similar to: Confusion on call forwarding

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like "promiscredir=yes" in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. Thanks in advance, Thiago Abra sua conta no Yahoo!

I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem?

HI, is there are official MIME type for Vorbis.... audio/x-vorbis ? Dave --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions?

> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.

Hi, when I'm trying to call the following extension exten => 6002,1,Verbose(1|Extension 6002) exten => 6002,n,Dial(Agent/6002) exten => 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem

> Done, and it's working... but not significantly better than before. > Whereas before I couldn't clear a single song without the stream devolving > into skipping, now I'm averaging between 15-20 minutes. Better, but still > not acceptable. THen I suggest you start to look elsewhere for your problem. libshout is quite well tested and every time someone has thought it was

I'm trying to address a problem with users transferring to invalid destinations. In my sip peer I'm setting both __FORWARD_CONTEXT and __TRANSFER_CONTEXT to a context with a extension defined below to set some CDR variables before the call is transferred. [customer-forward] exten => _X.,1,Progress() exten => _X.,n,Gosub(do-billing,s,1${EXTEN})) exten =>

It's kind of low for me. How does one control that volume?

Which speed recognition products will also recognize DTMF? In other words, I want to say "Please speak or dial the conference number". Does Vestec allow that? LumenVox? Any other way?

Signed-off-by: Goldwyn Rodrigues <rgoldwyn at suse.com> --- fs/ocfs2/stack_user.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/fs/ocfs2/stack_user.c b/fs/ocfs2/stack_user.c index 286edf1..1b18193 100644 --- a/fs/ocfs2/stack_user.c +++ b/fs/ocfs2/stack_user.c @@ -799,11 +799,31 @@ static int fs_protocol_compare(struct

Hi All, I have a question about how a particular situation would work between two PBX systems: If I were to connect my Mitel box to my Asterisk box via a SIP trunk (same rack, same network), and then pass a call from the Mitel to Asterisk to perform some functions (lookups, maybe routing), and then pass the call back to the Mitel to be routed to it's endpoint, would Asterisk stay in that

I need to include another helper module apart from the normal two (ApplicationHelper and [controllername]Helper). The inclusion needs to be dynamic and based on external parameters (ie what helper that get included differ from request to request). Is it possible? How? /Marcus

I recently tried to update my Aastra 57i to version 3.2 and ran into a problem. It won't properly register and says "contact mismatch". I added "sip contact matching: 2" to aastra.cfg, but that didn't help. When I look at the SIP trace, but I see is the Aastra sending a REGISTER and Asterisk replying with the 401. The phone then sends the REGISTER again, this time

Interesting. Are you using the included cbend.php script to terminate conferences? I occasionally get questions about using WMM with Confbridge, and to date I have not had an answer . If you can provide details, even vague ones, about how you did it, I can update the WMM package. Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at

I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject it. It supports G.729, and that works fine, but we'd prefer not to use compression. When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

How does one do this? We have a particular SIP phone that needs a large jitterbuffer, but all I can see is how to put it on the *read* side of the channel.