similar to: AbsoluteTimeout with canreinvite=yes

Displaying 20 results from an estimated 10000 matches similar to: "AbsoluteTimeout with canreinvite=yes"

2006 Jan 23
3
canreinvite always =no * no matter what we try :-(
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?
2004 Aug 09
3
AbsoluteTimeout Inside A Macro
Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? [macro-attended] ; ARG1 is the device to dial out on, SIP or Zap, or whatever ; ARG2 is the extension to dial using 'attended' dialing exten => s,1,AbsoluteTimeout(30) exten =>
2009 Aug 27
2
Selective canreinvite in multi-tenant environment
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
2005 Jun 14
1
canreinvite=yes not working with sipura device.
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?
2008 Dec 18
1
canreinvite question
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
2004 Jan 08
5
AbsoluteTimeout Users Messages
Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ
2005 Feb 17
4
functional difference: canreinvite=yes, no, or update
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
2008 Dec 03
3
canreinvite=yes problem
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
2007 May 08
3
Responding to SIP OPTIONS
I understand that it is customary for SIP User Agents to send OPTIONS packets every now and then to check that a peer is still alive and well. Indeed I understand that Asterisk itself sends them if qualify is set to yes in the peer configuration. How is one supposed to configure the dialplan so that Asterisk responds correctly to these requests? At the moment, I'm seeing "Looking
2004 Sep 17
1
Canreinvite=???
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
2009 Nov 16
0
SIP Change canreinvite=yes/no from dialplan?
Hi All, Currently I have voice calls from a certain SIP peer coming into an asterisk server where the specific [SIP] channel is set to 'canreinvite=no'. I would like to enable reinvites for certain calls, matched on DID. So I'm wondering if there is a mechanism in the dial plan to turn on/off reinvite capability or will every call on this channel be forced to use the SIP peer
2005 Sep 19
3
T.38 & Canreinvite (yes, again)
I know this has been asked before, but I've checked the archives and I haven't found anybody that has given a definitive yes or no, just "yeah, it should work.....". If I have a T.38 gateway like a Cisco 5300 and a T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work? I have it setup and it doesn't work, so I want to know if I am doing something wrong,
2006 Jan 12
2
conditional canreinvite
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,
2019 Aug 15
4
PJSIP reInvite
Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is there any possibility to deactivate this kind of reInvite? We have some race conditions while have multiple asterisk in the call flow and the different
2016 Aug 10
2
Asterisk & Vitelity Invite issues
On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends
2007 Jun 08
0
Asterisk, NAT and canreinvite=yes
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
2005 Jun 20
0
Suggestions for using AbsoluteTimeout
I just discovered an 18 hour call to Brazil that was 60 seconds of an employee calling a customer, then 18 hours and 47 minutes of background noise in their office. The Cisco 7960's have an issue where you sometime don't realize the phone is still off hook as was the case for this call. I'd like to use some options like AbsoluteTimeout, which would jump in the middle of a call every
2009 Aug 26
1
netfilter conntrack mangling canreinvite?
Hello, all. Since implementing an iptables firewall between the Asterisk PBX and several SIP phones, the Asterisk PBX ability to "reinvite" has been broken even when the phones are on the same network (i.e., no firewall between the phones). We've been beating our heads against the wall thinking it was the complex rule set but it appears the issue is ip_conntrack_sip. Before I drop
2009 Feb 05
0
Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ? [SOLVED]
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
2009 Feb 05
0
Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ?
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: