Re: [asterisk-users] Early media and IAX2
On Tue, Aug 31, 2010 at 8:11 PM, Matt Riddell li...@venturevoip.com wrote: On 28/08/10 10:18 AM, Russ Dill wrote: My IAX2 trunk provider, Teliax, seems to be forcing early media. Early media is cool and all, but my Asterisk install doesn't seem to be fully supporting it. My initial setting was using Dial() to call all of my dahdi (TDM400P) extensions. The results were that incoming calls would not hear any ringing tones and the call would be ended by Teliax after 21 seconds. You could just answer the call before dialling your internal extensions. It was problem on Teliax's end. They were very responsive and took care of the issue quickly. I'm still confused as to why I couldn't get Asterisk to send ringing as early media. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Wanted: UK-specific hardware recommendations (FXOand FXS)
- Original Message - Roger Burton West wrote: I want to hook one of them to the PSTN. Given that I am in the UK, what is a reasonably easily-available device to provide an FXO interface from a Linux box, with a minimum of faffing around with drivers? Just one line is needed, though in theory two might eventually be useful. My usual white-box hardware suppliers don't seem to play in this field. I've had good experiences with an OpenVox A400P, once you've done the Dahdi dance, it settles down to be very reliable. Reasonable price, too. I bought mine from Voipon, although I'm sure a bit of shopping around will find other vendors. Cheers, Ade. Snap. Same card and supplier. Have had no issues at all. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Snom phones recommended firmware
We're using firmware 7.3.30 on an installation of Snom 300 phones. Should we stick with it, or do the newer firmwares have better support for Asterisk? Thanks John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Freepbx + Asterisk problem - NEED HELP
Sorry for unnecessary requests. I have read the book and it is related to the old version. Now I am studying the reference book included in the sources... Asterisk Reference Information Version 1.6.2.11 # make pdf and /usr/src/asterisk-1.6.2.6/doc/tex/asterisk.pdf is there for pdf people. On 2 September 2010 19:31, Mehmet Kuzulugil mehmetkuzulu...@gmail.comwrote: One more thing. Can anybody point me to a sample configuration for 2 PSTN lines and 2 internal phones. (May be plus a SIP server) for Asterisk 1.6.x On 1 September 2010 15:41, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Wed, Sep 01, 2010 at 02:58:59PM +0300, Mehmet Kuzulugil wrote: Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 How nice. Why don't you use the latest? 2.3.0[.1]? (2.4.0 should be released shortly, but that's not really something users are expected to guess). DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 4 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 The problem is about asterisk CLI results: asterisk*CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) asterisk*CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service What's the output of lsdahdi ? This suggests a misconfigured /etc/dahdi/chan_dahdi.conf or one of its --- INFO: related lspci result: 07:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface related dahdi_hardware result: pci::07:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I One more thing: r...@asterisk:~# lsmod|grep dahdi dahdi_echocan_mg2 5729 4 dahdi_transcode 6836 1 wctc4xxp dahdi_voicebus 41854 2 wctdm24xxp,wcte12xp dahdi 210885 12 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 1675 2 dahdi,hisax Haveing just 'wctdm' in /etc/dahdi/modules would save you a second or two on loading (and a bunch of log messages). it seems my dahdi/system.conf is ok. # Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) fxsks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 fxoks=3 echocanceller=mg2,3 fxoks=4 echocanceller=mg2,4 # Global data loadzone= tr defaultzone= tr And this is zapata.conf: zapata.conf is ignored for asterisk = 1.6.0 . Please tell me you don't use 1.4.x . [channels] language=en ; include zap extensions defined in AMP #include zapata_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n context=from-pstn faxdetect=incoming echotraining=800 group=0 busydetect=yes busycount=4 hanguponpolarityswitch relaxdtmf=yes callprogress=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=2.0 txgain=2.0 immediate=yes signalling=fxs_ks channel=1-2 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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:
Re: [asterisk-users] Snom phones recommended firmware
Hi! We're using firmware 7.3.30 on an installation of Snom 300 phones. Should we stick with it, or do the newer firmwares have better support for Asterisk? So what is it that you are missing that firmware 8 does offer? 7.3.30 is rather stable and therefore a good choice. Actually I would say this works the other way around and you could consider moving to Asterisk 1.8 soon to better match the Snom capabilites (some keywords: SRTP and TLS, P-Asserted header, AOC). That said there are interesting features that firmware 8.4.11/17 added: * support for 802.1X with eap-md5. * support for LLDP-MED (Link Layer Discovery Protocol-Media Endpoint Discovery). http://wiki.snom.com/Firmware/V8/Release_Notes/Change_Log_V8 Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manuplating Queue
