the 2 together. -- _ -- 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] Connecting two calls with Originate
Wow, that was fast. Thanks for your reply!!! So if I were to do: Action: login Username: Secret: Events: off Action: Originate Channel: SIP/trunk Context: context-for-second-call Exten: secondCall Priority: 1 Callerid: CallerID Timeout: 30 I could connect the 2 calls? It's my first time using Originate, so be patient with me :) On Mon, Aug 9, 2010 at 10:00 AM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones *Subject:* [asterisk-users] Connecting two calls with Originate Hello list!! I want to connect an open call with an extension. I call in with a DID, them redirect to the extension using AGI. Can I use agi's originate to make the second call without dropping the first DID call? How would I go about this? snip I am not having much luck, am I going about this the wrong way? Thanks in advance for your replies. Assuming that you’re not trying to dial back out on the same line, this should not be problematic. The AGI originate is not necessarily aware that it is working in tandem with an existing call. The “Channelopendidcall” is the “wrong way” part of this equation. For example, if the call comes in on DAHDI/1-1, you can’t use DAHDI/1-1 to open a second call whilst it is active; you can make a call on DAHDI/1-2 and join the 2 together. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Allison Smith Hilarity
Greetings and salutations Asterisk community, I've been contacted by a man who has generously posted some prompts he commissioned from Allison Smith. If you haven't heard Allison in humor mode, you owe it to yourself to hear this. Joey Lindstrom has decided to place these in the public domain and he's asking Digium to include them in the Asterisk prompts collection. Because he's encouraging anyone who want to use them to do so, feel free to download the collection: http://vuc.li/FunnyAllison Kudos to Joey for this, his way, he says, to give something back to the people who make Asterisk so special. That's YOU! /r ps: Allison is at her top form too. Make sure you follow Allison on Twitter, she's @voicegal -- _ -- 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] op_div: non-numeric argument
s,n,GoToIf($[${AVAILORIGCHAN} = ]?fail:autoanswer) ; SUSPECTED
ISSUE
You need quotes around your variable as well as your evaluation ($
{AVILORIGCHAN} = ).
Thanks,
--Warren Selby
On Aug 9, 2010, at 7:27 AM, Positively Optimistic
positivelyoptimis...@gmail.com
wrote:
Ladies, Gentlemen
We are experiencing an unusual problem in our asterisk 1.4.34.. We
are attempting to determine if channels are in use before paging to
them.
This works correctly, as in it pages the phone.. however, we see
the error message below on the console... after googling, we
discovered limited information regarding the issue...
-- Executing [npanxx7...@from-pstn:1] Set(SIP/L2Net-
SS-00db, TIMEOUT(absolute)=60) in new stack
-- Channel will hangup at 2010-08-09 12:11:27 UTC.
-- Executing [npanxx7...@from-pstn:2] Page(SIP/L2Net-
SS-00db, Local/7...@page) in new stack
-- Executing [7...@page:1] Macro(Local/7...@page-9da0,2, page|
SIP/7299) in new stack
-- Executing [...@macro-page:1] ChanIsAvail(Local/
7...@page-9da0,2, SIP/7299|js) in new stack
[Aug 9 08:10:27] WARNING[29209]: ast_expr2.y:901 op_div: non-
numeric argument
-- Executing [...@macro-page:2] GotoIf(Local/7...@page-9da0,2,
0?fail:autoanswer) in new stack
-- Goto (macro-page,s,3)
-- Executing [...@macro-page:3] Set(Local/7...@page-9da0,2,
_ALERT_INFO=RA) in new stack
-- Executing [...@macro-page:4] SIPAddHeader(Local/
7...@page-9da0,2, Call-Info: sip:XXX.XXX.XXX.XXX;answer-
after=0) in new stack
-- Executing [...@macro-page:5] NoOp(Local/7...@page-9da0,2, )
in new stack
-- Executing [...@macro-page:6] Dial(Local/7...@page-9da0,2,
SIP/7299||) in new stack
-- Called 7299
-- Called 7...@page
-- SIP/ITSP-SS-00db Playing 'beep' (language 'en')
-- Got SIP response 486 Busy Here back from XXX.XXX.XXX.XXX
-- SIP/7299-00dd is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [...@macro-page:7] Hangup(Local/7...@page-9da0,2,
) in new stack
== Spawn extension (macro-page, s, 7) exited non-zero on 'Local/
7...@page-9da0,2' in macro 'page'
== Spawn extension (page, 7299, 1) exited non-zero on 'Local/
7...@page-9da0,2'
-- Created MeetMe conference 1023 for conference '2051129475d'
-- Hungup 'DAHDI/pseudo-1451849221'
== Spawn extension (from-pstn, NPANXX7298, 2) exited non-zero on
'SIP/L2Net-SS-00db'
sipy*CLI core show version
Pertinent DialPlan Logic...
