similar to: SIP Change canreinvite=yes/no from dialplan?

Hi All, I haven't started sip traces or debug yet, but was wondering what the deal is with the CCM and reinvite, why it doesn't work with Asterisk (using 1.2.9.1). I can make calls back and forth all day with canreinvite=no, when I try to reinvite an inbound sip call from the CCM with Asterisk server 1 to Asterisk Server 2, I get one-way audio issues. All the RTP ports are configured

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

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

I have a number of internal SIP phones and a number of external SIP clients with the server running Asterisk on the boundary between the two. ie the server has two network cards with an internal private address and an external public address. For security reasons no routing is allowed between the two thus no internal phone can talk directly to external phone or visa versa. I am very happy with

is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes? i'm trying to route calls to a gateway without keeping asterisk in the rtp stream. the gateway is first routing the call to a media server. when connecting the call to the downstream carrier a different codec is selected. the reinvite makes it to asterisk but asterisk isn't

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, 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'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?

been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan?

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

Can anyone give an example of the difference between the following: canreinvite=no canreinvite=yes canreinvite=update Here is the problem: I have an 800 number sent to me via SIP from a national carrier. Asterisk gets the number and rings my desk phone. Asterisk has 2 NICs, one with public IP and private IP. My phone is on private IP, the inbound call is on public. My phone rings and I answer

Hi, Sorry for this "newb" question (but maybe a newb question once in a while is ok): What's the current state about Asterisk handling DTMF features via SIP INFO (dtmfmode=info) even when the media path has been reinvited (canreinvite=yes) to go directly from one phone to another? Somewhat related to this suspended issue: https://issues.asterisk.org/view.php?id=14126 How widely

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

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, I have asterisk on public IP and phones in two locations behind firewall/nat, - when I have nat=yes and canreinvite=no, this is working fine, but rtp stream must go _always_ through asterisk, even if phones talk inside their locations - when I have nat=yes and canreinvite=yes, phones can speak only inside their location and rtp stream is connected directly between phones (this is, imho,

Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you -------------- next part -------------- An HTML

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

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

I've the following situation: Phone A: Codec GSM supported Phone B: Codec iLBC supported in sip.conf: [general] ... disallow=all allow=gsm allow=ilbc allow=alaw allow=ulaw canreinvite=yes ... (There's a lot of other SIP users, that's why I made the default codec list bigger than just GSM and/or ALAW) If phone A calls to phone B the conversation is established at SIP level, but