[asterisk-users] Music on hold stops on blind transfer
Asterisk [EMAIL PROTECTED] Client hears pure silence when waiting for call answer. Music on hold stops when transferer pics a number and client doesn't even hear ringing. Is this normal behaviour? How to change this? Log says everything, MOH should stop after call pickup, not before Dial. -- Executing [EMAIL PROTECTED]:1] Dial(SIP/zytek-08737000, SIP/113|180|Tt) in new stack -- Called 113 -- SIP/113-08758000 is ringing -- SIP/113-08758000 answered SIP/zytek-08737000 -- Started music on hold, class 'default', on channel 'SIP/zytek-08737000' -- SIP/113-08758000 Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/zytek-08737000 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/zytek-08737000, SIP/zytek|180|Tt) in new stack -- Called zytek -- SIP/zytek-0875d000 is ringing PS. I am not receiving ANY mails from the list since subscribe (?) -- .: Jakub Głazik, .: email jabber: zytekatnuxi.pl ___ --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] Asterisk and Hardware Requirements
Hello, I would like to put 1 asterisk box in Country A and 1 asterisk box in Country B. Let's assume : - Asterisk box in country A = GWA - Asterisk box in country B = GWB - Calling party number (located in country A) = CgPNA - Called party number (located in country B) = CdPNB - Second Called party number (located in country B) = sCdPNB - PSTN in country A = PSTNA - PSTN in country B = PSTNB The idea is: The end user in country B will have 2 phone numbers : CdPNB and sCdPNB CgPNA is dialling CdPNB through PSTNA. The call will be intercepted by GWA. GWA will process the call, will change the CdPNB to sCdPNB and direct it via IP to GWB. If the quality of the call via IP is not really good, then I want to redirect it through E1 lines via KPN which will be able to route the call to GWB. GWB will then transmit the call through the PSTNB and will reach sCdPNB. First Scenario -- GWA's requirements would be: - 1 E1 port receiving the calls through PSTNA - 1 E1 port connected to KPN to re-route the calls - Asterisk? being able to map ~5000 CdPNB to sCdPNB - Handling up to ~50 simultaneous calls - Using G711, G.723.1 G.729.a codec - H323 and SIP compliant GWB's requirements would be: - 1 E1 port receiving the calls routed through KPN from GWA - 5 E1 port connected to PSTNB - Handling up to ~150 simultaneous calls - Using G711, G.723.1 G.729.a codec - H323 and SIP compliant - Asterisk managing the whole thing Second Scenario --- GWA's requirements would be: - 5 E1 port receiving the calls through PSTNA - 10 E1 port connected to KPN to re-route the calls - Asterisk? being able to map ~5 CdPNB to sCdPNB - Handling up to 500 simultaneous calls - Using G711, G.723.1 G.729.a codec - H323 and SIP compliant GWB's requirements would be: - 10 E1 port receiving the calls routed through KPN from GWA - 50 E1 port connected to PSTNB - Handling up to ~1500 simultaneous calls - Using G711, G.723.1 G.729.a codec - H323 and SIP compliant - Asterisk managing the whole thing I was wondering whether Asterisk could handle each scenario and what kind of hardware I would need to support each scenario. I hope I've been clear enough in my explanations. Looking forward for your comments/inputs. Regards, cam ___ --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] List delays
On Wed, 4 Jul 2007 17:37:29 +0200, Christian Victor wrote I have the same problem. My mail sent yesterday around 20:00h and it still not arrived at the list. Sent from germany by the way. Christian email delays here are about 8 days. I don't expect to see this until 19th July -- Bill Maidment Maidment Enterprises Pty Ltd www.maidment.vu ___ --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] registering Asterisk on SIP/Nortel MCS server
