[Asterisk-Users] Canreinvite=???

[Carlos Arnt]

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 media streams 
directly to each other."

So if i really understand this using this option i can make the RTP packets 
flow from one device to another when they connect leaving only the SIP to 
asterisk .
So for example if then I put my Grandstream with a real ip address and use 
* with a real ip address i can make my calls from nufone flow direct to my 
grandstream leaving my * bandwidth free .

Like this :
Grandstream begin call SIP---> Asterisk |
   | -> 
Nufone.

Open RTP Channel
Grandstream Real IP <--> Nufone IP
Right ??
If i'm right , i try this and with tcpdump see the even with everyone using 
real ip's, the RTP still going over asterisk using my bandwidth .
(Note, I force grandstream to use the same codec then Nufone, G729)

Can someone give-me some light ?? ;)
Can i make this ??? Use asterisk only to begin the call and let the RTP 
flow over the client and nufone network ??

Thanks alot !
Carlos.
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[asterisk-users] canreinvite

[bilal ghayyad]

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
canreinvite=yes while other endpoint was IAX2 and
configured with canreinvite=yes, then they can send
traffic to each other directly or it will be via
Asterisk?

Regards
Bilal


  

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[Asterisk-Users] Canreinvite

[Il Neofita]

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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[asterisk-users] Canreinvite

[Giordano Grandis]

How can I check if 
SIP re-invite is really working ?
 
I'm trying it with 
two grandstream gxp2000.
 
Thanks
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[Asterisk-Users] Canreinvite issue

[kaiser]

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...

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Re: [Asterisk-Users] Canreinvite=???

[Eric Wieling]

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 these protocols.  I have no idea if
Asterisk supports this or not.

On Sat, 2004-09-18 at 00:52, Carlos Arnt wrote:
> 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 media streams 
> directly to each other."
> 
> So if i really understand this using this option i can make the RTP packets 
> flow from one device to another when they connect leaving only the SIP to 
> asterisk .
> So for example if then I put my Grandstream with a real ip address and use 
> * with a real ip address i can make my calls from nufone flow direct to my 
> grandstream leaving my * bandwidth free .
> 
> Like this :
> 
> Grandstream begin call SIP---> Asterisk |
> | -> 
> Nufone.
> 
> Open RTP Channel
> 
> Grandstream Real IP <--> Nufone IP
> 
> Right ??
> If i'm right , i try this and with tcpdump see the even with everyone using 
> real ip's, the RTP still going over asterisk using my bandwidth .
> (Note, I force grandstream to use the same codec then Nufone, G729)
> 
> Can someone give-me some light ?? ;)
> Can i make this ??? Use asterisk only to begin the call and let the RTP 
> flow over the client and nufone network ??
> 
> Thanks alot !
> 
> Carlos.
> 
> 
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upgrades can NOT be deducted as a gambling loss."

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[asterisk-users] canreinvite question

[Ron]

Hi All,


if i do this setup:

  |---[ext 100]
  |--[router/nat gw]--|
  |   |---[ext 101]
  |
[asterisk]--[internet]---|
  |
  |   |---[ext 200]
  |--[router/nat gw]--|
  |---[ext 201]


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.

what if 100 call 200 or 201? or 200 calls 100 or 100? will rtp still go 
thru asterisk?

thank you

regards,
Ron


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Re: [asterisk-users] canreinvite

[C F]

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 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
> canreinvite=yes while other endpoint was IAX2 and
> configured with canreinvite=yes, then they can send
> traffic to each other directly or it will be via
> Asterisk?
>
> Regards
> Bilal
>
>
>
> 
> Luggage? GPS? Comic books?
> Check out fitting gifts for grads at Yahoo! Search
> http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz
>
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Re: [asterisk-users] canreinvite

[bilal ghayyad]

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 traffic went directly between
the endpoints and did not go via the asterisk?

Regards
Bilal Ghayad
Mobile: 009659849460


-
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 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
> canreinvite=yes while other endpoint was IAX2 and
> configured with canreinvite=yes, then they can send
> traffic to each other directly or it will be via
> Asterisk?
>
> Regards
> Bilal


  

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Re: [asterisk-users] canreinvite

[Wai Wu]

Don't know about IAX. As for SIP, You will know what ip address and port
the audios should be transmitted to by looking at the sdp session. Just
goto the * console and enable sip debug.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal
ghayyad
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 still Asterisk should detect whethor it is necessary to
stay in the path or not? Please advise.

