[asterisk-users] c option doesn't work if used with q option in meetme
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. Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] high cpu average load
Date: Thu, 5 Sep 2013 12:11:36 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] high cpu average load On Thu, 5 Sep 2013, Kamlesh Kumar wrote: Running one asterisk server with below details. Only SIP to SIP calls. No real time configuration, no recording, no voicemail, no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150 to 200 concurrent calls and we start getting problem in call quality like delay in connectivity, voice breakage etc Hardware: 2 Physical processor Intel(R) Xeon(R) CPU5120 @ 1.86GHz 8 GB RAM 500 GB Sata HDD Asterisk: 1.6.2.9 PHP 5.3.3 (cli) MySQL: 5.0.77 Linux: CnetOS 5.5 (Final) Please suggest the solution. Need a bit more detail. The 5120 is kind of a wimpy processor, but what is keeping it busy? What do 'top' and 'htop' show are consuming the processor? What is your application? What are 200 calls doing? Are you calling a bunch of AGIs written in scripting languages? Eliminating translation is difficult. How do you know you were successful? Do 'module show like codec_' and 'module show like format_' show anything unexpected? Below are the further details:top and htop shows that 'asterisk' is consuming the whole cpu power.Application: Kind of SIP trunking - call is coming from IP and using dialplan routed to other third party IPAre you calling a bunch of AGIs written in scripting languages? only one AGI script written in PHP is called with 'h' extension once the call is hungupmodule show like codec_ vm*CLI module show like codec_ Module Description Use Count codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0 codec_alaw.so A-law Coder/Decoder 0 codec_lpc10.so LPC10 2.4kbps Coder/Decoder 0 codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 codec_g722.so ITU G.722-64kbps G722 Transcoder 0 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 codec_ulaw.so mu-Law Coder/Decoder 0 codec_gsm.so GSM Coder/Decoder0 9 modules loadedmodule show like format_ vm*CLI module show like format_ Module Description Use Count format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_ogg_vorbis.so OGG/Vorbis audio 0 format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 format_g729.so Raw G729 data0 format_wav.so Microsoft WAV format (8000Hz Signed Line 0 format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 format_g723.so G.723.1 Simple Timestamp File Format 0 format_gsm.so Raw GSM data 0 format_vox.so Dialogic VOX (ADPCM) File Format 0 format_mp3.so MP3 format [Any rate but 8000hz mono is 0 10 modules loaded Thank you,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] high cpu average load
Hello, Running one asterisk server with below details. Only SIP to SIP calls. No real time configuration, no recording, no voicemail, no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150 to 200 concurrent calls and we start getting problem in call quality like delay in connectivity, voice breakage etc Hardware: 2 Physical processor Intel(R) Xeon(R) CPU5120 @ 1.86GHz 8 GB RAM 500 GB Sata HDD Asterisk: 1.6.2.9 PHP 5.3.3 (cli) MySQL: 5.0.77 Linux: CnetOS 5.5 (Final) Please suggest the solution. Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] server for 500 concurrent SIP calls
Hi, Asterisk 1.6.2.9 PHP 5.3 Mysql 5.0 Can anyone suggest hardware specification for 500 hundred concurrent SIP only calls, no codec transcoding, no IVR, no Voicemail or so. Just plain switching. There is only one requirement is to execute one php script on call hangup (h extension) which will do some calculation and update the CDRs. Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] limitation on number of contexts in extensions.conf
but it seems that value of variable defined in external file is not getting
populated during the dialplan execution.
My example:
extract from one external file in /etc/asterisk/abc.conf
PROV=1.2.3.4
[abc]
exten = _1X.,1,Dial(SIP/${PROV}/${EXTEN})
and extensions.conf contains:
[globals]
#include abc.conf
if call is made by the user of abc context, variable ${PROV} is having empty
value. Please suggest where is the problem.
Thanks,
Kamlesh
From: asterisk_l...@earthshod.co.uk
To: asterisk-users@lists.digium.com
Date: Fri, 26 Jul 2013 11:12:28 +0100
Subject: Re: [asterisk-users] limitation on number of contexts in
extensions.conf
On Friday 26 July 2013, Kamlesh Kumar wrote:
Thank you Carlos,
you were right, there was one empty file among all included files which
were causing this problem.
Couple of more queries:
Will system performance be affected if there are 20k dialplan
entries(including all external files and contexts) in extensions.conf?
Not by as much as you think, because the dialplan is compiled into an
intermediate form when Asterisk starts (and again when you execute `dialplan
reload`) -- it doesn't have to parse the whole text file for every call.
Can we define variable in external file, and include that external file in
extensions.conf and then use that variable in dialplan?
Yes (and that's a sensible way of doing it anyway). Just remember, a
variable won't have a value until the include statement which includes the
file
with the line that defines it is parsed.
--
AJS
Answers come *after* questions.
-
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Re: [asterisk-users] limitation on number of contexts in extensions.conf
Thank you Carlos, you were right, there was one empty file among all included files which were causing this problem. Couple of more queries: Will system performance be affected if there are 20k dialplan entries(including all external files and contexts) in extensions.conf? Can we define variable in external file, and include that external file in extensions.conf and then use that variable in dialplan? Thanks, Kamlesh Date: Thu, 25 Jul 2013 08:50:39 -0700 From: car...@televolve.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] limitation on number of contexts in extensions.conf On Wed, Jul 24, 2013 at 11:49 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include filename) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. There probably is a limit, but I don't know what it is. We have many hundreds of contexts and around 80 include files in our main server. My guess is you have an error somewhere. If you show dialplan, does it seem to stop at a certain point as if it loaded only up to a certain file/directory? -- Carlos AlvarezTelEvolve602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.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] limitation on number of contexts in extensions.conf
Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include filename) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. Regards, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] G.729 codec in pass-thru mode
Matthew, allow=all is defined in sip.conf for the ITSP's SIP peer. Additionally, ITSP supports g729 codec as we are able to send the traffic from other soft switch. In case g729 on asterisk box, as I mentioned earlier, call even doesn't go out of the asterisk box. Below extracts from log also indicate the same thing. [Jun 5 12:46:49] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [Jun 5 12:46:49] == Using SIP RTP CoS mark 5 [Jun 5 12:46:49] -- Couldn't call yyy.yyy.yyy.yyy/12127773456 [Jun 5 12:46:49] == Everyone is busy/congested at this time (0:0/0/0) Regards, Kamlesh Date: Tue, 4 Jun 2013 10:27:11 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Kamlesh Kumar wrote: SIP.conf [100] username=100 secret=password type=friend host=dynamic nat=yes canreinvite=no insecure=port disallow=all allow=ulaw allow=alaw allow=g729 context=asterisk qualify=no Is there also an allow=g729 line in sip.conf for the ITSP's SIP peer? SIP Trace: 201.xxx.xxx.xxx = SIP Softphone which originates the call xxx.xxx.xxx.xxx = Asterisk server yyy.yyy.yyy.yyy = ITSP ... --- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060 From: 100 sip:1...@xxx.xxx.xxx.xxx;tag=as643c20b1 To: sip:12127773...@yyy.yyy.yyy.yyy;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 INVITE Contact: sip:12127773...@yyy.yyy.yyy.yyy:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Content-Length: 234 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy s=SIP Media Capabilities c=IN IP4 zzz.zzz.zzz.zzz t=0 0 m=audio 21996 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - [Jun 3 13:11:31] --- (11 headers 11 lines) --- [Jun 3 13:11:31] Found RTP audio format 0 [Jun 3 13:11:31] Found RTP audio format 101 [Jun 3 13:11:31] Found audio description format PCMU for ID 0 [Jun 3 13:11:31] Found audio description format telephone-event for ID 101 [Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996 [Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-34d9 is making progress passing it to SIP/100-34d8 [Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042 [Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP [Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP This response from the ITSP says that only u-law may be used for the call. Please contact the ITSP and confirm that they actually support the G.729 codec. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems 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
Re: [asterisk-users] G.729 codec in pass-thru mode
] --- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK78ec518c;rport=5060 From: 100 sip:1...@xxx.xxx.xxx.xxx;tag=as643c20b1 To: sip:12127773...@yyy.yyy.yyy.yyy;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 103 BYE Content-Length: 0 Regards, Kamlesh Date: Fri, 31 May 2013 08:50:38 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Kamlesh Kumar wrote: Yes that's correct, when I use u-law call works fine. In case of g729, I enabled sip debug with 'sip set debug on' and captured all the sip traces and got whatever I posted in last email. There was no other call on the system when I captured sip trace. Please suggest further proceedings. Kamlesh, Please provide a SIP trace (sip set debug on) of a successful u-law call. I'm especially interested in the dialog between the Asterisk server and the ITSP in this scenario. Also include the relevant sections of sip.conf and the dialplan. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems 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
Re: [asterisk-users] G.729 codec in pass-thru mode
Matthew, Yes that's correct, when I use u-law call works fine. In case of g729, I enabled sip debug with 'sip set debug on' and captured all the sip traces and got whatever I posted in last email. There was no other call on the system when I captured sip trace. Please suggest further proceedings. Regards, Kamlesh Date: Wed, 29 May 2013 08:42:39 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Kamlesh Kumar wrote: Call even doesn't go to the ITSP. I tried without AGI script and the results were same. Kamlesh, Your first message stated that the call is successful if the codec is u-law, so there must be communication between the Asterisk server and the ITSP. The key to understanding why the G.729 call fails is in this SIP signaling. How are you capturing the SIP trace? Are you enabling SIP debugging for the specific SIP softphone? If so, please use sip set debug on to enable it for all SIP packets. Then wait until there are no other calls on the Asterisk server, try another G.729 call, and post the CLI output. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems 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
Re: [asterisk-users] G.729 codec in pass-thru mode
Hello Matthew, Call even doesn't go to the ITSP. I tried without AGI script and the results were same. Regards, Kamlesh Date: Tue, 28 May 2013 18:32:19 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Kamlesh, Please provide SIP traces of both call legs for a failed call. Your last message only included a SIP trace of the call leg from the SIP softphone to the Asterisk server. There was no SIP trace for the call leg from the Asterisk server to the ITSP and, as shown below, that is probably where the answer to your problem can be found. First, the call leg from the SIP softphone to the Asterisk server successfully negotiated G.729 as the codec: [May 28 11:51:34] Found RTP audio format 18 ... [May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) However, the call.php AGI script then failed to create the call leg from the Asterisk server to the ITSP: [May 28 11:51:34] -- Executing AGI(SIP/100-115f, call.php) [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php [May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [May 28 11:51:34] == Using SIP RTP CoS mark 5 [May 28 11:51:34] -- Couldn't call yyy.yyy.yyy.yyy/12127773456 [May 28 11:51:34] Scheduling destruction of SIP dialog '142182ef20750fda512f8d2b0b071...@xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE) [May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0) [May 28 11:51:34] -- SIP/100-115fAGI Script call.php completed, returning 0 [May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-115f' status is 'CHANUNAVAIL' Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems 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