[from-pstn]
exten = NPANXX7299,1,Set(TIMEOUT(absolute)=60)
exten = NPANXX7299,2,Page(Local/7...@page)
[macro-page];
exten = s,1,ChanIsAvail(${ARG1}|js)
exten = s,n,GoToIf($[${AVAILORIGCHAN} = ]?fail:autoanswer) ;
SUSPECTED ISSUE
exten = s,n(autoanswer),Set(_ALERT_INFO=RA)
exten = s,n,SIPAddHeader(Call-Info: sip:XXX.XXX.XXX.XXX\;answer-
after=0)
exten = s,n,Dial(${ARG1}||)
exten = s,n(fail),Hangup
Thanks in advance for any insight that you may be able to provide...
(sensitive information masked for obvious reasons)
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Re: [asterisk-users] Connecting two calls with Originate
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Sent: Monday, August 09, 2010 11:22 AM Subject: Re: [asterisk-users] Connecting two calls with Originate On Mon, Aug 9, 2010 at 10:00 AM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: [asterisk-users] Connecting two calls with Originate Hello list!! I want to connect an open call with an extension. I call in with a DID, them redirect to the extension using AGI. Can I use agi's originate to make the second call without dropping the first DID call? How would I go about this? snip I am not having much luck, am I going about this the wrong way? Thanks in advance for your replies. -Assuming that you're not trying to dial back out on the same line, this should not be problematic. The AGI originate is not necessarily aware that it is working in tandem with an existing call. The Channelopendidcall is the wrong way part of this equation. For example, if the call comes in on DAHDI/1-1, you can't use DAHDI/1-1 to open a second call whilst it is active; you can make a call on DAHDI/1-2 and join the 2 together. Wow, that was fast. Thanks for your reply!!! So if I were to do: Action: login Username: Secret: Events: off Action: Originate Channel: SIP/trunk Context: context-for-second-call Exten: secondCall Priority: 1 Callerid: CallerID Timeout: 30 I could connect the 2 calls? As best as I know, yes this should work. You are actually creating a new leg with the originate, but the net effect is a joined call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 'System' application in asterisk
Hello, Is there any way to capture the output of the 'System' application in asterisk dialplan and evaluate it. For example, i would like to get the output of following System application and use its value in next line for decision making exten = 5000,n,System(command) -- _ -- 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] redirect based on incoming number
Barry Fawthrop wrote:
How does one redirect calls based on incoming number or caller ID or the
lack thereof?
current I have for number 123-4567 that it redirects all 800 , 877 and
866 numbers to Voicemail directly.
If the primary area code is 352 then accept this and pass it to
extension
exten = 1234567/_352XXX,4,Dial(SIP/,240)
exten = 1234567/_800XXX,4,Voicemail(5...@default,b)
exten = 1234567/_866XXX,4,Voicemail(5...@default,b)
exten = 1234567/_877XXX,4,Voicemail(5...@default,b)
exten = 1234567/1800XXX,4,Voicemail(5...@default,b)
exten = 1234567/1866XXX,4,Voicemail(5...@default,b)
exten = 1234567/1877XXX,4,Voicemail(5...@default,b)
exten = 1234567/+1800XXX,4,Voicemail(5...@default,b)
exten = 1234567/+1866XXX,4,Voicemail(5...@default,b)
exten = 1234567/+1877XXX,4,Voicemail(5...@default,b)
exten = 1234567/_*1866876.,4,Voicemail(5...@default,b)
exten = 1234567/_+18668762996,4,Voicemail(5...@default,b)
Any help will be greatly appriecated
Thanks
[menu]
exten = s,n,Set(NPA=${CALLERID(num):0:3}); grab area code from caller id
exten = s,n,GotoIF($[ ${NPA} = 800 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 888 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 877 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 866 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 855 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 844 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 833 ]?marketeer)
exten = s,n,GotoIF($[ ${NPA} = 822 ]?marketeer)
exten = s,n(marketeer),Set(TIMEOUT(digit)=6); allow humans to bypass
drop into VM
exten = s,n,Set(TIMEOUT(response)=10);
exten = s,n,Set(CALLERID(num)=51${CALLERID(num)})
exten = s,n,Background(missingcallerid); Dial 111 to actually ring a phone
exten = s,n,Voicemail(u111); no digits dialed drop into VM
exten = s,n,Hangup
Lyle
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Re: [asterisk-users] 'System' application in asterisk
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tino
Subject: [asterisk-users] 'System' application in asterisk
Hello,
Is there any way to capture the output of the 'System' application in
asterisk dialplan and evaluate it.
