Displaying 20 results from an estimated 10000 matches similar to: "functional difference: canreinvite=yes, no, or update"
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,
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
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
2016 Aug 08
2
Asterisk & Vitelity Invite issues
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 up
sending a reinvite which their side & they do not support us sending a
reinvite. Ive tried:
canreinvite=no which was supposedly replaced by:
directmedia=no
Can anyone shed
2009 Jan 17
1
canreinvite per route
Can I activate/deactive the canreinvite SIP flag on the dial plan?
The idea is to allow reinvite only for exten <-> exten calls, and not for
outbound calls
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090117/a53f3178/attachment.htm
2007 Jan 17
2
AbsoluteTimeout with canreinvite=yes
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
2005 Jan 17
1
Media Path Optimization & NAT
(This message is not a dumb NAT question!)
I have a bunch of setups where an Asterisk system with a public IP
doubles as a router/gateway/firewall for a set of phones on a private
network.
We're using external SIP providers.
Everything works quite nicely.
Now, I would very much like to remove the "canreinvite=no" from the
provider's definition on sip.conf, but doing so
2014 Dec 11
6
T.38 not working - help needed with log interpretation
Hello,
at first, thanks for helping!
In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was:
exten => _00., 1, NoOp()
same => n, Set(FAXOPT(gateway)=yes)
same => n,
2007 May 03
2
SIP peer / Maximum retries exceeded on transmission
Hi Everyone,
I was hoping someone might know why I am experiencing a problem with
Asterisk logging the event:
[May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission 03f007af2b15cd0b54b0c368265d97be@sip.externalprovider.com for seqno 669371069 (Critical Response)
This is happening after:
- call is setup, 2 way audio
- call can function correctly for up to 5
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
2008 Oct 12
5
One Way Audio Problem
Hello all,
I've been lobbying for some time at the #asterisk IRC channel. Until
now, I still can't find a solution to my one way audio problem. I
rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my
Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS
(channel 1). My SIP extension phone located inside the LAN is a SNOM
300 IP phone.
This one way audio
2007 Feb 10
1
canreinvite problems
Hi,
I've been working on migrating my asterisk from zap to sip (due to
compatibility issues between my TDM400P and my Hauppauge PVR500). I've
purchased a Linksys SPA-3102 and a Siemens Gigaset SL75 WLAN (wireless SIP
phone). I managed to get it all working with my asterisk 1.4.0 installation,
but I'm seeing some interesting things with the canreinvite option that I
can't explain,
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?
2006 Jun 17
6
Canreinvite
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if
I call the traffic still go throw the asterisk. How come?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm
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
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 Jan 05
4
Broadvoice / * re-register issues
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =>
##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234
[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
2004 Dec 16
3
Get asterisk out of the RTP stream?
Here is the setup:
Phone A (in NYC) on own bandwidth.
Phone B (in LA) on own bandwidth.
Asterisk box in Houston,TX on own bandwidth.
Both phones contact asterisk to register. Not much bandwidth used for this
as it is a few packets every hour or so.
Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk
calls phone B. Both phones are connected and both people are talking.
2006 Dec 10
1
NAT and Dial to two channels at once
We all love Asterisk's ability to Dial(chan1&chan2) and take the first that
answers.
However, I have been encountering a problem when one of the channels
is an external phone behind NAT and another is a local phone on the
same net as the asterisk server.
All have canreinvite=yes, and the phone behind NAT is correctly
using Stun to give its external ports, which are opened to it
in the