How can I know that the traffic went directly between the endpoints and
did not go via the asterisk?

Regards
Bilal Ghayad
Mobile: 009659849460


-
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 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 canreinvite=yes while

> other endpoint was IAX2 and configured with canreinvite=yes, then they

> can send traffic to each other directly or it will be via Asterisk?
>
> Regards
> Bilal


 


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Re: [asterisk-users] canreinvite

[mail-lists]

> 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 
there.



> 
> Regards
> Bilal Ghayad
> Mobile: 009659849460
> 
> 
> -
> 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 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
>> canreinvite=yes while other endpoint was IAX2 and
>> configured with canreinvite=yes, then they can send
>> traffic to each other directly or it will be via
>> Asterisk?
>>
>> Regards
>> Bilal
> 
> 
>   
> 
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> http://autos.yahoo.com/new_cars.html
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Re: [asterisk-users] canreinvite

[C F]

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 in the stream to listen for DTMF.

On 9/11/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> 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 traffic went directly between
> the endpoints and did not go via the asterisk?
>
> Regards
> Bilal Ghayad
> Mobile: 009659849460
>
>
> -
> 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 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
> > canreinvite=yes while other endpoint was IAX2 and
> > configured with canreinvite=yes, then they can send
> > traffic to each other directly or it will be via
> > Asterisk?
> >
> > Regards
> > Bilal
>
>
>   
> 
> Check out the hottest 2008 models today at Yahoo! Autos.
> http://autos.yahoo.com/new_cars.html
>

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[asterisk-users] canreinvite question

[Tim Johnson]

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


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[asterisk-users] canreinvite problems

[Stefan van der Eijk]

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, even after reading:
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite

My setup:

  - asterisk server with:
 - eth0 = 192.168.254.254 (internal network)
 - eth1 = Internet IP-address
 - ZAP/1 (FXO) not used
 - ZAP/2 (FXS) not used
 - ZAP/3 and ZAP/4 (FXS) with DECT phones
 - SPA-3102:
  - WAN interface configured with DHCP, it gets
192.168.254.104(internal network)
 - LAN interface is not being used
 - Line1: DECT phone
 - PSTN: is connected to the PSTN
 - Siemens SL75 WLAN: 192.168.254.105
  - Laptop (192.168.254.125) with an Eyebeam and idefisk softphone

All the SIP endpoints are connected to the internal network, there should be
no NAT issues.

In all situations I'm able to dial the other phone and make it ring.


From the ZAP endpoints to the SIP endpoints (and vice versa) I get sound.

Same applies to the IAX2 client (idefisk).

When I have 2 SIP endpoints that both aren't configured with
"canreinvite=no" then I get no sound.

Conclusion: all media needs to go through the asterisk server in order to
get sound.

Questions:

  1. Are all of my SIP endpoints incompatible with the canreinvite=yes
  option?
  2. Is there a list of SIP endpoints that are known to work with
  "canreinvite=yes"?
  3. Are other people also experiencing this?

with kind regards,

Stefan van der Eijk
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[Asterisk-Users] canreinvite=yes

[Trond Andersen]

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 anyone have experience with H263 on the 1.2.rc1 version? I think
there is a bug, and will trace and submit it to Bugzilla..??


Trond

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Re: [Asterisk-Users] Canreinvite

[Neil Cherry]

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.