For example, i would like to get the output of following System application
and use its value in next line
for decision making
exten = 5000,n,System(command)
I think this answer is no. system only returns ${SYSTEMSTATUS} as SUCCESS
or FAILURE to tell you that the command finished or died. You could however
do a bash AGI that would set a variable with the result of what you would
have sent to system
Replace
Exten = 5000,n,System('/bin/ls')
With
Exten = 5000,n,AGI(bashsys.sh,/bin/ls')
Exten = 5000,n.Gotoif(${RESULT}.
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Re: [asterisk-users] 'System' application in asterisk
On Monday 09 August 2010 13:08:19 Tino wrote:
Is there any way to capture the output of the 'System' application in
asterisk dialplan and evaluate it.
For example, i would like to get the output of following System application
and use its value in next line
for decision making
exten = 5000,n,System(command)
No, but you may use the SHELL dialplan function in versions which support it:
Set(output=${SHELL(command)})
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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[asterisk-users] check channels
Hi guys, is there a way to see how many channels of an specific tecnology are being used? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Correct Caller-ID
I've seen caller-id come through from carriers as: NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX- My question is: what is the correct way to send Caller-ID by set standards? -- _ -- 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] check channels
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe Figueiredo Subject: [asterisk-users] check channels Hi guys, is there a way to see how many channels of an specific tecnology are being used? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Thanks! /usr/sbin/asterisk -rx core show channels|grep Zap will show you how many Zap channels are open at any moment. You can grep for SIP, Zap or DAHDI (or IAX) depending on your release/technology. To do this as you specified, you would record a message (say busy15.gsm) and set up a call file to call yourself and play the busy15 message. I used to have my Asterisk call me whenever it restarted this way. -- _ -- 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] check channels
Thanks Danny, but the system won't know exactly how many channels are being used right? if I use the asterisk -rx cmd, this is the result: Zap/63-1 (None) Up Bridged Call(SIP/xxx) It won't show how many zap channels are busy . I need to count the busy channels, and with the count results, dial to an exten and tell how many channels are busy ; thanks On Mon, Aug 9, 2010 at 4:14 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felipe Figueiredo *Subject:* [asterisk-users] check channels Hi guys, is there a way to see how many channels of an specific tecnology are being used? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Thanks! /usr/sbin/asterisk –rx “core show channels”|grep Zap will show you how many Zap channels are open at any moment. You can grep for SIP, Zap or DAHDI (or IAX) depending on your release/technology. To do this as you specified, you would record a message (say busy15.gsm) and set up a call file to call yourself and play the busy15 message. I used to have my Asterisk call me whenever it restarted this way. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] check channels
On Mon, Aug 9, 2010 at 4:14 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe Figueiredo Subject: [asterisk-users] check channels Hi guys, is there a way to see how many channels of an specific tecnology are being used? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Thanks! /usr/sbin/asterisk -rx core show channels|grep Zap will show you how many Zap channels are open at any moment. You can grep for SIP, Zap or DAHDI (or IAX) depending on your release/technology. To do this as you specified, you would record a message (say busy15.gsm) and set up a call file to call yourself and play the busy15 message. I used to have my Asterisk call me whenever it restarted this way. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe Figueiredo Thanks Danny, but the system won't know exactly how many channels are being used right? if I use the asterisk -rx cmd, this is the result: Zap/63-1 (None) Up Bridged Call(SIP/xxx) It won't show how many zap channels are busy . I need to count the busy channels, and with the count results, dial to an exten and tell how many channels are busy ; /usr/sbin/asterisk -rx core show channels|grep Zap|wc Will give you just the number of channels in use of type Zap. It isn't too hard to evaluate and act from that. Just don't have my shell scripting hat on today. -- _ -- 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] check channels