Hi, I am implimenting a solution for a radio station where by calls are first received by an attendant, who interviews the caller and then places the call in a queue along with some information about the caller. The radio presenter can then choose which call to pick up depending on those in the queue. My question is, how can it be possible for call to skip other calls in the queue and be picked up? Are queues the best mothod of implimenting this? Thanks very much for your help. Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Manuplating Queue
Hello, We use call parking for this feature. It might not be the best solution, but it works quite well. We tweaked the parking values a little bit so that the parked callers don't timeout too quickly. The receptionist fills up a dynamic list that the presenter can consult, and knows which caller he picks up by dialing 701, 702, 703, etc. (call parking is at 700 in our setup). Of course, we would have loved to have a visual solution (a good receptionist console), but we don't have time to create one for our own usage, and many solutions over the web are not compatible with our Asterisk version (1.6.2.x). Hope this helps. Hoggins! Le 04/09/2010 14:42, Timothy Smith a écrit : Hi, I am implimenting a solution for a radio station where by calls are first received by an attendant, who interviews the caller and then places the call in a queue along with some information about the caller. The radio presenter can then choose which call to pick up depending on those in the queue. My question is, how can it be possible for call to skip other calls in the queue and be picked up? Are queues the best mothod of implimenting this? Thanks very much for your help. Tim signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] fast busy out?
why does this not work? i simply want to hear the recorded message exten = s,1,Answer() ;exten = s,n,Record(zipcodegutter1.gsm) ;zcg1 exten = s,n,Playback(zipcodegutter1) exten = s,n,Dial(SIP/c01s/159,120,A,(demo-thanks)) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Global Outage?
Is anyone else using Vitelity right now and having an issue with a global outage of sorts? Potral/WWW arent accessible and it would appear through monitoring that the outbound is flapipng like mad. The outbound can be rerouted, I know, but inbound is a huge problem right now. [Sep 4 10:26:13] NOTICE[27507]: chan_sip.c:15679 sip_poke_noanswer: Peer 'vitel-outbound' is now UNREACHABLE! Last qualify: 1193 [Sep 4 10:26:23] NOTICE[27507]: chan_sip.c:12528 handle_response_peerpoke: Peer 'vitel-outbound' is now Reachable. (176ms / 2000ms) --Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manuplating Queue
Thank you Hoggins! I am going to try it out and let you know. Regards, Tim On Sat, Sep 4, 2010 at 4:02 PM, Hoggins! fucks...@wheres5.com wrote: Hello, We use call parking for this feature. It might not be the best solution, but it works quite well. We tweaked the parking values a little bit so that the parked callers don't timeout too quickly. The receptionist fills up a dynamic list that the presenter can consult, and knows which caller he picks up by dialing 701, 702, 703, etc. (call parking is at 700 in our setup). Of course, we would have loved to have a visual solution (a good receptionist console), but we don't have time to create one for our own usage, and many solutions over the web are not compatible with our Asterisk version (1.6.2.x). Hope this helps. Hoggins! Le 04/09/2010 14:42, Timothy Smith a écrit : Hi, I am implimenting a solution for a radio station where by calls are first received by an attendant, who interviews the caller and then places the call in a queue along with some information about the caller. The radio presenter can then choose which call to pick up depending on those in the queue. My question is, how can it be possible for call to skip other calls in the queue and be picked up? Are queues the best mothod of implimenting this? Thanks very much for your help. Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Vitelity offline?
Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? Roger Marquis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] fast busy out?
I assume thtat you've already recorded the message, and the out commented the Record app.. try adding .gsm to the playback, to ensure that * doesn't look for other formats.. and according to http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record you should use file:format in the Record app, not file.format On Sat, Sep 4, 2010 at 3:40 PM, Thomas Perron thomas.per...@gmail.comwrote: why does this not work? i simply want to hear the recorded message exten = s,1,Answer() ;exten = s,n,Record(zipcodegutter1.gsm) ;zcg1 exten = s,n,Playback(zipcodegutter1) exten = s,n,Dial(SIP/c01s/159,120,A,(demo-thanks)) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Ondrej Škopek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fast busy out?