Re: [asterisk-users] G.729 codec in pass-thru mode
/12127773456 [May 28 11:51:34] Scheduling destruction of SIP dialog '142182ef20750fda512f8d2b0b071...@xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE) [May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0) [May 28 11:51:34] -- SIP/100-115fAGI Script call.php completed, returning 0 [May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-115f' status is 'CHANUNAVAIL' [May 28 11:51:34] --- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --- SIP/2.0 503 Service Unavailable v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 f: 100sip:1...@xxx.xxx.xxx.xxx;tag=c4446262 t: sip:12127773...@xxx.xxx.xxx.xxx;tag=as4e329d09 i: 052fcf17df558f7b CSeq: 2 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO k: replaces, timer l: 0 [May 28 11:51:34] --- SIP read from UDP:201.xxx.xxx.xxx:5060 --- ACK sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0 To: sip:12127773...@xxx.xxx.xxx.xxx;tag=as4e329d09 From: 100sip:1...@xxx.xxx.xxx.xxx;tag=c4446262 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;rport Call-ID: 052fcf17df558f7b CSeq: 2 ACK Content-Length: 0 - [May 28 11:51:34] --- (7 headers 0 lines) --- [May 28 11:51:34] -- Executing AGI(SIP/100-115f, hangup.php) [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.php [May 28 11:51:34] -- SIP/100-115fAGI Script hangup.php completed, returning 0 Thanks, Kamlesh From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Mon, 27 May 2013 11:51:53 -0400 Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Show us the sip debug for a failed call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kamlesh Kumar Sent: Monday, May 27, 2013 2:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] G.729 codec in pass-thru mode Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)--Asterisk(1.6.2.9)---SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456@default:1] AGI(SIP/100-, call.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php -- AGI Script Executing Application: (Dial) Options: (SIP/xxx.xxx.xxx.xxx/12127773456) -- Couldn't call xxx.xxx.xxx.xxx/12127773456 == Everyone is busy/congested at this time (0:0/0/0) -- SIP/100-AGI Script call.php completed, returning 0 -- Auto fallthrough, channel 'SIP/100-' status is 'CHANUNAVAIL' If I use, ulaw, call works fine. Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] G.729 codec in pass-thru mode
Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)--Asterisk(1.6.2.9)---SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456@default:1] AGI(SIP/100-, call.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php -- AGI Script Executing Application: (Dial) Options: (SIP/xxx.xxx.xxx.xxx/12127773456) -- Couldn't call xxx.xxx.xxx.xxx/12127773456 == Everyone is busy/congested at this time (0:0/0/0) -- SIP/100-AGI Script call.php completed, returning 0 -- Auto fallthrough, channel 'SIP/100-' status is 'CHANUNAVAIL' If I use, ulaw, call works fine. Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] failed to extend from 512 to 676 on cli
Hello, We are using around 100 real time sip peers with phpagi. On asterisk cli, getting frequent message 'failed to extend from 512 to 676'. Once we execute 'sip reload', this message disappear for some time and then comes back. Please let us know the solution for this. asterisk version 1.6.2.9mysql 5.0server: Intel(R) Core(TM) i5-2500 CPU @ 3.30GHzRAM: 4 GB Thanks,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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 with 1000 extensions
2013/3/7 Steve Edwards asterisk@sedwards.com Please don't top-post. On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) I bet it is a school assignment ... home work or the way you like to call them. However I have a box with 972 peers, no reinvite (but no transcoding), average usage of conference call and other audio mix feature, reaching a max of 60 CPS and an average of 150 channels without problems. The cpu is a double Intel(R) Xeon(R) CPU E5-2630 0 @ 2.30GHz, but it works fine even on the old hardware, a double Intel(R) Xeon(R) CPU 5150 @ 2.66GHz Leandro, This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flatSecondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Thanks,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk with 1000 extensions
Hello, We need to setup asterisk server for 1000 extensions and in this setup only extension to extension dialling is required (without call recording and voicemail), like intercom calling. Please let us know what can be the best economic solution/setup for this. Thanks,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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 with 1000 extensions
Technology is SIP and asterisk is not handling the media, what is cheapest solution to be used for SIP client. Thanks,Kamlesh Date: Wed, 6 Mar 2013 20:43:52 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk with 1000 extensions On Thu, 7 Mar 2013, Kamlesh Kumar wrote: We need to setup asterisk server for 1000 extensions and in this setup only extension to extension dialling is required (without call recording and voicemail), like intercom calling. Please let us know what can be the best economic solution/setup for this. The number of extensions is not the key factor. The number of simultaneous calls is. What technology? SIP? Dahdi? If all you are going to do is call from endpoint to endpoint, maybe something like Kamailio or OpenSIPS is appropriate. If Asterisk is not handling the media, probably any old crappy computer can handle the call setup/call teardown load. -- 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
Re: [asterisk-users] asterisk with 1000 extensions
softphone is not going to be used in this setup. Hardphone is required. Around 60-70 simultaneous calls would be required. Thanks,Kamlesh Date: Wed, 6 Mar 2013 21:15:51 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk with 1000 extensions On Thu, 7 Mar 2013, Kamlesh Kumar wrote: Technology is SIP and asterisk is not handling the media, what is cheapest solution to be used for SIP client. Client? How about a free SIP softphone? Server? How many calls per second? How many simultaneous calls? Any half-way recent box should do. An Atom, i3, etc. Reliability and redundancy are going to be important unless you want 1,000 people calling you :) -- 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
Re: [asterisk-users] asterisk with 1000 extensions
Server side installation with recent hardware is fine, we can build two parallel system for redundancy. We are more concern with the cost of SIP client (hardphone). What are the various ways to make this setup functional with low cost for SIP clients. Thanks,Kamlesh On Thu, 7 Mar 2013, Kamlesh Kumar wrote: softphone is not going to be used in this setup. Hardphone is required. Around 60-70 simultaneous calls would be required. OK. So figure on about 6 UDP packets, about 3.5 KB per call. Not a big deal. I'd look for a reliable system and build 2 so you can swap between them as needed. Going the full redundant, heartbeat kind of setup may be more trouble than it is worth depending on how tolerant your users are to the very occasional outage. A couple of years ago, I bought a used Supermicro server with a 3.2 Ghz P4 off Ebay for $150 including shipping. Earlier this week, I updated the OS and rebooted it. The uptime was 574 days. -- 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] set time zone in sip debug logs
Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxx...@xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: xx sip:xxx...@xxx.xxx.xxx.xxx;tag=as23a29r59To: sip:xxx...@xxx.xxx.xxx.xxx:5060Contact: sip:xxx...@xxx.xxx.xxx.xxxCall-ID: 2f17b2103ea4792d571e2dce7e14b...@xxx.xxx.xxx.xxxCSeq: 102 INVITEUser-Agent: Asterisk PBX 1.6.2.9Date: Tue, 26 Feb 2013 04:54:29 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Type: application/sdpContent-Length: 444 Thanks,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] set time zone in sip debug logs
Hello Qasim, I need to change it permanently. System date/time is correct. INVITE header always follows GMT irrespective of system's date/time zone. It would be nice if you can mention the steps to sync the system and INVITE header time permanently. Thanks,Kamlesh Date: Tue, 26 Feb 2013 12:30:55 +0500 From: qasimak...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] set time zone in sip debug logs Hi Kamlesh, Asterisk give you very less control over SIP messaging. You can how ever add/remove/modify SIP headers from initial invite only. To modify a sip header you can use asterisk function SIP_HEADER(name). If you want to permanently change date why not change system date/time? Regards, -Qasim On Tue, Feb 26, 2013 at 11:13 AM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxx...@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rport Max-Forwards: 70 From: xx sip:xxx...@xxx.xxx.xxx.xxx;tag=as23a29r59 To: sip:xxx...@xxx.xxx.xxx.xxx:5060 Contact: sip:xxx...@xxx.xxx.xxx.xxx Call-ID: 2f17b2103ea4792d571e2dce7e14b...@xxx.xxx.xxx.xxx CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9 Date: Tue, 26 Feb 2013 04:54:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 444 Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] failed to extend from 512 to 676 message on console
Hello, Asterisk Version 1.6.2.9 on below hardware. We are using 100 Realtime SIP extensions. CPU : 1 x Intel® Core-i5 3.3 GHz. RAM : 4 GB DDR-3 SDRAM Hard Disk : 500 GB Hard Disk For last few days, getting below messages on asterisk cli. We googled to find the solution for this but could not locate the preventive steps. failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 Thanks,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] libpri error
Hello, Trying to install libpri version 1.4.11.3 on Centos 5.5. but getting below errors. Please suggest the resolution. gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT q921.o -MF .q921.o.d -MP -c -o q921.o q921.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT prisched.o -MF .prisched.o.d -MP -c -o prisched.o prisched.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT q931.o -MF .q931.o.d -MP -c -o q931.o q931.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT pri_facility.o -MF .pri_facility.o.d -MP -c -o pri_facility.o pri_facility.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT asn1_primitive.o -MF .asn1_primitive.o.d -MP -c -o asn1_primitive.o asn1_primitive.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose.o -MF .rose.o.d -MP -c -o rose.o rose.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_etsi_aoc.lo -MF .rose_etsi_aoc.lo.d -MP -c -o rose_etsi_aoc.lo rose_etsi_aoc.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_etsi_diversion.lo -MF .rose_etsi_diversion.lo.d -MP -c -o rose_etsi_diversion.lo rose_etsi_diversion.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_etsi_ect.lo -MF .rose_etsi_ect.lo.d -MP -c -o rose_etsi_ect.lo rose_etsi_ect.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_other.lo -MF .rose_other.lo.d -MP -c -o rose_other.lo rose_other.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_q931.lo -MF .rose_q931.lo.d -MP -c -o rose_q931.lo rose_q931.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_aoc.lo -MF .rose_qsig_aoc.lo.d -MP -c -o rose_qsig_aoc.lo rose_qsig_aoc.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_ct.lo -MF .rose_qsig_ct.lo.d -MP -c -o rose_qsig_ct.lo rose_qsig_ct.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_diversion.lo -MF .rose_qsig_diversion.lo.d -MP -c -o rose_qsig_diversion.lo rose_qsig_diversion.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_mwi.lo -MF .rose_qsig_mwi.lo.d -MP -c -o rose_qsig_mwi.lo rose_qsig_mwi.c ... Thanks,Kamlesh-- _ -- Bandwidth and Colocation Provided by http://www.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] libpri error
No, I'm trying first time. thanks,Kamlesh From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Mon, 30 Jul 2012 11:22:54 +0100 Subject: Re: [asterisk-users] libpri error On Monday 30 July 2012, Kamlesh Kumar wrote: Hello, Trying to install libpri version 1.4.11.3 on Centos 5.5. but getting below errors. Please suggest the resolution. That output doesn't look like error messages, but normal compilation output. If there is an actual error stopping it, we need to see the last few lines with the actual error message. Important side question: Have you ever successfully compiled libpri on this machine before? -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.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] libpri error
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_etsi_diversion.lo -MF .rose_etsi_diversion.lo.d -MP -c -o rose_etsi_diversion.lo rose_etsi_diversion.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_etsi_ect.lo -MF .rose_etsi_ect.lo.d -MP -c -o rose_etsi_ect.lo rose_etsi_ect.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_other.lo -MF .rose_other.lo.d -MP -c -o rose_other.lo rose_other.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_q931.lo -MF .rose_q931.lo.d -MP -c -o rose_q931.lo rose_q931.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_aoc.lo -MF .rose_qsig_aoc.lo.d -MP -c -o rose_qsig_aoc.lo rose_qsig_aoc.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_ct.lo -MF .rose_qsig_ct.lo.d -MP -c -o rose_qsig_ct.lo rose_qsig_ct.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_diversion.lo -MF .rose_qsig_diversion.lo.d -MP -c -o rose_qsig_diversion.lo rose_qsig_diversion.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_mwi.lo -MF .rose_qsig_mwi.lo.d -MP -c -o rose_qsig_mwi.lo rose_qsig_mwi.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_name.lo -MF .rose_qsig_name.lo.d -MP -c -o rose_qsig_name.lo rose_qsig_name.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT version.lo -MF .version.lo.d -MP -c -o version.lo version.c gcc -shared -Wl,-hlibpri.so.1.4 -o libpri.so.1.4 copy_string.lo pri.lo q921.lo prisched.lo q931.lo pri_facility.lo asn1_primitive.lo rose.lo rose_address.lo rose_etsi_aoc.lo rose_etsi_diversion.lo rose_etsi_ect.lo rose_other.lo rose_q931.lo rose_qsig_aoc.lo rose_qsig_ct.lo rose_qsig_diversion.lo rose_qsig_mwi.lo rose_qsig_name.lo version.lo /sbin/ldconfig -n . ln -sf libpri.so.1.4 libpri.so[root@localhost libpri-1.4.11.3]# thanks,Kamlesh To: asterisk-users@lists.digium.com From: asterisk_l...@earthshod.co.uk Date: Mon, 30 Jul 2012 11:49:30 +0100 Subject: Re: [asterisk-users] libpri error (Do not write anything before the original message. The proper place for a reply is *after* the thing you are replying to.) On Monday 30 July 2012, Kamlesh Kumar wrote: From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Mon, 30 Jul 2012 11:22:54 +0100 Subject: Re: [asterisk-users] libpri error On Monday 30 July 2012, Kamlesh Kumar wrote: Hello, Trying to install libpri version 1.4.11.3 on Centos 5.5. but getting below errors. Please suggest the resolution. That output doesn't look like error messages, but normal compilation output. If there is an actual error stopping it, we need to see the last few lines with the actual error message. Important side question: Have you ever successfully compiled libpri on this machine before? No, I'm trying first time. thanks,Kamlesh OK, then. What are the *last* few lines you get before it stops? (All the stuff you reproduced before was just normal compiler output. If there was an error at all, it would have been just before compilation failed.) -- AJS Price Engines Ltd. DDI: 01283 707058. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.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] libpri error