On Mon, 9 Aug 2010, Felipe Figueiredo wrote: is there a way to see how many channels of an specific tecnology are being used? See? From where? Within the dialplan or from an external process? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Within the dial plan you can use the GROUP() and GROUP_COUNT() functions. You could set the group to the technology and check the count as each call enters your dial plan. (Asterisk will automagically decrement the count as the calls are terminated.) If the count exceeds your threshold, you could use system() to create a call file to call you and play an appropriate message. From an external process you can parse the output of asterisk -r -x 'show channels', asterisk -r -x 'sip show channels', or asterisk -r -x 'zap show channels'. Note that these are the 1.2 commands. You are probably using a more current version. If the count exceeds your threshold, create a call file to call you and play an appropriate message. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct Caller-ID
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Subject: [asterisk-users] Correct Caller-ID I've seen caller-id come through from carriers as: NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX- My question is: what is the correct way to send Caller-ID by set standards? The correct answer to this depends on where you are. -- _ -- 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] Correct Caller-ID
Continental US-48. On Mon, Aug 9, 2010 at 3:36 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Matt *Subject:* [asterisk-users] Correct Caller-ID I've seen caller-id come through from carriers as: NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX- My question is: what is the correct way to send Caller-ID by set standards? The correct answer to this depends on where you are. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct Caller-ID
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Subject: Re: [asterisk-users] Correct Caller-ID Continental US-48. On Mon, Aug 9, 2010 at 3:36 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Subject: [asterisk-users] Correct Caller-ID I've seen caller-id come through from carriers as: NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX- My question is: what is the correct way to send Caller-ID by set standards? The correct answer to this depends on where you are. IMO the answer would be #2, but #3 would probably be acceptable - Google wasn't helpful on this one. -- _ -- 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] Correct Caller-ID
I've seen caller-id come through from carriers as: NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX- My question is: what is the correct way to send Caller-ID by set standards? The correct answer to this depends on where you are. IMO the answer would be #2, but #3 would probably be acceptable – Google wasn’t helpful on this one. OK... the reason I asked is because I've seen Vonage send the 1, and a cellular carrier I have sends the 1, but it seems that most phones are designed to not know what to do with that (e.g. they only ask for the local area code NPA), thus when I call comes through with the 1, the phone has no clue what to do with it. -- _ -- 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] Correct Caller-ID
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Subject: Re: [asterisk-users] Correct Caller-ID I've seen caller-id come through from carriers as: NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX- My question is: what is the correct way to send Caller-ID by set standards? The correct answer to this depends on where you are. IMO the answer would be #2, but #3 would probably be acceptable - Google wasn't helpful on this one. OK... the reason I asked is because I've seen Vonage send the 1, and a cellular carrier I have sends the 1, but it seems that most phones are designed to not know what to do with that (e.g. they only ask for the local area code NPA), thus when I call comes through with the 1, the phone has no clue what to do with it. My best answer in this scenario would be to obtain/deduce the NPA-NXX combinations that are known cell providers (such as 205-616) and send those without the 1. you could build this information into an SQL table pretty easily. -- _ -- 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] Correct Caller-ID
On Monday 09 August 2010 14:03:36 Matt wrote: I've seen caller-id come through from carriers as: NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX- My question is: what is the correct way to send Caller-ID by set standards? Given that you can only send CallerID on a digital circuit, it's going to depend upon the signalling. If the prilocaldialplan is set to 'unknown' or 'dynamic', then you send the '1'. If the prilocaldialplan is set to 'domestic', then you leave the '1' off (because it's implied). In terms of SIP or another VoIP protocol, you use whatever your provider specifies. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two calls with Originate
I have been working on this for a while today, and still no luck. This is my
script:
#!/usr/bin/php
?php
$errno=0;
$errstr=0;
$fp = fsockopen (localhost,5038,$errno,$errstr,20);
if (!$fp) {
echo $errstr ($errno)br\n;
} else {
fputs($fp, Action: Login\r\n);
fputs($fp, Username: \r\n);
fputs($fp, Secret: \r\n);
fputs($fp, Events: off\r\n);
sleep(1);
fputs($fp, Action: Originate\r\n);
fputs($fp, Channel: SIP/trunk/1DIDNumber\r\n);
fputs($fp, Context: CallContext\r\n\r\n);
fputs($fp, Exten: NumberToCall\r\n);
fputs($fp, Priority: 1\r\n);
fputs($fp, Timeout: 3\r\n);
sleep(2);
fclose($fp);
}
?
