Re: [asterisk-users] dialplan reload: not working with large dialplans
Lief, I'd be glad to receive your feedback. I don't think it's a limit of lines by itself. I haven't found any useful debug information so far, but I think the dialplan parser stumbles over something. The problem is reproduceable on different hareware, it only sticks to newer 1.6 releases. Up until 1.6.0 I haven't seen this problem with the same dialplan. When I go down to 1.6.0 I don't have this problem anymore. -Andreas 2010/3/5 Leif Madsen leif.mad...@asteriskdocs.org Andreas Brodmann wrote: the dialplan currently holds 1792 lines. It's a plain old .conf file. That's interesting, because I have a dialplan over 2400 lines and it seems to load fine... However, I'm using this on an ABE machine which is based on 1.4. Perhaps I'll try loading this on a 1.6.2 box tomorrow. Leif! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5
Hi all, We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to any newer releases: We use the following cli command to feed a wave/mp3 file into an existing conference on an other serve: /opt/asterisk/sbin/asterisk -r -x channel originate Local/confgongad...@xy_features extension confgongp...@xy_features The corresponding extensions.conf part looks like that: -- [XY_Features] exten = ConfGongAdmin,1,NoCDR() exten = ConfGongAdmin,n,Set(TIMEOUT(absolute)=10) exten = ConfGongAdmin,n,Dial(SIP/12...@server) exten = ConfGongPlay,1,Answer() exten = ConfGongPlay,n,Set(TIMEOUT(absolute)=10) exten = ConfGongPlay,n,Wait(2) exten = ConfGongPlay,n,Playback(/etc/asterisk/sounds/gong) --- Until asterisk-1.6.2.0 this worked fine. With later releases including 1.6.2.5 asterisk does a call to confgongad...@xy_features but once that stands does not continue with a call to ConfGongPlay. Our asterisk system is a pure asterisk installation, no dahdi drivers for timing, as we don't have zaptel/dahi hardware. What we basically do is we try to play a sound file into an existing conference on another server. We have also tried to do the same thing with the ConfBridge application but have found so far that ConfBridge only works with phones, e.g. stations that provide RTP which asterisk can use for timing. When we try to play a sound file into such a conference from the same server asterisk won't play anything. Maybe I am just doing it wrong. Any suggestions or help would be appreciated. Andreas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] dialplan reload: not working with large dialplans
There is a problem that bothered me for a long time: Since one of the 1.6.0.x patch releases up until 1.6.2.5 a dialplan reload works only once with a bigger dialplan. If I issue dialplan reload again, it won't do anything. After doing so the cli won't show responses to any commands anymore. So if I have to do another change to the dialplan, I have to stop/start asterisk. Did anyone encounter a similar issue? -Andreas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] dialplan reload: not working with large dialplans
Hi Tzafrir, yes, I will have to 'anonymize' the dialplan, is this list the right place though? -Andreas 2010/3/2 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Mar 02, 2010 at 01:56:37PM +0100, Andreas Brodmann wrote: There is a problem that bothered me for a long time: Since one of the 1.6.0.x patch releases up until 1.6.2.5 a dialplan reload works only once with a bigger dialplan. If I issue dialplan reload again, it won't do anything. After doing so the cli won't show responses to any commands anymore. So if I have to do another change to the dialplan, I have to stop/start asterisk. Did anyone encounter a similar issue? Can you post somewhere a dialplan that reproduces this? -- 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: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reload: not working with large dialplans
Hi Warren, the dialplan currently holds 1792 lines. It's a plain old .conf file. -Andreas 2010/3/2 Warren Selby wcse...@selbytech.com On Tuesday, March 2, 2010, Andreas Brodmann andreas.brodm...@gmail.com wrote: Hi Tzafrir, yes, I will have to 'anonymize' the dialplan, is this list the right place though? -Andreas How big is your dialplan? How many lines / file size, etc. Are you using ael or lua or just the original .conf file? Thanks, --Warren Selby http://www.SelbyTech.com -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call-leg stays on MusicOnHold forever
Hi I have a strange behaviour; perhaps someone who had a similar issue can help. I have an Asterisk-1.4.21.2 connected via sip trunk to a Cisco Call-Manager 6.1 cluster. Two phones/users from the Cisco environment call extensions on the Asterisk. Phone 1 / Call 1 is parked on the asterisk using: exten = xyz,1,Answer() exten = xyz,n,Set(PARKEXTENSION=555) exten = xyz,n,Park() Phone 2 / Call 2 is picking it up: exten = xyz,1,Answer() exten = xyz,n,ParkedCall(555) so far so good, they can talk to each other. Now if one of them presses Hold, Asterisk will: [Sep 5 14:16:05] VERBOSE[6351] logger.c: -- Started music on hold, class 'default', on SIP/10.16.17.162-081bb720 [Sep 5 14:16:05] VERBOSE[6351] logger.c: -- Stopped music on hold on SIP/10.16.17.162-081bb720 [Sep 5 14:16:05] VERBOSE[6351] logger.c: -- Started music on hold, class 'default', on SIP/10.16.17.162-081bb720 start - stop - start strange, but it works ... If the same user/phone now presses hold/resume so that they could talk to each other again Asterisk does: [Sep 5 14:16:07] VERBOSE[6351] logger.c: -- Stopped music on hold on SIP/10.16.17.162-081bb720 [Sep 5 14:16:07] VERBOSE[6351] logger.c: -- Started music on hold, class 'default', on SIP/10.16.17.162-081bb720 stops the music and starts it again ... now the guy who pressed hold at first can hear the other party, but the other party only hears music from Asterisk. Has anyone had a similar phenomenon? Regards, Andreas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridge 2 incoming calls