make install gives below output, is it also ok? [root@localhost libpri-1.4.11.3]# make install mkdir -p /usr/lib mkdir -p /usr/include install -m 644 libpri.h /usr/include install -m 755 libpri.so.1.4 /usr/lib #if [ -x /usr/sbin/sestatus ] ( /usr/sbin/sestatus | grep SELinux status: | grep -q enabled); then /sbin/restorecon -v /usr/lib/libpri.so.1.4; fi ( cd /usr/lib ; ln -sf libpri.so.1.4 libpri.so) install -m 644 libpri.a /usr/lib if test $(id -u) = 0; then /sbin/ldconfig -n /usr/lib; fi [root@localhost libpri-1.4.11.3]# thanks,Kamlesh From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Mon, 30 Jul 2012 11:58:44 +0100 Subject: Re: [asterisk-users] libpri error (Do not write anything before the original message! The proper place for a reply is *after* the thing you are replying to.) On Monday 30 July 2012, Kamlesh Kumar wrote: when I issue 'make' command, below output comes. [root@localhost libpri-1.4.11.3]# make gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT q921.o -MF .q921.o.d -MP -c -o q921.o q921.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT prisched.o -MF .prisched.o.d -MP -c -o prisched.o prisched.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT q931.o -MF .q931.o.d -MP -c -o q931.o q931.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT pri_facility.o -MF .pri_facility.o.d -MP -c -o pri_facility.o pri_facility.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT asn1_primitive.o -MF .asn1_primitive.o.d -MP -c -o asn1_primitive.o asn1_primitive.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose.o -MF .rose.o.d -MP -c -o rose.o rose.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_address.o -MF .rose_address.o.d -MP -c -o rose_address.o rose_address.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_etsi_aoc.o -MF .rose_etsi_aoc.o.d -MP -c -o rose_etsi_aoc.o rose_etsi_aoc.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_etsi_diversion.o -MF .rose_etsi_diversion.o.d -MP -c -o rose_etsi_diversion.o rose_etsi_diversion.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_etsi_ect.o -MF .rose_etsi_ect.o.d -MP -c -o rose_etsi_ect.o rose_etsi_ect.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_other.o -MF .rose_other.o.d -MP -c -o rose_other.o rose_other.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_q931.o -MF .rose_q931.o.d -MP -c -o rose_q931.o rose_q931.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_aoc.o -MF .rose_qsig_aoc.o.d -MP -c -o rose_qsig_aoc.o rose_qsig_aoc.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_ct.o -MF .rose_qsig_ct.o.d -MP -c -o rose_qsig_ct.o rose_qsig_ct.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_diversion.o -MF .rose_qsig_diversion.o.d -MP -c -o rose_qsig_diversion.o rose_qsig_diversion.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_mwi.o -MF .rose_qsig_mwi.o.d -MP -c -o rose_qsig_mwi.o rose_qsig_mwi.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT rose_qsig_name.o -MF .rose_qsig_name.o.d -MP -c -o rose_qsig_name.o rose_qsig_name.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT version.o -MF .version.o.d -MP -c -o version.o version.c ar rcs libpri.a copy_string.o pri.o q921.o prisched.o q931.o pri_facility.o asn1_primitive.o rose.o rose_address.o rose_etsi_aoc.o rose_etsi_diversion.o rose_etsi_ect.o rose_other.o rose_q931.o rose_qsig_aoc.o rose_qsig_ct.o rose_qsig_diversion.o rose_qsig_mwi.o rose_qsig_name.o version.o ranlib libpri.a gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT copy_string.lo -MF .copy_string.lo.d -MP -c -o copy_string.lo copy_string.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT pri.lo -MF .pri.lo.d -MP -c -o pri.lo pri.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT q921.lo -MF .q921.lo.d -MP -c -o q921.lo q921.c gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT prisched.lo -MF .prisched.lo.d
Re: [asterisk-users] voicemail password with phone instrument
still waiting for valuable reply on this. Regards,Kamlesh From: kamlesh_...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 15 Jun 2012 12:19:48 + Subject: [asterisk-users] voicemail password with phone instrument Hello, voicemail password is not getting changed through phone handset while IVR indicates that password has been changed. During google I found that uniqueid column must not be changed so it is not changed. Please guide on this. During debug log I found below but in mysql db new password is not getting updated, [Jun 15 13:54:07] VERBOSE[6418] file.c: -- SIP/123-0005 Playing 'vm-newpassword.gsm' (language 'en') [Jun 15 13:54:10] DEBUG[6418] channel.c: Set channel SIP/123-0005 to write format ulaw [Jun 15 13:54:15] VERBOSE[6418] file.c: -- SIP/123-0005 Playing 'vm-reenterpassword.gsm' (language 'en') [Jun 15 13:54:22] DEBUG[6418] app_voicemail.c: User 123 set password to of length 4 [Jun 15 13:54:22] DEBUG[6418] channel.c: Set channel SIP/123-0005 to write format gsm [Jun 15 13:54:22] VERBOSE[6418] file.c: -- SIP/123-0005 Playing 'vm-passchanged.gsm' (language 'en') Regards, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] voicemail password with phone instrument
Hello, voicemail password is not getting changed through phone handset while IVR indicates that password has been changed. During google I found that uniqueid column must not be changed so it is not changed. Please guide on this. During debug log I found below but in mysql db new password is not getting updated, [Jun 15 13:54:07] VERBOSE[6418] file.c: -- SIP/123-0005 Playing 'vm-newpassword.gsm' (language 'en') [Jun 15 13:54:10] DEBUG[6418] channel.c: Set channel SIP/123-0005 to write format ulaw[Jun 15 13:54:15] VERBOSE[6418] file.c: -- SIP/123-0005 Playing 'vm-reenterpassword.gsm' (language 'en')[Jun 15 13:54:22] DEBUG[6418] app_voicemail.c: User 123 set password to of length 4 [Jun 15 13:54:22] DEBUG[6418] channel.c: Set channel SIP/123-0005 to write format gsm [Jun 15 13:54:22] VERBOSE[6418] file.c: -- SIP/123-0005 Playing 'vm-passchanged.gsm' (language 'en') Regards,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] use of Read cmd with AGI
Hello Steve, it's working fine, thanks for your suupport. thanks,Kamlesh Date: Tue, 22 May 2012 10:36:20 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] use of Read cmd with AGI Un-top-posting... From: alejandro.belt...@setcolombia.com Hi, try some like this: [PERL snippet using get_data AGI command] On Tue, 22 May 2012, Kamlesh Kumar wrote: I tried it but it doesn't work. beep file gets played, and when I enter any digit(s), it doesn't get stored in $keys variable. 1) Does enabling AGI debugging on the Asterisk console shed any clues? 2) Try reducing your AGI script to the bare minium. 3) Post the full source of your AGI and the Asterisk console log with AGI debugging enabled. -- 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] extension status using AMI
Hi, I'm using AMI to get the extension status but always get -1 i.e. extension not found. #!/usr/bin/php -q ?phpinclude_once (phpagi-2.14/phpagi.php); include_once (/phpagi-2.14/phpagi-asmanager.php); $agi = new AGI(); $as = new AGI_AsteriskManager(); $exten = $agi-request['agi_extension'];$as-connect(localhost, user, passwd);$status = $as-ExtensionState($exten,'context',1); $status1 = $status['Status']; $agi-verbose(Extension status is .$status1);? Always return Extension status is -1 Thanks,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime configuration for /etc/dahdi/system.conf
Hi, can we load the settings of /etc/dahdi/system.conf from database table in real time. thanks,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] realtime config for general settings in sip.conf