it used to be previous version of MCS, which was very easy to register with Ekiga recently ISP upgraded MCS to version 4 (I guess ?) and their proprietary SIP/unistim piece of software sends weird SIP/XML and other queries. I sniffered all of them, the most important I guess is SIP registration with XML things, can You help me how can I add those queries to asterisk ? On 7/4/07, Brian [EMAIL PROTECTED] wrote: Hi Kate, The Nortel MCS' SIP stack is a little special. You need to add special headers to the REGISTER messages in order to register with it. That means you'll need to touch Asterisk's source code. You can use wireshark to capture a successful REGISTER and see what headers you need. Regards, Brian Neotiq Consulting www.neotiq.com On 6/28/07, Kate Kretz [EMAIL PROTECTED] wrote: hello there... our telecom sold us VoIP-numbering, managed by Nortel MCS I successfully registered Ekiga to it ( http://sol.chel.skbkontur.ru/ekiga.png) What exactly do I have to write in sip.conf to make Asterisk register on this SIP ? Cheers, Kate ___ --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 ___ --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 ___ --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] how to force asterisk to read registration information from DB
In a redundancy scheme, I have two asterisk servers, each running on different machines although sharing the same MySQL DB for relatime sip. Problem arises when the second server assumes the production. When some phone tries to establish a new call, those INVITEs reach the new server, although this server seems to don't read the registration information kept in sip_buddies table to know if the destination phone is registered or not, and so, the call fails. Because the destination phone was registered in the first server, I was expecting that the second server when assuming production would first read the sip_buddies DB table to see if the destination phone was registered or not, but that seems to don't happen. It seems that registration information is only kept in memory and isn't read from DB! Is there any way that I can force Asterisk to read sip_buddies realtime DB table to know if destination phone is registered? Regards, Ricardo Carvalho. ___ --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] video calls - Windows / Linux interoperability ?
Hi 2007/7/11, Bruce Reeves [EMAIL PROTECTED]: For windows you can't really go wrong with X-lite 3.0, the free version of eye-beam. You should bear in mind, though, that Ekiga uses the H.261 video codec and X-lite uses H.263. Each of them only knows one video codec, so they are not interoperable, at least not at the time of writing. Asterisk does not do video translation. Regards, Bent ___ --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] List delays
On 7/11/07, Bill Maidment [EMAIL PROTECTED] wrote: email delays here are about 8 days. I don't expect to see this until 19th July When you do get the message, please reply with the email headers, so we have some chance of tracking down the problem. For example, below are the headers from your post as the message came to me. If my math is right, it shows about a sixteen minute delay from the time you send the message to the list to the time I got it. Without seeing the headers though, it's going to be difficult (if not impossible) for the Digium IT team to be able to see if the problem is on their end, or somewhere else in between. -Jared Received: from lists.digium.com (lists.digium.com [216.207.245.17]) by mx.google.com with ESMTP id 18si30015978agb.2007.07.11.03.14.08; Wed, 11 Jul 2007 03:14:09 -0700 (PDT) Received-SPF: pass (google.com: best guess record for domain of [EMAIL PROTECTED] designates 216.207.245.17 as permitted sender) Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I8Ywh-0006nM-CV; Wed, 11 Jul 2007 04:57:19 -0500 Received: from exprod8mx66.postini.com ([64.18.3.166] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I8Ywd-0006nG-8i for asterisk-users@lists.digium.com; Wed, 11 Jul 2007 04:57:15 -0500 Received: from source ([150.101.116.251]) by exprod8mx66.postini.com ([64.18.7.10]) with SMTP; Wed, 11 Jul 2007 02:57:13 PDT Received: from mail1.maidment.vu (mail1.maidment.vu [192.168.2.8]) by mail2.maidment.vu (8.14.1/8.14.1) with ESMTP id l6B9v8M8007038 for asterisk-users@lists.digium.com; Wed, 11 Jul 2007 19:57:08 +1000 Received: from mail1.maidment.vu (localhost.localdomain [127.0.0.1]) by mail1.maidment.vu (8.14.1/8.14.1) with ESMTP id l6B9v5km020073 for asterisk-users@lists.digium.com; Wed, 11 Jul 2007 19:57:05 +1000 ___ --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] Asterisk and Hardware Requirements