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[Asterisk-Users] canreinvite=yes

[ram]

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 support this option
 
ram
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Re: [Asterisk-Users] Canreinvite

[C F]

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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Re: [Asterisk-Users] Canreinvite

[Il Neofita]

This is the dial in extensionsexten => _40001,1,Dial(SIP/40001,30)    exten => _40002,1,Dial(SIP/40002,30)    
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Re: [Asterisk-Users] Canreinvite

[Il Neofita]

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 User/ANR    Call ID  Seq (Tx/Rx)  Form  Hold Last Message82.X2.XX3.X3 40002   146b518a4cd  00103/0  alaw  No   Tx: ACK
82.X2.XX3.X3 40001   CBD1DB85-8B  00102/30987  alaw  No   Tx: ACK2 active SIP channels
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Re: [Asterisk-Users] Canreinvite

[Philippe Lindheimer]

How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd  on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages going on.pFrom: "Il Neofita" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion" 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 - Non-Commercial Discussion" Date: Sun, 18 Jun 2006 05:22:35 -0400Subject: Re: [Asterisk-Users] Canreinvite 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 User/ANR    Call ID  Seq (Tx/Rx)  Form  Hold Last Message82.X2.XX3.X3
 40002   146b518a4cd  00103/0  alaw  No   Tx: ACK 82.X2.XX3.X3 40001   CBD1DB85-8B  00102/30987  alaw  No   Tx: ACK2 active SIP channels 
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Re: [Asterisk-Users] Canreinvite

[Il Neofita]

I will try your suggestion and I will let you know. Thank you On 6/18/06, Philippe Lindheimer <[EMAIL PROTECTED]
> wrote:How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd  on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages 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 extensionsexten => _40001,1,Dial(SIP/40001,30)    
exten =>
 _40002,1,Dial(SIP/40002,30)     From: "Il Neofita" <[EMAIL PROTECTED]
>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com
>Date: Sun, 18 Jun 2006 05:22:35 -0400Subject: Re: [Asterisk-Users] Canreinvite 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 channels
Peer User/ANR    Call ID  Seq (Tx/Rx)  Form  Hold Last Message82.X2.XX3.X3
 40002   146b518a4cd  00103/0  alaw  No   Tx: ACK 82.X2.XX3.X3 40001   CBD1DB85-8B  00102/30987  alaw  No   Tx: ACK2 active SIP channels
 
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Re: [asterisk-users] Canreinvite

[Joshua Colp]

- 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 working ?

If you do a sip debug you should see two INVITEs to each side after the call is 
established with the IP address of the GXP2000 in the SDP. You can also run rtp 
debug to see if the RTP audio stream is running through Asterisk.

> I'm trying it with two grandstream gxp2000.
>  
> Thanks
> 

Joshua Colp
Digium
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R: [asterisk-users] Canreinvite

[Giordano Grandis]

Ok, thanks, also if i do not have rtp debug (i'm using asterisk 1.0.9)

Hi

-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Joshua Colp
Inviato: venerdì 28 luglio 2006 12.54
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-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-invite is really working ?

If you do a sip debug you should see two INVITEs to each side after the call is 
established with the IP address of the GXP2000 in the SDP. You can also run rtp 
debug to see if the RTP audio stream is running through Asterisk.

> I'm trying it with two grandstream gxp2000.
>  
> Thanks
> 

Joshua Colp
Digium
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[Asterisk-Users] canreinvite and transcoding

[Glenn Dalgliesh]

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 reference to canreinvite=update. Does this command
exist and if so what does it do.

Thanks

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Re: [Asterisk-Users] Canreinvite issue

[snacktime]

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 always, maybe 2-5% only.
> But it make user crazy.
> 
> Thanks...

So that's what causes that.  Had it happen a few times with my Sipura 2000.  

Chris
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Re: [asterisk-users] canreinvite question

[Vincent]

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 question: If the phones have the
option "Enable NAT", I expected them to be able to talk to each other
directly, but they didn't, and I had to set them to "canreinvite=no"
in sip.conf. Why is that?


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[asterisk-users] canreinvite=yes problem

[BERGANZ François]

 

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

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[asterisk-users] canreinvite=yes -->problems

[BERGANZ François]

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

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Re: [asterisk-users] canreinvite question

[BERGANZ François]

In the sip.conf


[2001]
...
Canreinvite=yes

[2002]
...
Canreinvite=no


Cordialement,
BERGANZ François


http://www.acropolistelecom.net
 Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

-Message d'origine-
De : asterisk-users-boun...@lists.digium.com 
[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 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


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Re: [asterisk-users] canreinvite problems

[Luki]

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. However, the default "Auto
NetService Private IP Ranges:" includes 192.168.0.0-192.168.255.255,
so your 192.168.254.0/24 network would be considered a LAN address by
the 3102 and hence the traffic would go out the LAN interface (not
WAN). Change this setting by removing this range. It's on the Admin >
Advanced > LAN Setup tab.

If that doesn't help, then you need to check what traffic is being
sent. Since all devices are on the same internal network I assume they
can see each other. You need to look at the Invite (and ReInvite)
messages sent and received and see if the IP addresses for RTP listed
there make sense. Then I suggest you use tcpdump to see what traffic
is sent by each device, and where. If you have a switched network
environment this will be a bit tricky as your * box won't see this
traffic, so you may want to use a hub for this test (just temporarily)
or if available set up port mirroring to sniff the traffic.

Good luck and keep us posted.

--Luki
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Re: [asterisk-users] canreinvite problems

[Stefan van der Eijk]

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,
especially when there is no NAT involved. However, the default "Auto
NetService Private IP Ranges:" includes 192.168.0.0-192.168.255.255,
so your 192.168.254.0/24 network would be considered a LAN address by
the 3102 and hence the traffic would go out the LAN interface (not
WAN). Change this setting by removing this range. It's on the Admin >
Advanced > LAN Setup tab.

If that doesn't help, then you need to check what traffic is being
sent. Since all devices are on the same internal network I assume they
can see each other. You need to look at the Invite (and ReInvite)
messages sent and received and see if the IP addresses for RTP listed
there make sense. Then I suggest you use tcpdump to see what traffic
is sent by each device, and where. If you have a switched network
environment this will be a bit tricky as your * box won't see this
traffic, so you may want to use a hub for this test (just temporarily)
or if available set up port mirroring to sniff the traffic.

Good luck and keep us posted.



Luki,

I just configured the wlan phone and my eyebeam endpoints with
canreinvite=yes (which should put the sipura out of the picture). Calling
the wlan phone from eyebeam: no sound gets through. Putting a canreinvite=no
in either one of the configurations (for the wlan01 or the eyebeam) forces
the media through the asterisk and sound gets through.

I'll get wireshark running on my laptop so I can post the SIP conversations
here.

regards,

Stefan
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[asterisk-users] canreinvite per route

[Gabriel Ortiz Lour]

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] canreinvite in sip.conf

[Giordano Grandis]

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%20SIP%20clients%20connect%20directly
says.

Before going on (with
sniffer eth traffic between * and two phones) I’d like to known if it can
works. Does anyone just did it?

 

Thanks in advance

 

Gio

 






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Re: [Asterisk-Users] canreinvite=yes

[Kevin P. Fleming]

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 Asterisk?


All RTP streams are handled identically, regardless of their content.
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Re: [Asterisk-Users] canreinvite=yes

[Paul Hales]

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 +0530, ram wrote:
> 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 support this option
>  
> ram
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Re: [Asterisk-Users] canreinvite=yes

[ram]

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 lots of reasons that it might not quite work, and there was abig discussion on the list about it a little while ago.PaulHOn Thu, 2006-03-02 at 01:55 +0530, ram wrote:
> 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 support this option>> ram> ___
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[Asterisk-Users] canreinvite and codec negotations...

[Billy Huddleston]

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
allow=ulaw
allow=g729

[123]
type=friend
secret=abc
nat=yes
host=dynamic
canreinvite=yes
qualify=200
context=default
dtmfmode=rfc2833
mailbox=2199
callerid="Joe Blow" <123-456-7890>
disallow=all
allow=g729

[321]
type=friend
secret=abc
nat=yes
host=dynamic
canreinvite=yes
qualify=200
context=default
dtmfmode=rfc2833
mailbox=321
callerid="Joe Blow" <321-456-7890>
disallow=all
allow=ulaw


Okay, in this configs, gateway would be my cisco 26xx gateway..  ext 123
would be a g729 customer.. and 321 would be a ulaw customer.  When someone
calls ext 123, the cisco send the call to asterisk, it *WILL* use ulaw...
then it will initiate a call to g729, well... Now we have a codec mismatch,
and canreinvite won't work... EVEN though gateway can do g729.. ext 321
won't have any problems.. It'll work fine for them..  What can we do to get
this to work like it should?