contrib/realtime/ directory talks about sip peer/client parameters not general section(sip.conf) parameters like bindaddr, bindport, domain, realm, qualify etc... thanks, Kamlesh From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Thu, 3 May 2012 08:39:28 +0100 Subject: Re: [asterisk-users] realtime config for general settings in sip.conf You need 2 but they can point to the same table sipusers = sippeers = You can get table definitions by downloading the source and then looking in the contrib/realtime/ directory Ish On Thu, 2012-05-03 at 04:56 +, Kamlesh Kumar wrote: Hello, For realtime configuration, in /etc/asterisk/extconfig.conf file, what should be the family name to configure general sip.conf parameters. family name = driver,database name,table name thanks, Kamlesh From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Date: Wed, 2 May 2012 13:59:58 +0100 Subject: Re: [asterisk-users] realtime config for general settings in sip.conf On Wed, 2012-05-02 at 12:04 +, Kamlesh Kumar wrote: Hi, I need to configure global parameters in sip.conf like rtptimeout, rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time architecture. Please suggest the way to do it. thanks, Kamlesh Hi You can set defaults in the column definitions and you can still set globals in the sip.conf Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.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 -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.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] realtime config for general settings in sip.conf
Hi, I need to configure global parameters in sip.conf like rtptimeout, rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time architecture. Please suggest the way to do it. thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] use of Read cmd with AGI
Hello,
Using AGI script to accept the input from caller but input value is not getting
stored in variable.
Extract from AGI Script:
$agi = new AGI();
$agi- exec('Background','press_one0press_two0press_zero0');
$agi- exec('Read','NUMBER,,1,3');
$agi- verbose (You have entered.$NUMBER);
Console Output:
AGI Script Executing Application: (Background) Options:
(press_one0press_two0press_zero0)
-- DAHDI/21-1 Playing 'press_one0.gsm' (language 'en')
-- DAHDI/21-1 Playing 'press_two0.gsm' (language 'en')
-- DAHDI/21-1 Playing 'press_zero0.gsm' (language 'en')
-- AGI Script Executing Application: (Read) Options: (NUMBER,,1,3)
-- Accepting a maximum of 1 digits.
-- User entered '1'
pridid.php: You have entered
[Mar 2 16:56:37] ERROR[20454]: utils.c:1128 ast_carefulwrite: write() returned
error: Broken pipe
-- DAHDI/21-1AGI Script pridid.php completed, returning 0
Thanks,
Kamlesh
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[asterisk-users] error during dahdi installation on centos
Hello, # rpm -qa | grep kernel kernel-headers-2.6.18-274.18.1.el5 kernel-PAE-2.6.18-128.el5 kernel-devel-2.6.18-274.18.1.el5 kernel-PAE-devel-2.6.18-274.18.1.el5 [root@localhost ~]# uname -i i386 Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below error. Can you please assist in this? [root@localhost dahdi-linux-complete-2.3.0.1+2.3.0]# make make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-128.el5PAE kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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 during dahdi installation on centos
thank you very much for your quick response. make KVERS=2.6.18-274.18.1.el5PAE It started the installation but stuck at below error LD [M] /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/wcte12xp/wcte12xp.o CC [M] /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/card_bri.o In file included from /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/xpd.h:31, from /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/card_bri.c:29: include/linux/device.h:407: error: expected identifier or â(â before âconstâ make[4]: *** [/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/card_bri.o] Error 1 make[3]: *** [/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp] Error 2 make[2]: *** [_module_/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-274.18.1.el5-PAE-i686' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Regards, Kamlesh Date: Wed, 15 Feb 2012 14:39:00 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] error during dahdi installation on centos On Wed, Feb 15, 2012 at 12:11:18PM +, Kamlesh Kumar wrote: Hello, # rpm -qa | grep kernel kernel-headers-2.6.18-274.18.1.el5 kernel-PAE-2.6.18-128.el5 kernel-devel-2.6.18-274.18.1.el5 kernel-PAE-devel-2.6.18-274.18.1.el5 [root@localhost ~]# uname -i i386 Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below error. Can you please assist in this? [root@localhost dahdi-linux-complete-2.3.0.1+2.3.0]# make make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-128.el5PAE kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Boot to the newer kernel and/or use: make KVERS=2.6.18-274.18.1.el5PAE -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.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] DIALSTATUS Values
Can anybody please reply on this? Regards, Kamlesh From: kamlesh_...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 27 Dec 2011 09:49:21 + Subject: Re: [asterisk-users] DIALSTATUS Values Hello, After investing some time, I could come to know the reason for not getting the data value is that if I use system command with any of asterisk cli command as given below, data value is returned blank. $output=system(/usr/sbin/asterisk -rx 'sip show peers' | grep OK | cut -f 1 -d / | grep '100' ) Could you please suggest now how to rectify this? Regards, Kamlesh To: asterisk-users@lists.digium.com From: t...@softins.co.uk Date: Fri, 2 Dec 2011 12:27:19 + Subject: Re: [asterisk-users] DIALSTATUS Values In article snt142-w54267269808afd17bccd5891...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: In addition to my reply: I used to fetch the value using print_r function but that also tells that there is no value in data section. $dialstatus=$agi-get_variable(DIALSTATUS); print_r($dialstatus); SIP/10036-00b8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b8AGI Tx 200 result=1 (CANCEL) SIP/10036-00b8AGI Rx Array SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx ( SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [code] = 200 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [result] = 1 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [data] = Well since the AGI return string does indeed contain the value, shown above as (CANCEL), that suggests there is definitely a bug in php-agi. It appears to be creating a ['data'] element, but not setting it. You will have to study the source code and work out how to fix it. I did a quick google for php agi get variable and found other reports of it not working properly, but I didn't see anyone offer a solution. It's only programming, so it shouldn't be hard to fix. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.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] read dtmf digits on connected calls
Hello, I need to capture the DTMF digits dialled by user on current connected calls and store them in variable. scenario: Manual Call Transfer: User A dialed to B B answered the call and want to transfer the call to user C manually. User B dials *2 to get the ring tone again and then dial to 3. This works but I want to capture the digits *2 dialed by User B in some variable. Please suggest. Regards, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] execute command just after Dial()
hangup extension works once the call is terminated but I want to know the
status of call immediately after connected, cancelled, or rejected and so on.
