similar to: SIP transfers failing

using CVS HEAD :) Some config snippets: extensions.conf: [from-ip] exten => 51,1,Dial(SIP/1301,20,t) exten => 52,1,Queue(ddi831,t) exten => 53,1,Queue(marketing,t) [internal] exten => _13XX,1,Dial(SIP/${EXTEN},20,Tt) queues.conf: [ddi831] strategy=roundrobin timeout=10 announce-frequency=0 announce-holdtime=no member => SIP/1301 [marketing] strategy=roundrobin timeout=10

> Date: Mon, 1 May 2000 16:25:11 +0200 (CEST) > From: gb <gb at stat.umu.se> > > On 1 May 2000, Douglas Bates wrote: > > gb <gb at stat.umu.se> writes: > > > > > How about 'Ainv <- qr.solve(A)'? > > > > > > I happened to read the help page for 'qr.solve' the other day, and there I > > > found that

Hi, I've been trying to implement this [1] on 1.2.13 and whilst my twiddlings seem to work, I just wanted confirmation that I'm not doing something really stupid which could cause a segfault under certain conditions. My chan_queue.c addition is this one line: ast_queue_log(queue, qe->chan->uniqueid, outgoing->chan->name, "RINGNOANSWER", "%d", orig);

Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and

Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective System's H323 Configuration example for tvcti ; ooh323c driver configuration ; ; [general]

Hello everyone, I'm on a small dovecot director -> dovecot mailbox setup and I try to get doveadm command proxying to work. Though I don't get the expected output. My directors do not forward the doveadm commands to the expected backend host. Doveadm is working as expected on the backend hosts, the director hosts just dont log anything about the proxying that should happen or

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1

I am successfully using dovecot purely as a personal local mail store on my desktop. (There is only one account, and it's only ever accessed by local mail clients on the machine. The point is to have a common store I can use with any client; plus, I prefer dovecot's Mailbox storage to Thunderbird's mboxes.) Now I'd like if possible, to replicate this setup on my laptop and

Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated

Why don't u attach the setup page of the phone ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di FaberK Inviato: gioved? 13 ottobre 2005 17.56 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] PA168S/AT320P Right now, but nothing changed. 2005/10/13,

For those of you who may be interest.... IAX2 loads are now available for the standard builds... http://www.aredfox.com/edownloadsiax2.htm Just a word of caution... Remember to change the ports over to 4569 from whatever. And don't forget to grab the palmtool from http://www.aredfox.com/download/tools/PalmTool.zip My own testing of IAX2 with both a phone and an ATA is that IAX2 is

Hi all, I am struggling to make my asterisk server work. The problem is I can not place a call from a phone outside, but I can call out from a phone in the local network where the asterisk server sits. I turn the debug on, and the log are shown below. I can see "REGISTER" method is OK. ( SIP/2.0 200 OK) But Later, in the "INVITE" method, the SIP addresses become

Sorry, i also tried this: exten => 6000,1,Answer exten => 6000,2,MusicOnHold(default) and i got this result: *CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack -- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class

This is my musiconhold.conf and my folder: root@voip:/etc/asterisk# less musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => mp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters

I have been hearing a lot about the new Gnet SIP phones. Is anyone using them? How do they perform? Sean

Hi, I installed mpg123 v0.59r without error and defined as defaut folder /var/lib/asterisk/mohmp3. When i set a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk 1.0.8 *CLI> -- Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new stack -- Called 2391 -- SIP/2391-79a0 is ringing -- Saved useragent "PA168S" for

Hi, I'm using asterisk 1.0.6 and I would let media path be connected directly between the phones without going through Asterisk. I have to it with an AtCom320 (with pa168s chip). I just saw and tryied to do what this page http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%2 0clients%20connect%20directly says. Before going on (with sniffer eth traffic between * and two

I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps someone on the list has experience with this. Is there a way to get MWI support for PA168V-based ATAs? Apparently some IP phones based on the PA168V chip has this support already (Atcom AT-320 for example) by configuring Asterisk with 'mailboxdetails=yes' in iax.conf. On my ATA, however, it does nothing. Any

Does anybody know why I can NOT hear the MusicOnHold - using SJphone on another PC in our network (normal playback is not a problem) . See the * output and the line configured in extension.conf below (also mp3player does not function) Any suggestions? *Asterisk output:* *CLI> -- Executing WaitMusicOnHold("SIP/jeroen-bf54", "30") in new stack --

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