On 7/11/07, Josh [EMAIL PROTECTED] wrote: The idea is: The end user in country B will have 2 phone numbers : CdPNB and sCdPNB Wow... sounds like a fun setup! Your question is probably a little in-depth for the mailing list (as it depends on a lot of variables), but let's see what we can do to at least get you pointed in the right direction. First Scenario -- GWA's requirements would be: - 1 E1 port receiving the calls through PSTNA - 1 E1 port connected to KPN to re-route the calls - Asterisk? being able to map ~5000 CdPNB to sCdPNB - Handling up to ~50 simultaneous calls - Using G711, G.723.1 G.729.a codec - H323 and SIP compliant OK, first of all you'd need 2 E1 ports in GWA -- something like the Digium TE220P card if you've got a PCI Express slot, or the TE210P if you've got a 3.3v PCI slot, or the TE205P if you've got a 5v PCI slot. You'd also need a TC400B card in the box to be able to G.723.1 (and handle G.729A more efficiently). GWB's requirements would be: - 1 E1 port receiving the calls routed through KPN from GWA - 5 E1 port connected to PSTNB - Handling up to ~150 simultaneous calls - Using G711, G.723.1 G.729.a codec - H323 and SIP compliant - Asterisk managing the whole thing This would be very similar to the configuration for GWA, except that you'd need 6 E1 ports (which you could get with a combination of a 2-port and 4-port E1 card). You'd also want to make this a much beefier box, as handling 150 simultaneous calls might bring a weak server to its knees. Second Scenario --- GWA's requirements would be: - 5 E1 port receiving the calls through PSTNA - 10 E1 port connected to KPN to re-route the calls - Asterisk? being able to map ~5 CdPNB to sCdPNB - Handling up to 500 simultaneous calls - Using G711, G.723.1 G.729.a codec - H323 and SIP compliant Doing 500 simultaneous calls on a single Asterisk server is probably not the wisest thing to do. I'd suggest you spread the load across several Asterisk boxes (maybe across 4 boxes, each with a 4-port E1 card such as the Digium TE420P). GWB's requirements would be: - 10 E1 port receiving the calls routed through KPN from GWA - 50 E1 port connected to PSTNB - Handling up to ~1500 simultaneous calls - Using G711, G.723.1 G.729.a codec - H323 and SIP compliant - Asterisk managing the whole thing Again, connecting 60 E1s and handling 1500 simultaneous calls on a single Asterisk box is a recipe for disaster. You'll need to spread the load across many servers to be able to handle that kind of traffic. -Jared ___ --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] iax2 peer become UNREACHABLE
Hi, I have 2 Asterisk connected over a VPN with iax2 and has been working perfectly for the last two weeks, but now, one of the boxes has decided stop seeing the other, so now I can only place calls on one direction: bart*CLI iax2 show peers Name/UsernameHost Mask Port Status moe/bart 192.168.1.201 (S) 255.255.255.255 4569 (T) UNREACHABLE 1 iax2 peers [0 online, 1 offline, 0 unmonitored] moe*CLI iax2 show peers Name/UsernameHost Mask Port Status bart/moe 192.168.2.201 (S) 255.255.255.255 4569 (T) OK (43 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] This is my iax.conf file: [general] bindport=4569 bindaddr=0.0.0.0 language=es bandwidth=low disallow=all allow=gsm allow=lpc10 [moe] type=friend host=192.168.1.201 context=extensions username=bart secret=secret auth=plaintext peercontext=extensions qualify=yes trunk=yes The other: [general] bindport=4569 bindaddr=0.0.0.0 language=es bandwidth=low disallow=all allow=gsm allow=lpc10 [bart] type=friend host=192.168.2.201 context=extensions peercontext=extensions auth=plaintext username=moe secret=secret qualify=yes trunk=yes I have restarted both Asterisk with no luck, any clues to where should I search for the soluton? thanks, Jonathan. ___ --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] softphone with g729 codec
you can prove this www.portsip.com Gordon Henderson wrote: On Mon, 2 Jul 2007, jonny hashem wrote: Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? eyeBeam - the commercial version of X-Lite: http://www.counterpath.com/index.php?menu=Productssmenu=eyeBeam Gordon ___ --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 ___ --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] iax2 peer become UNREACHABLE