It seems simple enough, And I have no compilation errors. This is my output:
-- Launched AGI Script /var/lib/asterisk/agi-bin/MyScript.php
SIP/xx.xx.xxx.xx-0111AGI Tx agi_request: MyScript.php
SIP/xx.xx.xxx.xx-0111AGI Tx agi_channel: SIP/xx.xx.xxx.xx-0111
SIP/xx.xx.xxx.xx-0111AGI Tx agi_language: en
SIP/xx.xx.xxx.xx-0111AGI Tx agi_type: SIP
SIP/xx.xx.xxx.xx-0111AGI Tx agi_uniqueid: 128139.000
SIP/xx.xx.xxx.xx-0111AGI Tx agi_version: 1.6.2.6
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callerid: 1PhoneThatCalled The DID
SIP/xx.xx.xxx.xx-0111AGI Tx agi_calleridname: unknown
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callingpres: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callingani2: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callington: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callingtns: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_dnid: IncomingExt
SIP/xx.xx.xxx.xx-0111AGI Tx agi_rdnis: unknown
SIP/xx.xx.xxx.xx-0111AGI Tx agi_context: default
SIP/xx.xx.xxx.xx-0111AGI Tx agi_extension: incomingExt
SIP/xx.xx.xxx.xx-0111AGI Tx agi_priority: 3
SIP/xx.xx.xxx.xx-0111AGI Tx agi_enhanced: 0.0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_accountcode:
SIP/xx.xx.xxx.xx-0111AGI Tx agi_threadid: -123700
SIP/xx.xx.xxx.xx-0111AGI Tx
== Manager 'Man' logged on from 127.0.0.1
== Manager 'Man' logged off from 127.0.0.1
SIP/xx.xx.xxx.xx-0111AGI Rx
SIP/xx.xx.xxx.xx-0111AGI Tx 510 Invalid or unknown command
[Aug 9 17:44:10] ERROR[25594]: utils.c:1128 ast_carefulwrite: write()
returned error: Broken pipe
[Aug 9 17:44:10] ERROR[25594]: utils.c:1128 ast_carefulwrite: write()
returned error: Broken pipe
-- SIP/xx.xx.xxx.xx-0111AGI Script MyScript.php completed,
returning 0
Could someone please point me in the right direction?
On Mon, Aug 9, 2010 at 11:15 AM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones
*Sent:* Monday, August 09, 2010 11:22 AM
*Subject:* Re: [asterisk-users] Connecting two calls with Originate
On Mon, Aug 9, 2010 at 10:00 AM, Danny Nicholas da...@debsinc.com wrote:
***From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones
*Subject:* [asterisk-users] Connecting two calls with Originate
Hello list!!
I want to connect an open call with an extension. I call in with a DID,
them redirect to the extension using AGI. Can I use agi's originate to make
the second call without dropping the first DID call? How would I go
about this?
snip
I am not having much luck, am I going about this the wrong way? Thanks
in advance for your replies.
-Assuming that you’re not trying to dial back out on the same line, this
should not be problematic. The AGI originate is not necessarily aware that
it is working in tandem with an existing call. The “Channelopendidcall” is
the “wrong way” part of this equation. For example, if the call comes in on
DAHDI/1-1, you can’t use DAHDI/1-1
to open a second call whilst it is active; you can make a call on
DAHDI/1-2 and join the 2 together.
Wow, that was fast. Thanks for your reply!!!
So if I were to do:
Action: login
Username:
Secret:
Events: off
Action: Originate
Channel: SIP/trunk
Context: context-for-second-call
Exten: secondCall
Priority: 1
Callerid: CallerID
Timeout: 30
I could connect the 2 calls?
As best as I know, yes this should work. You are actually creating a “new
leg” with the originate, but the net effect is a joined call.
--
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Re: [asterisk-users] SIP response 500 Server Internal Error
I try to disable firewall but no working. I use a softphone to connect on the same lan segment, it works. Dial in is no problem but dial out always have this error On Mon, Aug 9, 2010 at 11:21 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk *Subject:* Re: [asterisk-users] SIP response 500 Server Internal Error Hi, I have problem in initiating an dial out call with SIP response 500 Server Internal Error The sip debug as snip --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 500 Server Internal Error t: sip:27101...@s2hkbntel.net:5060;tag=301677433 f: IAX-cklee sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as4cffc48a i: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 I have no idea how to make it work. CK It looks like your firewall is blocking 203.80.89.139? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct Caller-ID
Matt wrote: I've seen caller-id come through from carriers as: NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX- My question is: what is the correct way to send Caller-ID by set standards? The correct answer to this depends on where you are. IMO the answer would be #2, but #3 would probably be acceptable – Google wasn’t helpful on this one. OK... the reason I asked is because I've seen Vonage send the 1, and a cellular carrier I have sends the 1, but it seems that most phones are designed to not know what to do with that (e.g. they only ask for the local area code NPA), thus when I call comes through with the 1, the phone has no clue what to do with it. Also in the US. My Vonage account does NOT send the 1 My t-mob...@home account DOES send the 1 Neither REQUIRE the 1 to be dialed!! IMO the + is not correct when sending CLID. It is correctly used in print when referencing international numbers. Why would you expect anyone to follow a standard? John Novack -- Dog is my Co-pilot -- _ -- 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] Connecting two calls with Originate
On Mon, 9 Aug 2010, Kathryn Jones wrote:
I have been working on this for a while today, and still no luck. This is my
script:
#!/usr/bin/php
?php
$errno=0;
$errstr=0;
$fp = fsockopen (localhost,5038,$errno,$errstr,20);
if (!$fp) {
echo $errstr ($errno)br\n;
} else {
fputs($fp, Action: Login\r\n);
fputs($fp, Username: \r\n);
fputs($fp, Secret: \r\n);
fputs($fp, Events: off\r\n);
sleep(1);
fputs($fp, Action: Originate\r\n);
fputs($fp, Channel: SIP/trunk/1DIDNumber\r\n);
fputs($fp, Context: CallContext\r\n\r\n);
fputs($fp, Exten: NumberToCall\r\n);
fputs($fp, Priority: 1\r\n);
fputs($fp, Timeout: 3\r\n);
sleep(2);
fclose($fp);
}
?