thanks,
Kamlesh
Date: Tue, 27 Dec 2011 16:59:35 +0530
From: dhaval.it01...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] execute command just after Dial()
You can also try special extension hangup and manage your scenario
On Sat, Dec 24, 2011 at 12:44 PM, Sammy Govind govoi...@gmail.com wrote:
Hi,
Please see the Dial application documents from CLI, i.e core show application
dial. There is an option which will let you continue in the DIal-plan after
the Dial command on hangup.
Regards,
Sammy.
On Fri, Dec 23, 2011 at 5:54 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
Hello,
I'm using AGI scripting with asterisk and need to execute certain commands just
after Dial(). But once dial command is executed, further commands/instructions
are ignored.
$agi-exec(Dial,SIP/100);
$dialstatus = $agi - get_variable(DIALSTATUS);
if($dialstatus[data]==ANSWER)
{
do something...
}
thanks,
Kamlesh
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Re: [asterisk-users] DIALSTATUS Values
Hello, After investing some time, I could come to know the reason for not getting the data value is that if I use system command with any of asterisk cli command as given below, data value is returned blank. $output=system(/usr/sbin/asterisk -rx 'sip show peers' | grep OK | cut -f 1 -d / | grep '100' ) Could you please suggest now how to rectify this? Regards, Kamlesh To: asterisk-users@lists.digium.com From: t...@softins.co.uk Date: Fri, 2 Dec 2011 12:27:19 + Subject: Re: [asterisk-users] DIALSTATUS Values In article snt142-w54267269808afd17bccd5891...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: In addition to my reply: I used to fetch the value using print_r function but that also tells that there is no value in data section. $dialstatus=$agi-get_variable(DIALSTATUS); print_r($dialstatus); SIP/10036-00b8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b8AGI Tx 200 result=1 (CANCEL) SIP/10036-00b8AGI Rx Array SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx ( SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [code] = 200 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [result] = 1 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [data] = Well since the AGI return string does indeed contain the value, shown above as (CANCEL), that suggests there is definitely a bug in php-agi. It appears to be creating a ['data'] element, but not setting it. You will have to study the source code and work out how to fix it. I did a quick google for php agi get variable and found other reports of it not working properly, but I didn't see anyone offer a solution. It's only programming, so it shouldn't be hard to fix. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] execute command just after Dial()
Hello,
I'm using AGI scripting with asterisk and need to execute certain commands just
after Dial(). But once dial command is executed, further commands/instructions
are ignored.
$agi-exec(Dial,SIP/100);
$dialstatus = $agi - get_variable(DIALSTATUS);
if($dialstatus[data]==ANSWER)
{
do something...
}
thanks,
Kamlesh --
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Re: [asterisk-users] get start-time of all active calls
Hello, 'sip show channel' also does not give this info. sip show channel f600ed29f561d57 localhost*CLI * SIP CallI Curr. trans. direction: Incoming Call-ID:f600ed29f561d57f Owner channel ID: SIP/100- Our Codec Capability: 14 Non-Codec Capability (DTMF): 1 Their Codec Capability: 302 Joint Codec Capability: 14 Format: 0x2 (gsm) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open NAT Support:Always Audio IP: xxx.xxx.xxx.xxx (local) Our Tag:as2a60820a Their Tag: 1b7d6a7d SIP User agent: eyeBeam release 3007n stamp 17816 Username: 10036 Peername: 10036 Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 Caller-ID: 100 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:1...@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive regards, Kamlesh Date: Wed, 14 Dec 2011 12:43:14 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get start-time of all active calls
finally I got it with 'core show channel' channel-id thanks for your support. Date: Wed, 14 Dec 2011 15:11:49 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls oops, you got it. On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield t...@softins.co.uk wrote: In article CAJUJwthT=mpyxq+omt5hrextl1iqvd0kbs+jhtqlvsqscay...@mail.gmail.com, Sammy Govind govoi...@gmail.com wrote: Hi, Not sure why you didnt get it, when I did thta command for originator channel it showed me the CDR variables list which included That's from show channel, not sip show channel. Cheers Tony CDR Variables: level 1: dnid= level 1: clid=XXX level 1: src= level 1: dst= level 1: dcontext=SIP-incoming level 1: channel= level 1: dstchannel= level 1: lastapp=Dial level 1: lastdata=SIP/ *level 1: start=2011-12-14 09:15:54* level 1: answer=2011-12-14 09:16:01 level 1: duration=11 level 1: billsec=4 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1323854154.856 level 1: linkedid=1323854154.856 level 1: sequence=1096 Thats valid for an ongoing bridged call-initiator side only. Regards, Sammy On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, 'sip show channel' also does not give this info. sip show channel f600ed29f561d57 localhost*CLI * SIP CallI Curr. trans. direction: Incoming Call-ID:f600ed29f561d57f Owner channel ID: SIP/100- Our Codec Capability: 14 Non-Codec Capability (DTMF): 1 Their Codec Capability: 302 Joint Codec Capability: 14 Format: 0x2 (gsm) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open NAT Support:Always Audio IP: xxx.xxx.xxx.xxx (local) Our Tag:as2a60820a Their Tag: 1b7d6a7d SIP User agent: eyeBeam release 3007n stamp 17816 Username: 10036 Peername: 10036 Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 Caller-ID: 100 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:1...@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive regards, Kamlesh -- Date: Wed, 14 Dec 2011 12:43:14 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=- -=-=-=-=-=- -- _ -- Bandwidth and Colocation Provided by http://www.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 -=-=-=-=-=- -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t
[asterisk-users] get start-time of all active calls
Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] DIALSTATUS Values
Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1 Please help me in this. Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.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] DIALSTATUS Values