Thank, Billy


 +--+
 | Billy HuddlestonSenior System Administrator  |
 | Net-Express  http://www.nxs.net  |
 | 114 Sherway Rd. Voice: 865-691-2011  |
 | Knoxville, TN  37922  Fax: 865-691-9894  |
 | [EMAIL PROTECTED]|
 +--+

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Re: [asterisk-users] canreinvite=yes problem

[BERGANZ François]

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 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

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Re: [asterisk-users] canreinvite=yes problem

[Carlos Rojas]

Hello,

canreinvite, don't work with all softphone or hardphone.


Regards

On Wed, Dec 3, 2008 at 12:38 PM, BERGANZ François <
[EMAIL PROTECTED]> 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-users] 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-outbridge.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
>
> ___
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Re: [asterisk-users] canreinvite=yes problem

[Robert Lister]

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.com/zimage/canreinvite.php
>  
> 
> Canreinvite=yes work for all phones or just asterisk?...

I believe canreinvite=yes is the default option unless you set it
to canreinvite=no

I would leave it set to yes unless there is some reason to change it, 
for example the phone is behind NAT, or transfers etc don't work 
correctly without it being set to no.

If it's still not doing the right thing, then it's worth also
checking the nat= option

There are also other settings which can cause asterisk to stay in the media 
path, as BOTH sip devices need canreinvite=yes, otherwise it will stay in 
the media path. Specifying certain options on the Dial() cmd may also cause 
it to stay in the media path.

Rob


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Re: [asterisk-users] canreinvite=yes problem

[Eric "ManxPower" Wieling]

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 
looking at the RTP audio.  The SIP signaling will not and has never had 
a "reinvite" feature for signaling.

Why did you post the same message at :23, :28, and :35 mins past the 
hour?  If you need immediate support you should contact Digium support 
and pay for a service contract.


BERGANZ François wrote:
> 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?...

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html


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Re: [asterisk-users] canreinvite=yes problem

[BERGANZ François]

Now, I have :

Client 1
-Asterisk1--Asterisk2
Client 2

I need that sip sign go to Asterisk2
But RTP go to Asterisk1 and no more.

Where have I to insert canreinvite ?

Thank 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

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 
looking at the RTP audio.  The SIP signaling will not and has never had 
a "reinvite" feature for signaling.

Why did you post the same message at :23, :28, and :35 mins past the 
hour?  If you need immediate support you should contact Digium support 
and pay for a service contract.


BERGANZ François wrote:
> 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?...

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html


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Re: [asterisk-users] canreinvite=yes problem

[Steve Howes]

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-users] 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-outbridge.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

Yes.

1. POST ONCE
2. If no one replies within 20 mins, don't start chasing
3. If its that important pay for support
4. Read documentation


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Re: [asterisk-users] canreinvite=yes problem

[BERGANZ François]

I still have:
Client 1
-Asterisk1--Asterisk2
Client 2


When client1 do a call, asterisk1 forward to asterisk2, asterisk2 forward to
Asterisk1
At this moment, asterisk1 say : 404Not found
But I have insecure=very

  






This is the sip debug at that moment:





<->
--- (11 headers 0 lines) ---

<--- SIP read from UDP://192.168.1.151:5060 --->
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2b4a242e;rport
Max-Forwards: 70
From: "103" ;tag=as636875d3
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Thu, 04 Dec 2008 14:55:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 1545198644 1545198644 IN IP4 192.168.1.151
s=Asterisk PBX 1.6.0.1
c=IN IP4 192.168.1.151
t=0 0
m=audio 12272 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<->
--- (14 headers 13 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 192.168.1.151 : 5060 (NAT)
Using INVITE request as basis request -
[EMAIL PROTECTED]
No user '103' in SIP users list
Found peer 'media' for '103' from 192.168.1.151:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.151:12272
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.151:12272
Looking for 33170725012 in media (domain 192.168.1.153)

<--- Reliably Transmitting (no NAT) to 192.168.1.151:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.1.151:5060;branch=z9hG4bK2b4a242e;received=192.168.1.151;rport=5060
From: "103" ;tag=as636875d3
To: ;tag=as242de969
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0






Have you an idea why ?





































