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
14:38:01.229941 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length:
889
14:38:01.230127 IP 192.168.4.248.sip > 192.168.4.240.sip: SIP, length:
515
14:38:01.251558 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length:
497
14:38:01.271714 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length:
106
Sent: Saturday, April 18, 2009 5:33 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Canreinvite=yes // native bridging // 2 sip
channels with different Call-ID
I have 2 SIP-clients defined in my sip.conf :
[GXP1200]
type=friend
context=intern
host=dynamic
username=GXP1200
secret
I have 2 SIP-clients defined in my sip.conf :
[GXP1200]
type=friend
context=intern
host=dynamic
username=GXP1200
secret=testpaswoord
canreinvite=yes
[BT201]
type=friend
context=intern
host=dynamic
username=BT201
secret=testpaswoord
canreinvite=yes
When I make a call from one to another this is d
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 messag
reinvite option and not send re-invites.
cheers
- Ben
--- On Sat, 1/17/09, Gabriel Ortiz Lour wrote:
> From: Gabriel Ortiz Lour
> Subject: [asterisk-users] canreinvite per route
> To: asterisk-users@lists.digium.com
> Date: Saturday, January 17, 2009, 10:06 PM
> Can I activate/dea
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
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asterisk-users mailing
mailto:asterisk-users-boun...@lists.digium.com] De la part de Tim Johnson
Envoyé : jeudi 18 décembre 2008 19:49
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] canreinvite question
Is it possible to allow reinvites to/from specific devices?
For example;
exten 2001 and 2002 can reinvit
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 ar
ot;ManxPower" Wieling
> Envoyé : mercredi 3 décembre 2008 19:25
> À : Asterisk Users Mailing List - Non-Commercial Discussion
> Objet : Re: [asterisk-users] canreinvite=yes problem
>
> canreinvite=yes should work as long as 1) there is no NAT involved
> anywhere in the call
nk you
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Eric
"ManxPower" Wieling
Envoyé : mercredi 3 décembre 2008 19:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] canreinvite=yes problem
canre
On 3 Dec 2008, at 17:38, BERGANZ François wrote:
> Someone have a solution for me ?
>
> De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> ] De la part de BERGANZ François
> Envoyé : mercredi 3 décembre 2008 18:24
> À : asterisk-users@lists.digium.com
> Objet : [asterisk-u
;ManxPower" Wieling
Envoyé : mercredi 3 décembre 2008 19:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] canreinvite=yes problem
canreinvite=yes should work as long as 1) there is no NAT involved
anywhere in the call path, 2) All legs of the call are
canreinvite=yes should work as long as 1) there is no NAT involved
anywhere in the call path, 2) All legs of the call are using the same
codec, 3) you do not have the t/T/w/W (and maybe a few other) options to
the Dial line.
Remember the only way you can really tell if a reinvite happens is by
On Wed, Dec 03, 2008 at 06:23:32PM +0100, BERGANZ François wrote:
> 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-outbridge.png
>
> But I have that http://www.zimagez
e* BERGANZ François
> *Envoyé :* mercredi 3 décembre 2008 18:24
> *À :* asterisk-users@lists.digium.com
> *Objet :* [asterisk-users] canreinvite=yes problem
>
>
>
>
>
> Hello,
>
>
>
> I need to test canreinvite=yes with 2softphones and 1 asterisk.
>
>
>
&
Someone have a solution for me ?
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de BERGANZ
François
Envoyé : mercredi 3 décembre 2008 18:24
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] canreinvite=yes problem
Hello,
I need to test canreinvite=yes with
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
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 ju
On Fri, 22 Feb 2008 18:50:16 +0800, Ron <[EMAIL PROTECTED]> wrote:
>If i set, canreinvite=yes on all ext, assuming all ip phones have the
>same codec, if 100 calls 101, or vice versa will rtp still go thru
>asterisk? and same scenario for 200 to 202 or vice versa.
... and I'd like to add to this
Hi All,
if i do this setup:
|---[ext 100]
|--[router/nat gw]--|
| |---[ext 101]
|
[asterisk]--[internet]---|
|
Hi list,
can anyone tell me how problematic it is setting canreinvite=yes ? I know if
its to avoid issues with bad implementatins of
SIP on other devices then maybe you cant give a black and white answer, but any
constructive comments welcome!
Reason being I think I have to set this to yes to
The others answered correctly personal I like using rtp debug.