Why don´t you try to change this line: qualify=yes to this one: qualify=3000 or 4000 I think that the problem could be in the time that Asterisk uses to look at the another peer. Try it and tell us. MCelo. 2007/7/11, Jonathan Unai Marquez [EMAIL PROTECTED]: Hi, I have 2 Asterisk connected over a VPN with iax2 and has been working perfectly for the last two weeks, but now, one of the boxes has decided stop seeing the other, so now I can only place calls on one direction: bart*CLI iax2 show peers Name/UsernameHost Mask Port Status moe/bart 192.168.1.201 (S) 255.255.255.255 4569 (T) UNREACHABLE 1 iax2 peers [0 online, 1 offline, 0 unmonitored] moe*CLI iax2 show peers Name/UsernameHost Mask Port Status bart/moe 192.168.2.201 (S) 255.255.255.255 4569 (T) OK (43 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] This is my iax.conf file: [general] bindport=4569 bindaddr=0.0.0.0 language=es bandwidth=low disallow=all allow=gsm allow=lpc10 [moe] type=friend host=192.168.1.201 context=extensions username=bart secret=secret auth=plaintext peercontext=extensions qualify=yes trunk=yes The other: [general] bindport=4569 bindaddr=0.0.0.0 language=es bandwidth=low disallow=all allow=gsm allow=lpc10 [bart] type=friend host=192.168.2.201 context=extensions peercontext=extensions auth=plaintext username=moe secret=secret qualify=yes trunk=yes I have restarted both Asterisk with no luck, any clues to where should I search for the soluton? thanks, Jonathan. ___ --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 ___ --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] List delays
No, as I explained before with the reasons why, please don't post them here. Send them DIRECTLY to the list admins. It is 100% off topic to keep discussing a list administration / mail delivery problem here. List USERS can not help you. Considering that the vast majority of users do not experience such delays, and that it's HIGHLY unlikely that Digium maintains a list of who to delay mail for, the problem is 99% likely to be something wrong with the recipient's system. It could be DNS, routing problems, anti-spam mechanisms (greylisting, active sender verification, dspam, SA, etc.) or timeouts caused by slow responses due to said anti-spam mechanisms, etc. Many people fail to realize that high-volume mail servers (especially for large mailing lists) don't have long timeouts and therefore can't tolerate slow recipient servers. It takes too many resources. Make sure that you whitelist list mail at all phases of your protection systems. Make sure you are NOT doing sender callouts, running every message through spamassassin, greylistging, etc. for list mail. Lastly, there is nothing Digium is going to be able to do if your DNS servers are flakey, or route path is. Headers just tell you that there is a delay. We already know this. Only the sending AND receiving server logs can tell you WHY, and then you may only know if the session was run in debugging mode. On Wed, Jul 11, 2007 at 07:33:43AM -0400, Jared Smith said: On 7/11/07, Bill Maidment [EMAIL PROTECTED] wrote: email delays here are about 8 days. I don't expect to see this until 19th July When you do get the message, please reply with the email headers, so we have some chance of tracking down the problem. For example, below ___ --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] List delays
Walter, I know you just said that list users can not help with this problem, however I must beg to differ with you. The information that you just provided is a big help, if people take your advice about the configuration of their internal systems. So in one way it is off the topic of Asterisk, but is on target for helping people solve their issues if they interact more efficiently with the list and the resources that it provides. IMHO, Dave On Wed, 2007-07-11 at 08:47 -0400, Walt Reed wrote: No, as I explained before with the reasons why, please don't post them here. Send them DIRECTLY to the list admins. It is 100% off topic to keep discussing a list administration / mail delivery problem here. List USERS can not help you. Considering that the vast majority of users do not experience such delays, and that it's HIGHLY unlikely that Digium maintains a list of who to delay mail for, the problem is 99% likely to be something wrong with the recipient's system. It could be DNS, routing problems, anti-spam mechanisms (greylisting, active