-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de BERGANZ
François
Envoyé : jeudi 4 décembre 2008 09:15
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : Re: [asterisk-users] canreinvite=yes problem

Now, I have :

Client 1
-Asterisk1--Asterisk2
Client 2

I need that sip sign go to Asterisk2
But RTP go to Asterisk1 and no more.

Where have I to insert canreinvite ?

Thank 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

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 
looking at the RTP audio.  The SIP signaling will not and has never had 
a "reinvite" feature for signaling.

Why did you post the same message at :23, :28, and :35 mins past the 
hour?  If you need immediate support you should contact Digium support 
and pay for a service contract.


BERGANZ François wrote:
> 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?...

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html


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Re: [asterisk-users] canreinvite=yes problem

[Eric "ManxPower" Wieling]

Reinvites will happen by default.  Post your sip.conf [general] and the 
peers in sip.conf masking only the passwords.  Also paste the part of 
extensions.conf that you use to Dial.

BERGANZ François wrote:
> Now, I have :
> 
> Client 1
> -Asterisk1--Asterisk2
> Client 2
> 
> I need that sip sign go to Asterisk2
> But RTP go to Asterisk1 and no more.
> 
> Where have I to insert canreinvite ?
> 
> Thank 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
> 
> 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 
> looking at the RTP audio.  The SIP signaling will not and has never had 
> a "reinvite" feature for signaling.
> 
> Why did you post the same message at :23, :28, and :35 mins past the 
> hour?  If you need immediate support you should contact Digium support 
> and pay for a service contract.
> 
> 
> BERGANZ François wrote:
>> 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?...
> 

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html


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Re: [asterisk-users] canreinvite per route

[Benjamin Jacob]

Have canreinvite set for your internal extens.

You can also have canreinvite enabled by default for all and use one or more of 
the 't','T','h','H','w','W' or 'L' options set in your dial commands which will 
override the canreinvite 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/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] Canreinvite after media connection

[carl Lougher]

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 it?

Cheers,
Taff



  

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[Asterisk-Users] canreinvite = yes with PAP2

[Tomas Florian]

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 translation)  But it shouldn't do that ... right? ... canreinvite is
set to yes ...

What's the best way to deal with this issue?  I've also read that the only
way to get the following situation ...

UA --- NAT --- Internet --- NAT --- UA 

... to work without passing the media path through asterisk is to use SER
together with asterisk.  Is that still true or was that because I was
reading stuff from back in 2003? 

Some other discussions mention that canreinvite will simply not work with
certain UAs .. is PAP2 one of those?

.. Couple of other discussions that I've seen conclude that passing media
stream UA-to-UA is just not practical when NAT is involved and is best to be
avoided all together ... I'd like to make it work because it seems like a
great way to save expensive server bandwidth.  But if it will cause more
trouble than it's worth then I will probably pass the media path through
Asterisk and live with the fact that it will eat up my bandwidth.

Also, IAX is superior when dealing with NATs , does it also handle UA-to-UA
in NATed environment smoothly?  What would be a good PAP2 alternative that
uses IAX?
 
This is my sip.conf:

[1001]
username=1001
type=friend
secret=
qualify=yes
port=5060
nat=yes
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=yes
callerid="Test1" <1001>

... My PAP2 is configured with:

STUN=yes
STUN=stun.xten.net
NAT Keepalive = 15
Outbound proxy = blank
Proxy = IP of asterisk

Any suggestions?

Thank you,
Tomas



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[Asterisk-Users] canreinvite=no being ignored?

[Chris A. Icide]

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  

Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A)
has a sip ua (2608) attached which is generating a call, the other
machine (box B) has the final destination.