* then 2010/9/4 Ondrej Škopek skopekond...@gmail.com I assume thtat you've already recorded the message, and the out commented the Record app.. try adding .gsm to the playback, to ensure that * doesn't look for other formats.. and according to http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record you should use file:format in the Record app, not file.format On Sat, Sep 4, 2010 at 3:40 PM, Thomas Perron thomas.per...@gmail.comwrote: why does this not work? i simply want to hear the recorded message exten = s,1,Answer() ;exten = s,n,Record(zipcodegutter1.gsm) ;zcg1 exten = s,n,Playback(zipcodegutter1) exten = s,n,Dial(SIP/c01s/159,120,A,(demo-thanks)) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Ondrej Škopek -- -- Ondrej Škopek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fast busy out?
On 09/04/2010 08:40 PM, Thomas Perron wrote: why does this not work? i simply want to hear the recorded message exten = s,1,Answer() ;exten = s,n,Record(zipcodegutter1.gsm) ;zcg1 exten = s,n,Playback(zipcodegutter1) exten = s,n,Dial(SIP/c01s/159,120,A,(demo-thanks)) hi, try to put exten = s,n,Playback(silence/1) after Answer(), before your actual playback anton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Vitelity offline?
Looks like they have twitter. Its good that you mentioned them in the subject unlike the guy who wrote an hour and a half ago with subject Global Outage? http://twitter.com/vitelity http://twitter.com/vitelityWe are currently experiencing network difficulty on Vitelity's core router. We are working to resolve the issue. On Sat, Sep 4, 2010 at 8:52 AM, Roger Marquis marq...@roble.com wrote: Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? Roger Marquis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Vitelity offline?
Not that I'm aware of short of our direct contact. It would appear from the traceroutes that i've done this morning, that this appears to be a big part of the issue Tracing route to portal.vitelity.net [64.74.178.100] 1074 ms74 ms75 ms pos-1-14-0-0-cr01.denver.co.ibone.comcast.net[68.86.85.118] After that, the trace dies. On Sat, Sep 4, 2010 at 11:52 AM, Roger Marquis marq...@roble.com wrote: Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? Roger Marquis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fast busy out?
no I am not sorry, and please reply to this list, and not to me directly.. On Sat, Sep 4, 2010 at 6:16 PM, Thomas Perron thomas.per...@gmail.comwrote: thank you for your note on the Asterisk users group list Are you in Scandanavia somewhere? Cheers Tom -- -- Ondrej Škopek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity offline?
Roger Marquis wrote: Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? 09:30 PDT, Inland Northwest (Spokane, WA; Hayden, ID). I went directly to their website -- http://www.vitelity.net. Then called my business number and got through. Haven't tried calling out though. \\||/ Rod -- Roger Marquis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Vitelity offline?
Just a heads up. It would appear that Vitelity is back online and processing calls and the portal is back up and running. On Sat, Sep 4, 2010 at 12:14 PM, Matt Desbiens desbie...@gmail.com wrote: Not that I'm aware of short of our direct contact. It would appear from the traceroutes that i've done this morning, that this appears to be a big part of the issue Tracing route to portal.vitelity.net [64.74.178.100] 1074 ms74 ms75 ms pos-1-14-0-0-cr01.denver.co.ibone.comcast.net [68.86.85.118] After that, the trace dies. On Sat, Sep 4, 2010 at 11:52 AM, Roger Marquis marq...@roble.com wrote: Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? Roger Marquis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fast busy out?
Thank for your the tip Ondrej. Here is what worked on my CentOS box.
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Record(zipcodegutter%d:gsm)
exten = s,n,Wait(2)
exten = s,n,Playback(${RECORDED_FILE})
exten = s,n,Wait(2)
exten = s,n,Hangup()
2010/9/4 Ondrej Škopek skopekond...@gmail.com:
no I am not sorry, and please reply to this list, and not to me directly..