Tim, you may want to try: 1) Park call 1 2) Pickup call 1 with call 2 (using ParkedCall) Regards, Andreas 2008/9/5 Tim Panton [EMAIL PROTECTED] I think I've forgotten something obvious I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital. I'm thinking of dialing chan_local with a call-id but I'm sure I am missing something simpler. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP AGI script
Nicholas Blasgen wrote: I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use normal AGI but I just need to catch a SIGTERM or something like that and process it. Nicholas, I solved that using the following extension: exten = h,1,DeadAGI(log_exit.php) If you catch any signals you're 'cheating' asterisk. Using the 'h' extension and DeadAGI should be fine. -Andreas ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmf detection not working on sip trunks using asterisk-1.4.15
Hi all, I am using an asterisk-1.4.13 connected to our carrier via SIP trunk. I use rfc2833 as dtmf detection method. After upgrading to asterisk-1.4.15 our system would not detect dtmf from a caller from PSTN anymore. When investigating the SIP traffic at call initiation I realized that in the SDP message asterisk is no longer offering the telephone-event/8000 capability. So the carrier does not send the rfc2833 messages anymore. Does anyone know about this or has seen an open bug case for it (I haven't found any myself)? Thanks for help and feedback. Kind Regards, Andreas Brodmann ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls being dropped
We have a similar problem at our place, since a few months. oej, mentioned a patch he has made after the release of asterisk-1.4.4. So we're all desperately waiting for asterisk-1.4.5 to be released; unless you want to install from svn. 2007/6/4, Compnet Bobby [EMAIL PROTECTED]: We have the latest version of asterisk, on a xeon dell server (2gb ram), with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest stable firmware) and are having a few problems. We have a basic menu that transfers calls to different extensions. The problems can be found on all extensions. We have 2 different incoming providers and the problem happens on both providers. I want your input on 2 problems, they are the following: 1. 60% of the time everything works fine and there are no problems, 40% of times when the calls are transferred to an extension, after a few seconds , the call drops. The log from the server is below(this is from pickup to hangup, the main area of concern is where it says warning). -- Executing [EMAIL PROTECTED]:1] Answer(SIP/9097406868-09e110f8, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/9097406868-09e110f8, menus/welcome-to-exec) in new stack -- SIP/9097406868-09e110f8 Playing 'menus/welcome-to-exec' (language 'en') == CDR updated on SIP/9097406868-09e110f8 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/9097406868-09e110f8, SIP/103|50|m) in new stack -- Called 103 -- Started music on hold, class 'default', on SIP/9097406868-09e110f8 -- SIP/103-09dedd68 is ringing [May 29 09:05:22] WARNING[2678]: chan_sip.c:1899 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1 (Critical Response) [May 29 09:05:22] WARNING[2678]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. -- Stopped music on hold on SIP/9097406868-09e110f8 == Spawn extension (from-sip, 103, 1) exited non-zero on 'SIP/9097406868-09e110f8' 2. When a call comes in or is transferred(not on outgoing), there is a delay until the person on the incoming line can hear you. We can hear them, but they can't hear us. Sometimes there is no delay, sometimes for person calling in cant hear you for 6 seconds. Thanks for the help in advance!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco CP-7970G
This is correct. To download firmware from cisco.com you need an account with the respective service agreement. When buying phones make sure you buy them with the respective firmware already present. AFAIK this agreement for a single phone is affordable though. Andreas 2007/5/25, [EMAIL PROTECTED] [EMAIL PROTECTED]: On Thu, May 24, 2007 6:27 pm, Anthony Francis [EMAIL PROTECTED] said: [EMAIL PROTECTED] wrote: Hi all, I just bought the 7970G phone. It's a beautiful phone. In trying to make it work with Asterisk, I've read many posts on the net. However, all of them make reference to having to install the SIP firmware on the phone. Where can I get it? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Download it from cisco www.cisco.com Just tried that. It seems that you need a Cisco Service Agreement before you can download it. Is that correct? Is that crazy? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rxgain/txgain in chan_sip
Hello All This or similar topics have already been mentioned but without any solution yet. I have built a oneway conference system for a client using one caller's input and broadcast it to all the other participants using app_meetme. E.g. one talker multiple listeners. Unfortunately some of the talkers (I have got multiple rooms) are not loud enough (e.g. use just half the amplitude, so making it louder by factor of 2.0would be necessary). My question: Is there a possibility in asterisk-1.4 to double/quadrouple the loudness of a channel's input/output using chan_sip? All clients come in via chan_sip so using another channel type is no solution in my situation. I use G711ulaw only. Btw: I would be very glad if someone could point me to a solution of this. Kind regards, Andreas Brodmann ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
