search for: trunktimestamps

...enario which seems to be showing a 1.2 box talking to a 1.4 box dropping calls sometimes, and the error message on the 1.2 box is showing that it does not like the unrecognised AST_CONTROL_SRCUPDATE frame that it receives. This issue may be exagerated by the fact that the Asterisk 1.2 box has "trunktimestamps=no" set to ensure compatability with an old Asterisk 1.0.x service. Help? Is there a workaround? Might it be enough to enable trunktimestamps in this instance? Thanks, Steve

Hi, this is the scenario: One * is placed in a central location with more than enough up/down bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via IAX trunking. Everything is fine until the upload channel of the remote site is filled with a download, then heavy voice distortion starts. Well of course this is expected. So I fooled around with HFSC QoS scheduling on the remote

Two asterisk, one in a 2.4.x and another one in a 2.6.x are connection ok using IAX, before i upgraded one of them (the one with 2.6.x) Trunk was working ok. Since i upgraded the one that its now 2.6.x, i m getting this message: chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?) ..and no sound comming from the

...rs, one running a recent HEAD, the other 1.0.9. I have both ends set up with trunk=yes, notransfer=yes, type=friend. I notice that the trunking works from HEAD to 1.0.9 only (the direction in which calls are originated). I know this by bandwidth usage and by iax2 trunk debug. I did have to use trunktimestamps=no on the HEAD end to keep it quiet. I assume this is the new jitterbuffer code. I know I should just upgrade the remote, but that option is difficult currently. Does anyone know why the "return" leg doesn't trunk? Mark

...- Average: 99.996930, Difference: 99.997334 <br> <br> ; iax.conf server_master <br> ; <br> [general] <br> bindport=XXXX <br> bindaddr=222.222.222.222 <br> delayreject=yes <br> language=fr <br> allow=alaw <br> maxjitterbuffer=800 <br> trunktimestamps=yes <br> tos=ef <br> <br> [jourdain] <br> type=friend <br> host=111.111.111.111 <br> port=XXXX <br> context=iax-jourdain <br> trunk=yes <br> disallow=all <br> allow=alaw <br> jitterbuffer=no <br> forcejitterbuffer=no <br>...

...new PLC/jitterbuffer stuff, but that doesn't seem to be it because I can transcode from ulaw -> g729 on the phone with my MOH extension with no problems. "sip show channels" shows that the channel is using g729, and the MOH plays for forever and ever. My iax.conf has trunk=no trunktimestamps=no and jitterbuffer=yes. If I set jitterbuffer=no, MOH works perfectly. So it appears to be the jitterbuffer in IAX that is causing problems with MOH there. I hope that this message wasn't too verbose, but I hope that any of these PLC/jitterbuffer problems can get ironed out, because it...

Hi Is anyone successfully using iax2 trunking with CVS head ? The reason I am asking is that I have heard there may be some audio problems, which I would like to know about before sending customer's calls over a iax2 trunked connection. Thanks in advance. Clive

I'm having some problems with a bunch of older IAXy devices. I have three S100I devices working. I can call each device from each other and everything works perfectly. When an inbound call comes in (from PSTN) and is routed to the device, it works perfectly as well. The problem is that when I place an outbound call (to PSTN), the remote party can hear me well, but I cannot hear the

Two asterisk boxes 150 miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only

...e. How to do to work correct. Or Asterisk has not yet have jitter and PLC ,hasn't it? In 'asterisk 1' , ================================================================== write on sip.conf => jbenable=yes ,and , jbimpl=adaptive write on iax.conf => jitterbuffer=yes ,and, trunktimestamps=yes write on codecs.conf => genericplc => true and on extensions.conf when use sip trunk => ------------------------------- exten => 3003,1,Dial(Local/3000 at extd/nj) exten => 3000,1,Set(CALLERID(num)="some id") exten => 3000,2,Dial(SIP/${EXTEN}@"some exten...

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On Tuesday 21 March 2006 11:19, Adam Robins wrote: > All switches and routers give