sender verification, dspam, SA, etc.) or timeouts caused by slow responses due to said anti-spam mechanisms, etc. Many people fail to realize that high-volume mail servers (especially for large mailing lists) don't have long timeouts and therefore can't tolerate slow recipient servers. It takes too many resources. Make sure that you whitelist list mail at all phases of your protection systems. Make sure you are NOT doing sender callouts, running every message through spamassassin, greylistging, etc. for list mail. Lastly, there is nothing Digium is going to be able to do if your DNS servers are flakey, or route path is. Headers just tell you that there is a delay. We already know this. Only the sending AND receiving server logs can tell you WHY, and then you may only know if the session was run in debugging mode. On Wed, Jul 11, 2007 at 07:33:43AM -0400, Jared Smith said: On 7/11/07, Bill Maidment [EMAIL PROTECTED] wrote: email delays here are about 8 days. I don't expect to see this until 19th July When you do get the message, please reply with the email headers, so we have some chance of tracking down the problem. For example, below ___ --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 ___ --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] VoIP + IM unified client
Dear all, I have a Debian/Asterisk server and I connect several softphones using SIP in a first test and IAX in a second test. They work OK in both cases; I use Twinkle client for SIP conversations and Kiax for IAX. But now I want to have IM also, I mean a voip client with a chat messenger incorporated, always using Asterisk. My questions are: 1) Do I have to add some module/package to my Asterisk in order to have IM ??? 2) What SIP+IM client do you recommend to me ??? 3) And what IAX+IM client do you recommend to me ??? Thanks in advance, Alejandro ___ --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] softphone with g729 codec
On Wed, 2007-07-11 at 08:32 -0400, Maximo Villamayor wrote: you can prove this www.portsip.com You can use the older version of firefly that supports IAX2/SIP protocols and g729 codec. Get the sofhophone and codec from: http://razametal.is-a-geek.org/asterisk/Softphones/Windows/firefly/firefly-thirdparty.exe http://razametal.is-a-geek.org/asterisk/Softphones/Windows/firefly/g729.zip To enable the g729: 1.- Install firefly-thirdparty.exe; 2.- close firefly program; 3.- extract g729.dll from g729.sip to c:/program files/firefly; 4.- start firefly, setup a new account and enable the g729 check box; Regards, Gordon Henderson wrote: On Mon, 2 Jul 2007, jonny hashem wrote: Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? eyeBeam - the commercial version of X-Lite: http://www.counterpath.com/index.php?menu=Productssmenu=eyeBeam Gordon ___ --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 ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] MOH stop and resume when i hold
Hi list, I have a strange comportment of the MOH system on my asterisk. When i respond to a call and after fews second i set this call in hold mode the correspondent listen the music fine. When i re-take my correspondent at T0 instant the music is paused. And when i re-hold him at T60 (60 second later) the sound is always at T0 when he was stopped at T0. So the music is stopped and don't continue (normal comportment) when i speak with peoples. Someone know the problem? a solution? Thanks ___ --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] Asterisk and IPv6
On 16:06, Mon 02 Jul 07, Bent Bagger wrote: Hi Russell Again, thanks for info. 2007/7/2, Russell Bryant [EMAIL PROTECTED]: I can assure you that once we have some plans in place, I will announce them. I'll wait for it, but I won't hold my breath ;-) $ svn co http://svn.digium.com/svn/asterisk/team/blanchet/v6 asterisk-ipv6 I downloaded, built and installed it without problems. However, when I run it, it aborts after about 8 seconds with a pthread error. Where would be the best place to ask for help - this list? the asterisk-devel list? or some specific person? Best you can do is report a bug on http://bugs.digium.com Read the guidelines before doing so ;) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --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] Monitor events?