It seems simple enough, And I have no compilation errors. This is my output:
-- Launched AGI Script /var/lib/asterisk/agi-bin/MyScript.php
SIP/xx.xx.xxx.xx-0111AGI Tx agi_request: MyScript.php
SIP/xx.xx.xxx.xx-0111AGI Tx agi_channel: SIP/xx.xx.xxx.xx-0111
SIP/xx.xx.xxx.xx-0111AGI Tx agi_language: en
SIP/xx.xx.xxx.xx-0111AGI Tx agi_type: SIP
SIP/xx.xx.xxx.xx-0111AGI Tx agi_uniqueid: 128139.000
SIP/xx.xx.xxx.xx-0111AGI Tx agi_version: 1.6.2.6
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callerid: 1PhoneThatCalled The DID
SIP/xx.xx.xxx.xx-0111AGI Tx agi_calleridname: unknown
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callingpres: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callingani2: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callington: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callingtns: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_dnid: IncomingExt
SIP/xx.xx.xxx.xx-0111AGI Tx agi_rdnis: unknown
SIP/xx.xx.xxx.xx-0111AGI Tx agi_context: default
SIP/xx.xx.xxx.xx-0111AGI Tx agi_extension: incomingExt
SIP/xx.xx.xxx.xx-0111AGI Tx agi_priority: 3
SIP/xx.xx.xxx.xx-0111AGI Tx agi_enhanced: 0.0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_accountcode:
SIP/xx.xx.xxx.xx-0111AGI Tx agi_threadid: -123700
SIP/xx.xx.xxx.xx-0111AGI Tx
== Manager 'Man' logged on from 127.0.0.1
== Manager 'Man' logged off from 127.0.0.1
SIP/xx.xx.xxx.xx-0111AGI Rx
SIP/xx.xx.xxx.xx-0111AGI Tx 510 Invalid or unknown command
[Aug 9 17:44:10] ERROR[25594]: utils.c:1128 ast_carefulwrite: write() returned
error: Broken pipe
[Aug 9 17:44:10] ERROR[25594]: utils.c:1128 ast_carefulwrite: write() returned
error: Broken pipe
-- SIP/xx.xx.xxx.xx-0111AGI Script MyScript.php completed, returning 0
Could someone please point me in the right direction?
This is not an AGI, this is an AMI :)
AGI is a protocol where Asterisk creates a process and sends it the AGI
environment (all the AGI Tx agi_xxx cruft above) and then waits for
your process to issue requests and read responses. This request,
response is repeated as your process completes it's tasks and exits.
Are you expecting your script to execute within the context of a channel
within Asterisk or as a process external to Asterisk?
I read your original request:
I want to connect an open call with an extension. I call in with a DID,
them redirect to the extension using AGI. Can I use agi's originate to
make the second call without dropping the first DID call? How would I go
about this?
as I call in, I execute an AGI that looks up an extension based on some
criteria, I want to dial that extension.
If this is close, the AGI should set a channel variable with the value of
the extension and exit. Your dialplan would look something like:
exten = my-did,1, verbose(${ext...@${context})
exten = my-did,n, agi(lookup-extension)
exten = my-did,n, dial{${LOOKED-UP-EXTENSION})
exten = my-did,n, hangup()
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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Re: [asterisk-users] Connecting two calls with Originate
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, August 09, 2010 7:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Connecting two calls with Originate
On Mon, 9 Aug 2010, Kathryn Jones wrote:
I have been working on this for a while today, and still no luck. This is my
script:
#!/usr/bin/php
?php
$errno=0;
$errstr=0;
$fp = fsockopen (localhost,5038,$errno,$errstr,20);
if (!$fp) {
echo $errstr ($errno)br\n; } else {
fputs($fp, Action: Login\r\n);
fputs($fp, Username: \r\n);
fputs($fp, Secret: \r\n);
fputs($fp, Events: off\r\n);
sleep(1);
fputs($fp, Action: Originate\r\n);
fputs($fp, Channel: SIP/trunk/1DIDNumber\r\n);
fputs($fp, Context: CallContext\r\n\r\n);
fputs($fp, Exten: NumberToCall\r\n);
fputs($fp, Priority: 1\r\n);
fputs($fp, Timeout: 3\r\n);
sleep(2);
fclose($fp);
}
?