Hello, in /etc/extension.conf [privoip] exten = _00X.,n,AGI(isdcall.php) Regards, Kamlesh Date: Fri, 2 Dec 2011 16:16:27 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS Values Hi, How are you calling this AGI in your dialplan !!? Regards, Sammy. On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1 Please help me in this. Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS Values
Here it is: SIP/10036-00a8AGI Tx agi_request: isdcall.php SIP/10036-00a8AGI Tx agi_channel: SIP/10036-00a8 SIP/10036-00a8AGI Tx agi_language: en SIP/10036-00a8AGI Tx agi_type: SIP SIP/10036-00a8AGI Tx agi_uniqueid: 1322853473.198 SIP/10036-00a8AGI Tx agi_version: 1.6.2.7 SIP/10036-00a8AGI Tx agi_callerid: 10036 SIP/10036-00a8AGI Tx agi_calleridname: 10036 SIP/10036-00a8AGI Tx agi_callingpres: 0 SIP/10036-00a8AGI Tx agi_callingani2: 0 SIP/10036-00a8AGI Tx agi_callington: 0 SIP/10036-00a8AGI Tx agi_callingtns: 0 SIP/10036-00a8AGI Tx agi_dnid: 0012127773456 SIP/10036-00a8AGI Tx agi_rdnis: unknown SIP/10036-00a8AGI Tx agi_context: privoip SIP/10036-00a8AGI Tx agi_extension: 0012127773456 SIP/10036-00a8AGI Tx agi_priority: 3 SIP/10036-00a8AGI Tx agi_enhanced: 0.0 SIP/10036-00a8AGI Tx agi_accountcode: 10036 SIP/10036-00a8AGI Tx agi_threadid: -1220478064 SIP/10036-00a8AGI Rx VERBOSE 10036 1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Rx VERBOSE 0012127773456 1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Rx VERBOSE 10036 1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Rx VERBOSE Dialling 1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Rx EXEC Dial SIP/202.89.78.21/12127773456 SIP/10036-00a8AGI Tx 200 result=-1 SIP/10036-00a8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00a8AGI Tx 200 result=1 (ANSWER) SIP/10036-00a8AGI Rx VERBOSE Status 1 SIP/10036-00a8AGI Tx 200 result=1 Regards, Kamlesh Date: Fri, 2 Dec 2011 16:26:50 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS Values Can you also paste the Asterisk Console logs around the part where AGI is dialing and after the dialing part ! make sure AGi debug is enabled as well. On Fri, Dec 2, 2011 at 4:24 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, in /etc/extension.conf [privoip] exten = _00X.,n,AGI(isdcall.php) Regards, Kamlesh Date: Fri, 2 Dec 2011 16:16:27 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS Values Hi, How are you calling this AGI in your dialplan !!? Regards, Sammy. On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1 Please help me in this. Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS Values
I believe the syntax is correct because, If I use $dd=$dialstatus[code]; $agi-verbose(Status.$dd); it gives me: SIP/10036-00b2AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b2AGI Tx 200 result=1 (ANSWER) SIP/10036-00b2AGI Rx VERBOSE Status200 1 If I use $dd=$dialstatus[result]; $agi-verbose(Status.$dd); it gives me: SIP/10036-00b4AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b4AGI Tx 200 result=1 (CANCEL) SIP/10036-00b4AGI Rx VERBOSE Status1 1 but if I use $dd=$dialstatus[data]; $agi-verbose(Status.$dd); SIP/10036-00b6AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b6AGI Tx 200 result=1 (CANCEL) SIP/10036-00b6AGI Rx VERBOSE Status 1 Regards, Kamlesh To: asterisk-users@lists.digium.com From: t...@softins.co.uk Date: Fri, 2 Dec 2011 11:44:34 + Subject: Re: [asterisk-users] DIALSTATUS Values In article snt142-w45a64e4743de653da591...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); Shouldn't that be: $dialstatus=$agi-get_variable(DIALSTATUS); Having DIALSTATUS as a bare word might work in some versions of php, but is likely to produce a warning. Although in your case, it does appear to have worked. $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) This shows that AGI is indeed returning the value of DIALSTATUS, which is ANSWER. SIP/10036-0096AGI Rx VERBOSE Status 1 But you are not picking it up. SIP/10036-0096AGI Tx 200 result=1 Please help me in this. I'm not familiar with php-agi (I usualy write my AGI in C), but it looks like $dialstatus[data] is not the correct way to retrieve the returned value. Or else there is a bug in php-agi. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS Values
In addition to my reply: I used to fetch the value using print_r function but that also tells that there is no value in data section. $dialstatus=$agi-get_variable(DIALSTATUS); print_r($dialstatus); SIP/10036-00b8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b8AGI Tx 200 result=1 (CANCEL) SIP/10036-00b8AGI Rx Array SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx ( SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [code] = 200 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [result] = 1 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [data] = Regards, Kamlesh From: kamlesh_...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] DIALSTATUS Values Date: Fri, 2 Dec 2011 11:58:26 + I believe the syntax is correct because, If I use $dd=$dialstatus[code]; $agi-verbose(Status.$dd); it gives me: SIP/10036-00b2AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b2AGI Tx 200 result=1 (ANSWER) SIP/10036-00b2AGI Rx VERBOSE Status200 1 If I use $dd=$dialstatus[result]; $agi-verbose(Status.$dd); it gives me: SIP/10036-00b4AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b4AGI Tx 200 result=1 (CANCEL) SIP/10036-00b4AGI Rx VERBOSE Status1 1 but if I use $dd=$dialstatus[data]; $agi-verbose(Status.$dd); SIP/10036-00b6AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b6AGI Tx 200 result=1 (CANCEL) SIP/10036-00b6AGI Rx VERBOSE Status 1 Regards, Kamlesh To: asterisk-users@lists.digium.com From: t...@softins.co.uk Date: Fri, 2 Dec 2011 11:44:34 + Subject: Re: [asterisk-users] DIALSTATUS Values In article snt142-w45a64e4743de653da591...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); Shouldn't that be: $dialstatus=$agi-get_variable(DIALSTATUS); Having DIALSTATUS as a bare word might work in some versions of php, but is likely to produce a warning. Although in your case, it does appear to have worked. $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) This shows that AGI is indeed returning the value of DIALSTATUS, which is ANSWER. SIP/10036-0096AGI Rx VERBOSE Status 1 But you are not picking it up. SIP/10036-0096AGI Tx 200 result=1 Please help me in this. I'm not familiar with php-agi (I usualy write my AGI in C), but it looks like $dialstatus[data] is not the correct way to retrieve the returned value. Or else there is a bug in php-agi. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org
