similar to: canreinvite always =no * no matter what we try :-(

I''m looking to write a regular expression that will match valid URLs. My problem is that it almost works, except it accepts URLs with / in the middle of them, suchs as: http://www.ruby/rails.com It looks (to me) like my regular expression should not match strings like that, but it does. Here is the regular expression:

I'm trying to get canreinvite=yes to work. I would like asterisk to release the line and let the 2 ports on the sipura device to talk to each other directly. Is there a setting I need to activate on the sipura device, or is there something else I need to do? It's possible that it is a nat problem as the sip device is behind a firewall, but it works fine otherwise. Any suggestions?

Hi, I found this message concerning VAD and was wondering whether VAD has been ported to fixed-point in the latest version? Thanks, SingHui ---------- Forwarded message ---------- From: Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> Date: Jul 22, 2005 1:02 AM Subject: Re: [Speex-dev] Fixed-point To: gue baja <gue_baja@yahoo.com> Cc: speex-dev@xiph.org Hi Baja, Here's a quick

Is there any place where I can see a summary of what is being done and what is still pending with the fixed point version of the libraries? I have some experience with vocoders and fairly good experience with de-jitters and suchs. I may be able to help. Thanks Baja

Is AbsoluteTimeout designed to work with canreinvite=yes? If not, are the any other options for disconnecting a call after a predefined duration when using canreinvite=yes? Thanks! David

Is it possible to allow reinvites to/from specific devices? For example; exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004 exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002 Can that be done? Devices 2001 & 2002 are behind one firewall, and 2003 & 2004 are behind another. Tim

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

Hi Rowland, I've added 'dns update command' on global section of smb.conf file and I've configured namesever on '/etc/resolv.conf' as 127.0.0.1 (I've tried with 'kings' IP address too), but I don't know if this has worked. I've seen some dns updates errors on 'systemctl status samba-ad-dc' though the same command has returned status 'Active

Hi, everyone ! Looking at this explanation : "When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

Hi, list, I'm going to deploy postfix + dovecot + CephFS( as Mail Storage). Basically I want to use two servers for them, which is kind of HA. My idea is that using keepalived or Pacemaker to host a VIP, which could fail over the other server once one is down. And I'll use Haproxy or Nginx to schedule connections to one of those server based on source IP( Session stickiness),

Hello, Is there some version of FLAC that allows very very fast encoding (i.e. able to process at least 150 MB / second of .wav input data on a standard computer : laptop computer, Core i5/i7, Windows 7 64 bit, 8 GB RAM) ? (It's ok to have a compression ratio which is a little bit lower than traditionnal FLAC) I'm looking for something which is between FLAC (very good ratio, slower than

2009/2/5 Olivier <oza-4h07 at myamail.com> > Hi, > > Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a > table listing ATA/Gateways combinations. > Could anyone successfully set a Patton M-ATA to work with another one, > using Asterisk 1.4 ? > > Is reinvite (canreinvite=yes) necessary or not ? > > Regards > > Replying to myself, I

Hi, I can not get this working: Asterisk on public IP. Two SIP phones behind NAT - in the same LAN. I works perfectly (two way sound) when each peer (friend) can not reinvite - audio stream goes through Asterisk. The problem pops up when I define canreinvite=yes on each peer definision so I suppose to stream audio directly between phones (in the same local LAN). Right after called party

Howdy, Is it possible to send a reinvite after the media has connected? Scenario: Inbound call hits asterisk ivr then is sent out to an extension using the dial command. We have to carry the rtp streams in this case as asterisk cant send the reinvite after the ivr has stopped playing the message as we already connected the call. Question: Any way around this or is there a better way we can do

Hey Guys! I have a stupid question about canreinvite. We are using asterisk 1.8.3.2 as a PBX we don't have NAT or firewall thing in between asterisk and phone. so i should use conreinvite=no right ? what is the default value of conreinvite in asterisk 1.8.3.2 ? i meant yes or no ? -S -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a table listing ATA/Gateways combinations. Could anyone successfully set a Patton M-ATA to work with another one, using Asterisk 1.4 ? Is reinvite (canreinvite=yes) necessary or not ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

So I have many Cisco 7960's that are running the latest 5.1 Cisco SIP code and I cannot get the phones to talk/RTP to each other. jtodd has had this problem in the past with the 186's. Just wondering if anyone has a reason why "Cisco sometimes poop on reinvite" is the Cisco code broke? if so we can push on Cisco to fix it. the U is a MAJOR Cisco shop so we have some puhs

I've gotten canreinvite=yes to work with a sip device behind NAT!! You *MUST* port forward the SIPPort to in your gateway router to your phone. This is a MUST. Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the

Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the negotation between the 2 devices very well.. For example.. [gateway] type=friend host=1.2.3.4 canreinvite=yes qualify=200 dtmfmode=rfc2833 context=default disallow=all