Stefan Reuter wrote: Hey Daniel, I think adding the events would be a good idea. Just open an issue on http://bugs.digium.com/ and attach your patch there. Be sure to send a disclaimer to digium so your patch can be included in the distribution (see http://asterisk.org/developers/bug-guidelines for details). actually you probably know i am using your java-asterisk :) and of course if you already have patches for Asterisk-Java that support your new events post it to http://jira.reucon.org referencing the digium issue. =Stefan ___ --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 WTF ___ --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] Monitor events?
Anthony Francis wrote: Stefan Reuter wrote: Hey Daniel, I think adding the events would be a good idea. Just open an issue on http://bugs.digium.com/ and attach your patch there. Be sure to send a disclaimer to digium so your patch can be included in the distribution (see http://asterisk.org/developers/bug-guidelines for details). actually you probably know i am using your java-asterisk :) and of course if you already have patches for Asterisk-Java that support your new events post it to http://jira.reucon.org referencing the digium issue. =Stefan ___ --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 WTF ___ --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 Hi, I guess I should clarify. My name is Anthony, I was the one that said I had written a patch, if Daniel also said he had done so and I missed that email I apologize. Anthony ___ --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] Monitor events?
Dear Sir, On Wed, 11 Jul 2007, Anthony Francis wrote: WTF I am intrigued by your ideas and would like to subscribe to your quarterly newsletter. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Call Waiting
Since the beginning (of my Asterisk life) I have an install that is, supposedly, set up for call waiting. Using a TDM400p, with FXO and FXS modules. On the Analog phones, I can hear the Incoming call (call waiting) tone, but the system does not respond to a hook flash, to place the current call on hold and answer the incoming call. I have not attempted, nor research how/if this can be done on SIP. What am I not grasping here? About the Analog phone/Asterisk actions. Not too vague, I hope. joe a. ___ --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] Music on hold stops on blind transfer
Hi, Jakub: Jakub Głazik wrote: Asterisk [EMAIL PROTECTED] Client hears pure silence when waiting for call answer. Music on hold stops when transferer pics a number and client doesn't even hear ringing. Is this normal behaviour? How to change this? Sadly, this is normal behaviour. Log says everything, MOH should stop after call pickup, not before Dial. Well, no -- here, the log shows that MOH is stopping before Dial. -- Executing [EMAIL PROTECTED]:1] Dial(SIP/zytek-08737000, SIP/113|180|Tt) in new stack -- Called 113 -- SIP/113-08758000 is ringing -- SIP/113-08758000 answered SIP/zytek-08737000 -- Started music on hold, class 'default', on channel 'SIP/zytek-08737000' -- SIP/113-08758000 Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/zytek-08737000 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/zytek-08737000, SIP/zytek|180|Tt) in new stack -- Called zytek -- SIP/zytek-0875d000 is ringing PS. I am not receiving ANY mails from the list since subscribe (?) Yes, users outside North America seem to be getting messages very late. I'm told that Digium is aware of the problem. You might want to contact the list owner. Contact information can be found at: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Monitor events?
Anthony Francis wrote: I guess I should clarify. My name is Anthony, I was the one that said I had written a patch, if Daniel also said he had done so and I missed that email I apologize. Well then disclaim it and post it to the asterisk bug tracker or post its issue id if you already did. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ --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] List delays
Walt Reed wrote: No, as I explained before with the reasons why, please don't post them here. Send them DIRECTLY to the list admins. It is 100% off topic to keep discussing a list administration / mail delivery problem here. List USERS can not help you. Considering that the vast majority of users do not experience such delays, and that it's HIGHLY unlikely that Digium maintains a list of who to delay mail for, the problem is 99% likely to be something wrong with the recipient's system. It could be DNS, routing problems, anti-spam mechanisms (greylisting, active sender verification, dspam, SA, etc.) or timeouts caused by slow responses due to said anti-spam mechanisms, etc. I don't buy this. This problem did not appear until Digium changed list servers. If there's a DNS problem, it's probably with the list server configuration. -Stephen- PS: Did they name the Army Medical Center after you? ___ --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