It seems simple enough, And I have no compilation errors. This is my output:
-- Launched AGI Script /var/lib/asterisk/agi-bin/MyScript.php
SIP/xx.xx.xxx.xx-0111AGI Tx agi_request: MyScript.php
SIP/xx.xx.xxx.xx-0111AGI Tx agi_channel:
SIP/xx.xx.xxx.xx-0111 SIP/xx.xx.xxx.xx-0111AGI Tx
agi_language: en SIP/xx.xx.xxx.xx-0111AGI Tx agi_type: SIP
SIP/xx.xx.xxx.xx-0111AGI Tx agi_uniqueid: 128139.000
SIP/xx.xx.xxx.xx-0111AGI Tx agi_version: 1.6.2.6
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callerid: 1PhoneThatCalled
The DID SIP/xx.xx.xxx.xx-0111AGI Tx agi_calleridname: unknown
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callingpres: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callingani2: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callington: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_callingtns: 0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_dnid: IncomingExt
SIP/xx.xx.xxx.xx-0111AGI Tx agi_rdnis: unknown
SIP/xx.xx.xxx.xx-0111AGI Tx agi_context: default
SIP/xx.xx.xxx.xx-0111AGI Tx agi_extension: incomingExt
SIP/xx.xx.xxx.xx-0111AGI Tx agi_priority: 3
SIP/xx.xx.xxx.xx-0111AGI Tx agi_enhanced: 0.0
SIP/xx.xx.xxx.xx-0111AGI Tx agi_accountcode:
SIP/xx.xx.xxx.xx-0111AGI Tx agi_threadid: -123700
SIP/xx.xx.xxx.xx-0111AGI Tx
== Manager 'Man' logged on from 127.0.0.1
== Manager 'Man' logged off from 127.0.0.1
SIP/xx.xx.xxx.xx-0111AGI Rx SIP/xx.xx.xxx.xx-0111AGI Tx
510 Invalid or unknown command [Aug 9 17:44:10] ERROR[25594]:
utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe
[Aug 9 17:44:10] ERROR[25594]: utils.c:1128 ast_carefulwrite: write()
returned error: Broken pipe
-- SIP/xx.xx.xxx.xx-0111AGI Script MyScript.php completed,
returning 0
Could someone please point me in the right direction?
This is not an AGI, this is an AMI :)
AGI is a protocol where Asterisk creates a process and sends it the AGI
environment (all the AGI Tx agi_xxx cruft above) and then waits for your
process to issue requests and read responses. This request, response is
repeated as your process completes it's tasks and exits.
Are you expecting your script to execute within the context of a channel
within Asterisk or as a process external to Asterisk?
I read your original request:
I want to connect an open call with an extension. I call in with a
DID, them redirect to the extension using AGI. Can I use agi's
originate to make the second call without dropping the first DID call?
How would I go about this?
as I call in, I execute an AGI that looks up an extension based on some
criteria, I want to dial that extension.
If this is close, the AGI should set a channel variable with the value of the
extension and exit. Your dialplan would look something like:
exten = my-did,1, verbose(${ext...@${context})
exten = my-did,n, agi(lookup-extension)
exten = my-did,n, dial{${LOOKED-UP-EXTENSION})
exten = my-did,n, hangup()
/snip
If you are doing what Steve has described, and you require php, you should
really check out PHPAGI (http://phpagi.sourceforge.net/). It's a great
framework for using both AGI and the AMI in PHP. If you need just the AGI
component and are going to be doing this on a large scale, google cagi
(http://sourceforge.net/projects/cagi/). It is structured similarly to PHPAGI.