As for making sure in the DialPlan that the RTP goes end to end
without asterisk.
1. Make sure they both use the same codec and protocol.
2. Don't put any options in app_dial, like tTwW or anything else that
will force asterisk to stay
> How can I know that the traffic went directly between
> the endpoints and did not go via the asterisk?
I'm sure there are many ways to do this
one way would be to do rtp debug on the cli and watch for media packets
another would be to do tcpdump on the command line and watch for packets
ther
Sent: Tuesday, September 11, 2007 10:14 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] canreinvite
Dear C F;
So in that case, if I placed canrenvite=yes for both endpoint, it is not
condition that traffic will be directly via the endpoint while signaling
via Asterisk as
Dear C F;
So in that case, if I placed canrenvite=yes for both
endpoint, it is not condition that traffic will be
directly via the endpoint while signaling via Asterisk
as still Asterisk should detect whethor it is
necessary to stay in the path or not? Please advise.
How can I know that the traffi
By default assuming you have no global setting otherwise, if asterisk
doesnt see a need to stay in the path then it wont. hence if it has to
transcode between different codecs, capture DTMF or different
protocols it will stay in the path.
On 9/9/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Hi Li
Hi List;
If I need traffic to be directly between the
endpoints, then I have to set the canreinvite = yes?
If I did not configure the canrenvite at all, then by
default it will pass the traffic via Asterisk and not
directly between the endpoints?
What if one endpoint was SIP and configured with
On 2/10/07, Luki <[EMAIL PROTECTED]> wrote:
Stefan,
> When I have 2 SIP endpoints that both aren't configured with
> "canreinvite=no" then I get no sound.
The Sipura 3102 definitely works fine with canreinvite=yes and I never
really had a problem with any of the Sipura devices in this respect,
Stefan,
When I have 2 SIP endpoints that both aren't configured with
"canreinvite=no" then I get no sound.
The Sipura 3102 definitely works fine with canreinvite=yes and I never
really had a problem with any of the Sipura devices in this respect,
especially when there is no NAT involved. Howev
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
- Original Message -
From: Gary Richardson
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 14:34:31 -0300
Subject: Re: [asterisk-users]
canreinvite=yes and RTP dropping in and out
> My next attempt
lists.digium.com]Sent:Wed, 02 Aug 2006 13:54:04 -0300Subject: [asterisk-users] canreinvite=yesand RTP dropping in and out> Hey guys,>> I'm having yet another strange problem. I've recently set canreinvite=yes,
> allowing the RTP streams to avoid our * server. Now, a few people
- Original Message -
From: Gary Richardson
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 13:54:04 -0300
Subject: [asterisk-users] canreinvite=yes
and RTP dropping in and out
> Hey guys,
>
Hey guys,I'm having yet another strange problem. I've recently set canreinvite=yes, allowing the RTP streams to avoid our * server. Now, a few people are experience one way audio drops on internal calls. External calls are working fine (they re-invite directly to a Cisco router). Sometimes, if you
Joshua Colp wrote:
We have a customer who would like to do RTP directly between SIP
devices. The devices are not registered directly to Asterisk, but to SER
on another machine.
It seems in this case "canreinvite = yes" is never used. Does anyone
know of a way of persuading Asterisk to issue r
- Original Message -
From: Alistair Cunningham
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Mon, 31 Jul 2006 07:10:43 -0300
Subject: [asterisk-users] Canreinvite and
remotely registered devices
> We have a custo
Patrick wrote:
It seems in this case "canreinvite = yes" is never used. Does anyone
know of a way of persuading Asterisk to issue re-invites in this case?
Although not clear from your posting I assume that the call between the
two phones is setup through the Asterisk server. Asterisk will no
On Mon, 2006-07-31 at 11:10 +0100, Alistair Cunningham wrote:
> We have a customer who would like to do RTP directly between SIP
> devices. The devices are not registered directly to Asterisk, but to SER
> on another machine.
>
> It seems in this case "canreinvite = yes" is never used. Does anyo
We have a customer who would like to do RTP directly between SIP
devices. The devices are not registered directly to Asterisk, but to SER
on another machine.