On Sat, Sep 4, 2010 at 6:16 PM, Thomas Perron thomas.per...@gmail.com
wrote:
thank you for your note on the Asterisk users group list
Are you in Scandanavia somewhere?
Cheers
Tom
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Re: [asterisk-users] How to finish an AGI
IMHO, is more easy in Perl that in dialplan but if for you work ..
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*
Date: Fri, 3 Sep 2010 10:29:02 +0200
From: ing.diasda...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to finish an AGI
Any particular reason you don't want to put the logic of the macro in your AGI?
Yes...i've no idea how to do it...it's a PERL script, i'm already checking how
to do this...but it will be a little complicated :(
2010/9/3 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias wrote:
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro from
my AGI, like this:
$agi-exec(Macro,check-call-limit);
If the Macro checks that the group_name is bigger than a number specified for
every peer with setvar it should Hangup the call (frobidden,1 in the Gotoif...)
but this
is not happening, the AGI always continue with is process and it doesn´t play
attention to the Hangup in the macro, the macro is here:
[macro-check-call-limit]
exten = s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten = s,n,Set(GROUP()=${group_name})
exten = s,n,GotoIf($[${GROUP_COUNT(${group_name})}
${MAX_OUT_CALLS_PER_USER}] forbidden,1)
; EXITO:
exten = s,n,MacroExit
; FRACASO:
exten = forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas
salientes)
exten = forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
The concept of calling a macro from within an AGI seem convoluted, but may
work. I've never tried it.
Any particular reason you don't want to put the logic of the macro in your AGI?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?
Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last frame transmitted by the carrier side endpoint, before the beginning of a period of discontinuous transmission has 20 bytes of payload. We have verified that VAD/DTX is used by the carrier side endpoint by noting that there exist successive RTP packets that differ by 1 in their sequence number but have a timestamp difference 160 and MARK bits are set in the RTP header. Our understanding is that for G729B, the SID frame that is transmitted before a period of discontinuous transmission has a size of 2 bytes. However we see that ALL RTP packets sent by the carrier side end point has a length of 20 bytes. Has anybody else seen this behavior from a carrier side endpoint ? Is there an RFC or document that specifies -- Thanks and Regards, Vikram Ragukumar. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to tell if there is a transfer from CDR?
Last time I analyzed this (I believe back in 1.2) there was no way of telling. However a blind transfered call would generate 2 CDR recoreds: 1. For the part of the call with the transferrer and transfered. 2. For the part of the call with the transferee and transfered. The call duration for the 2nd record would include the time of the 1st record as well. So if part one took 20 seconds and part 2 40 seconds, then the 2nd record would have 60 seconds as billable. The only workaround was to check the BLINDTRANSFER var and reset cdr if it was populated. Please members of this list, I would love to hear more input as I'm sure this has changed. Also I would not be surprised that I'm wrong in my analysis as more than 4 years has passed since and I might have forgotten. TIA On Fri, Sep 3, 2010 at 5:06 PM, Carlos Chavez cur...@telecomabmex.com wrote: Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?
On 09/05/2010 04:08 AM, Vikram Ragukumar wrote: Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last frame transmitted by the carrier side endpoint, before the beginning of a period of discontinuous transmission has 20 bytes of payload. We have verified that VAD/DTX is used by the carrier side endpoint by noting that there exist successive RTP packets that differ by 1 in their sequence number but have a timestamp difference 160 and MARK bits are set in the RTP header. Our understanding is that for G729B, the SID frame that is transmitted before a period of discontinuous transmission has a size of 2 bytes. However we see that ALL RTP packets sent by the carrier side end point has a length of 20 bytes. Has anybody else seen this behavior from a carrier side endpoint ? Is there an RFC or document that specifies Your understanding is correct. You need to infer from the length of the last frame being 2 bytes that it is a SID frame, and SID frames should only ever occur as the last frame in an RTP packet. If the SDP negotiation has agreed to used the annex B (CNG/DTX/VAD) option for G.729 you would normally expect to see a SID frame at the end of transmission. If the SDP negotiation has agree to do CNG/DTX/VAD by another means (which it can do) you won't see those SID frames. Even when annex B is used, I think some systems may miss out the SID frames. The use of proper annex B processing requires additional patent licence payments, and I suspect some people try to fudge things to save a little cost. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