--
Elliot Otchet
Calling Circles LLC
This message is intended only for the use of the individual (s) or entity to
which it is addressed and may contain information that is privileged,
confidential, and/or proprietary to Calling Circles LLC and its affiliates. If
the reader of this message is not the intended recipient, you are hereby
notified that any dissemination, distribution, forwarding or copying of this
communication is
Re: [asterisk-users] check channels
Steve, you are right, i'm gonna use the group function, I tested here and it works pretty fine. Thanks. Danny, thanks for the help once again! On Mon, Aug 9, 2010 at 4:36 PM, Steve Edwards asterisk@sedwards.comwrote: On Mon, 9 Aug 2010, Felipe Figueiredo wrote: is there a way to see how many channels of an specific tecnology are being used? See? From where? Within the dialplan or from an external process? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Within the dial plan you can use the GROUP() and GROUP_COUNT() functions. You could set the group to the technology and check the count as each call enters your dial plan. (Asterisk will automagically decrement the count as the calls are terminated.) If the count exceeds your threshold, you could use system() to create a call file to call you and play an appropriate message. From an external process you can parse the output of asterisk -r -x 'show channels', asterisk -r -x 'sip show channels', or asterisk -r -x 'zap show channels'. Note that these are the 1.2 commands. You are probably using a more current version. If the count exceeds your threshold, create a call file to call you and play an appropriate message. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playback during call
Hi all,
How can I playback a file within an active call?
I've tried with ChanSpy whisper mode like this (using AMI):
Action: Originate
Channel: Local/9...@default
Priority: 0
Variable: MSG=test
Application: ChanSpy
Data: SIP/1234-123
Async: 1
and in the dialplan:
[default]
exten = ,1,Answer()
exten = ,n,Wait(2)
exten = ,n,Playback(${MSG})
Where SIP/1234-123 is the up bridged channel.
But this is not working (it seams that will work on the rolling CLI, but no
sound at all)
Is there a better way to do it?
--
_
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[asterisk-users] DEBUG: Cannot find variable 'XXX' ??
On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup, such as: == Registered custom function 'SIP_HEADER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPPEER' in tree 'description' == Registered custom function 'SIPPEER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPCHANINFO' in tree 'description' Maybe they've always been there, and I've been asleep. But what do they mean? Should I do something about them? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DEBUG: Cannot find variable 'XXX' ??
On Monday 09 August 2010 21:13:49 sean darcy wrote: On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup, such as: == Registered custom function 'SIP_HEADER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPPEER' in tree 'description' == Registered custom function 'SIPPEER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPCHANINFO' in tree 'description' Maybe they've always been there, and I've been asleep. But what do they mean? Should I do something about them? It means there is no XML documentation for those functions, which means there will be no inline help. You're welcome to write the documentation and submit a patch on issues.asterisk.org. Or not. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback during call
Your ami packet is not setting the w option for chanspy, nor I am sure you can
do this.
You might want to create an additional exten that takes a variable from your
ami packet and does the chanspy that way.
I use an ami packet like this with extension that do the work.
Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280_18-0002
ActionID: PlayBack
Async: true
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 9, 2010, at 5:19 PM, Gabriel Ortiz Lour wrote:
Hi all,
How can I playback a file within an active call?
I've tried with ChanSpy whisper mode like this (using AMI):
Action: Originate
Channel: Local/9...@default
Priority: 0
Variable: MSG=test
Application: ChanSpy
Data: SIP/1234-123
Async: 1
and in the dialplan:
[default]
exten = ,1,Answer()
exten = ,n,Wait(2)
exten = ,n,Playback(${MSG})
Where SIP/1234-123 is the up bridged channel.
But this is not working (it seams that will work on the rolling CLI, but no
sound at all)
Is there a better way to do it?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] check channels
You need to write an external application either on AMI to keep track of channels or an external application can get channel list by using shell command "/usr/sbin/asterisk -rx 'show channels'|grep zap" and then can count the output and generate a callback file to sent alert call. Signatures fai...@vopium.com Regards, Faisal Hanif VoIP Manager n 8/9/2010 11:56 PM, Felipe Figueiredo wrote: Hi guys, is there a way to see how many channels of an specific tecnology are being used? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] speciality of SIPp and SER(Sip Express Router)
Dear all, What is the difference between SIPp and SER(Sip Express Router)? Which one is better load performance testing? Is there any one who knows about this? Could you please give me details informtaion? Thans in advance Nahar -- _ -- 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] speciality of SIPp and SER(Sip Express Router)
Hi, SER is a most powerful SIP router but a SIPp is a VoIP load generation software. So both are totally different and can not be used interchangably. Regards, Faisal Hanif /VoIP Manager /**Vopium A/S// On 8/10/2010 10:44 AM, kamrun nahar bina wrote: Dear all, What is the difference between SIPp and SER(Sip Express Router)? Which one is better load performance testing? Is there any one who knows about this? Could you please give me details informtaion? Thans in advance Nahar -- _ -- 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