It seems in this case "canreinvite = yes" is never used. Does anyone
know of a way of persuading Asterisk to issue re-invites in this ca
erisk-users] Canreinvite
- Original Message -
From: Giordano Grandis
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Fri, 28 Jul 2006 07:01:08 -0300
Subject: [asterisk-users] Canreinvite
> How can I check if SIP re-i
- Original Message -
From: Giordano Grandis
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Fri, 28 Jul 2006 07:01:08 -0300
Subject: [asterisk-users] Canreinvite
> How can I check if SIP re-invite is really work
How can I check if
SIP re-invite is really working ?
I'm trying it with
two grandstream gxp2000.
Thanks
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es going on.
pFrom: "Il Neofita" <
[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
asterisk-users@lists.digium.com>Date: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsext
on" Date: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsexten => _40001,1,Dial(SIP/40001,30) exten =>
_40002,1,Dial(SIP/40002,30) From: "Il Neofita" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List -
cosa vedo a console -- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760
-- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI> sip show channelsPeer
This is the dial in extensionsexten => _40001,1,Dial(SIP/40001,30) exten => _40002,1,Dial(SIP/40002,30)
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What does your dial command look like?
On 6/17/06, Il Neofita <[EMAIL PROTECTED]> wrote:
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?
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Il Neofita wrote:
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?
Are you using the same codecs on the SPA3000 and the xlite? If no
then there's your reason.
--
Linux Home Automation Neil Cherry
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?
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Hi, folks.
If i use canreivite=no option in my sip.conf for users is this mean that
i need to load 729 and 723 codecs for thos UA that want to transmit it?
Or this is just traffic redirection feature? How this option reflect on
server load?
--
==
I just read:
Certain options to the Dial() statement require that Asterisk is in the
media path, and consequently Asterisk will not let go of it: /t/, ''T",
"h", "H", "w", "W" or "L" (with multiple arguments). Probably there are
more.
I had in my memory that "r", "R", "m" would also prevent
Hi
thanks, would mind pointing to me that
let me check and see
is that discussion will help me
ram
On 3/2/06, Paul Hales <[EMAIL PROTECTED]> wrote:
canreinvite = yes tells the phones to try and talk to each other andleave Asterisk out of the mix.
The important word here is TRY.There are lot
canreinvite = yes tells the phones to try and talk to each other and
leave Asterisk out of the mix.
The important word here is TRY.
There are lots of reasons that it might not quite work, and there was a
big discussion on the list about it a little while ago.
PaulH
On Thu, 2006-03-02 at 01:55
Hi all
iam working with * just started
can some one explain me canreinvite=yes when should i use the above options
I would like to use my * server for authentication and directly talk SIP user to SIP user
with out consuming my * bandwidth, is that correct
Does any one know, which provider s
> Actully ethereal
OK...
Try canreinvite=yes in the [general] section; this makes it the
default setting for all peers unless specified otherwise. Do the same
for nat=no in [general] to rule out all NAT'ing related issues. You
don't have tT in your Dial() statement, that's good. You say you
verifi
Hello and thanks for replying!
> Steve,
>
>> The mission is to actually get a reinvite to work on the lan.
> There isn't anything special to get this working... normally. I trust
> you verified the traffic flow with a network monitor tool (tcpdump?),
Actully ethereal,
It is encouraging to hear
On Mon, 23 Jan 2006, Steve Gladden wrote:
> 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 wha
> How are you testing if asterisk is in the media path?
Two ways:
One phone on a hub with ethereal on a laptop and watching the rtp
packets, pretty obvious that asterisk is staying in the media path.
and that the rtp i not coming from the other phone.
Way two, in the middle of an active/establi
Steve,
> The mission is to actually get a reinvite to work on the lan.
There isn't anything special to get this working... normally. I trust
you verified the traffic flow with a network monitor tool (tcpdump?),
correct? Does SIP debug give you any info (i.e., does it match the
right peer) -- you d
please turn on all the debug, warning, error etc messages in the
console, see logger.conf, then type sip peer debug and sip
peer debug to see the SIP messages.
How are you testing if asterisk is in the media path?
Regards
On 1/23/06, Steve Gladden <[EMAIL PROTECTED]> wrote:
> been testing with
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
Trond Andersen wrote:
Just one question. The documentation I have seen says that the RTP
audio stream is routed directly(if allowed ...), but never anything
about video streams? Is this just because documents are pre 1.2 or is it
true that audio can go directly, but video must pass through Aste
Hi,
Just one question. The documentation I have seen says that the RTP
audio stream is routed directly(if allowed ...), but never anything
about video streams? Is this just because documents are pre 1.2 or is it
true that audio can go directly, but video must pass through Asterisk?
Anyone?
Does
Am I reading the data below incorrectly, or does it appear that even
though I have the directive canreinvite=no set for the two asterisk
boxes, they are trying to do a reinvite (which fails) anyway?
Is this expected behaviour in this situation? If so, how can I prevent
this?
Lots of output
Has anyone made this work? For me everything is fine until I switch
canreinvite form no to yes. What happens is that asterisk hangs on
"attempting native bridge" ... from what I understand "attempting native
bridge" means that the RTP is routed through asterisk (just without any
codec translatio
Hi,
I’m using asterisk 1.0.6 and I would let media path be connected directly between the
phones without going through Asterisk. I have to it with an AtCom320 (with
pa168s chip).
I just saw and tryied to
do what this page http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20
> I'm trying to get canreinvite=yes to work.
As the name says, this setting allows reinvites but does not force
them. I just ran into the same issue last week. Here the caveats:
Reinvites will only happen when both ends use the same codes, there is
no t or T option in the dial command when making
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 proble
On Apr 7, 2005 8:36 PM, kaiser <[EMAIL PROTECTED]> wrote:
> Hi , all:
> Anyone try sip channel with canreinvite=yes?
>
> sometimes we see a new INVITE will be send to UA immediately after user
> hangup the call.
> It makes the phone ring again after hangup.
> Anyone know what happen?
> It not alwa
Hi , all:
Anyone try sip channel with canreinvite=yes?
sometimes we see a new INVITE will be send to UA immediately after user
hangup the call.
It makes the phone ring again after hangup.
Anyone know what happen?
It not always, maybe 2-5% only.
But it make user crazy.
Thanks...
_
The KEY thing you are missing is that IAX does NOT use RTP for audio.
IAX uses IAX for audio and IAX for signaling. You CANNOT "reinvite"
between a SIP/RTP endpoint and an IAX endpoint.
Since SIP, H323, MGCP, and SCCP/Skinny all use RTP for audio, in theory
you could do RTP reinvites between the
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
Based on my post yesterday, and the call trace I have, if Asterisk were
to make a decision a little differently when sending the the ReINVITEs
to phone B in your example (lets say Phone A is the one behind NAT)
media might work both directions. In the trace I posted, asterisk first
send a reinvite
Senad Jordanovic wrote:
brian wrote:
That's the only way to make it work.
Devices behind nat, on same network, can call each other ONLY if
"canreinvite" is set to no? Is that what you are saying?
Canreinvite=yes *only* works if all devices are on the same side of the NAT, the
outside or the inside.
Does anyone know if it is possible to force a extension to not allow
transcoding? If you spec canreinvite=yes the cal may still transcoded if the
parties do not choose a the same code on each end. In my situation it is
better that the call fail than have it transcoded.
Also, I see some limited ref
Hi Zen,
From: Zen Kato <[EMAIL PROTECTED]>
Does these "t" and "T" are used for transfer(blind/consaltation) from
called user and calling user, respectively? If we don't have these
't' and 'T', can't we do transfer?
'T' and 't' are used for transfer using #
The 'T' allows the calling user to tran
t and T are for "#" transfers. Other types of transfer are done in
other ways. Zap FLASH transfers are set in /etc/asterisk/zapata.conf.
I don't know how you enable/disable SIP or other types of transfers.
On Thu, 2004-03-04 at 06:51, Zen Kato wrote:
> Hi,
>
> Thank you for the information. Th
Hi,
Thank you for the information. There are "t"s in Dial command in
extensions.conf. When I deleted these "t"s, each sip phones were
directly communicating. I just wrote these "t"s from the examples.
Does these "t" and "T" are used for transfer(blind/consaltation) from
called user and calling u
Zen,
I am trying to confirm the command 'canreinvite=yes' in sip.conf
using grandstream BT101/2s and snom100s. In either case, no description
nor 'canreinvite=yes', media stream always go through *.
Do I need another settings for confirming sip clients directly
communicate each other?
Do you have
Hi,
I am trying to confirm the command 'canreinvite=yes' in sip.conf
using grandstream BT101/2s and snom100s. In either case, no description
nor 'canreinvite=yes', media stream always go through *.
Do I need another settings for confirming sip clients directly
communicate each other?
--
Zen
__
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=
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