Sip config for the phone on box A (via Realtime):

pbx3*CLI> sip show peer 2608

 

  * Name   : 2608
  Secret   : 
  MD5Secret: 
  Context  : assigned-device
  Language :
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : -1
  Inc. limit   : 0
  Outg. limit  : 0
  Dynamic  : Yes
  Callerid : "" <>
  Expire   : 386
  Expiry   : 900
  Insecure : no
  Nat  : RFC3581
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr->IP : 192.168.10.32 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 2608
  SIP Options  : (none)
  Codecs   : 0x4 (ulaw)
  Codec Order  : (ulaw)
  Status   : OK (16 ms)
  Useragent: Sipura/SPA841-0.9.1
  Reg. Contact : sip:[EMAIL PROTECTED]:5060


Sip config for Box B on box A:

pbx3*CLI> sip show peer pbx1

 

  * Name   : boxb
  Secret   : 
  MD5Secret: 
  Context  : inter-system-inbound-main
  Language :
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : -1
  Inc. limit   : 0
  Outg. limit  : 0
  Dynamic  : No
  Callerid : "" <>
  Expire   : -1
  Expiry   : 900
  Insecure : no
  Nat  : RFC3581
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   : 
  Addr->IP :  Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username:
  SIP Options  : (none)
  Codecs   : 0x4 (ulaw)
  Codec Order  : (ulaw)
  Status   : Unmonitored
  Useragent:
  Reg. Contact :


Sip config for Box A on Box B

pbx1*CLI> sip show peer pbx3
pbx1*CLI>

 

  * Name   : boxa
  Secret   : 
  MD5Secret: 
  Context  : inter-system-inbound-main
  Language :
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : -1
  Inc. limit   : 0
  Outg. limit  : 0
  Dynamic  : No
  Callerid : "" <>
  Expire   : -1
  Expiry   : 900
  Insecure : no
  Nat  : No
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   : 
  Addr->IP :  Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username:
  SIP Options  : (none)
  Codecs   : 0x4 (ulaw)
  Codec Order  : (ulaw)
  Status   : Unmonitored
  Useragent:
  Reg. Contact :


Dial command as appears on boxa

-- Executing Dial("SIP/2608-8049", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
Aug 31 02:01:29 NOTICE[10496]: chan_sip.c:9028 handle_response: Failed
to authenticate on INVITE to '"2608"
>;tag=as4124f74a'
-- SIP/boxb-ae96 is circuit-busy

SIP Debug as it appears on boxb from the call above

<-- SIP read from :5060:
INVITE sip:c1#1234@ SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK1d216175;rport
From: "2608" ;tag=as4124f74a
To: 
Contact: 
Call-ID: 3f1250096c1a12b0259689006888f106@
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 09:01:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 214

 

v=0
=root 10496 10496 IN IP4 
s=session
c=IN IP4 
t=0 0
m=audio 19014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 

--- (12 headers 10 lines)---
Using INVITE request as basis request -
3f1250096c1a12b0259689006888f106@
Sending to  : 5060 (non-NAT)
Reliably Transmitting (NAT) to :5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
:5060;branch=z9hG4bK1d216175;received=;rport=5060
From: "2608" >;tag=as4124f74a
To: >;tag=as2ac1a098
Call-ID: 3f1250096c1a12b0259689006888f106@
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK

[Asterisk-Users] canreinvite, bandwidth, dial option

[Ronald Wiplinger]

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 a reinvite. Can 
anybody say something on that? Below is a list of all options.


 o *t*: Allow the /called/ user to transfer the call by hitting #
 o *T*: Allow the /calling/ user to transfer the call by hitting #
 o *r*: Generate a ringing tone for the calling party, passing
   no audio from the called channel(s) until one answers. Use
   with care and don't insert this by default into all your
   dial statements as you are killing call progress information
   for the user. Really, you almost certainly do not want to
   use this. Asterisk will generate ring tones automatically
   where it is appropriate to do so. "r" makes it go the next
   step and additionally generate ring tones where it is
   probably not appropriate to do so.
 o *R*: Indicate ringing to the calling party when the called
   party indicates ringing, pass no audio until answered. This
   is available only if you are using kapejod's bristuff
   .
 o *m*: Provide Music on Hold to the calling party until the
   called channel answers. This is mutually exclusive with
   option 'r', obviously. Use m(class) to specify a class for
   the music on hold.
 o *n*: (Asterisk 1.1 and later) July 2005 bug 752
    was included in CVS
   (Asterisk 1.1) and enhances the privacy manager
   considerably. As part of this patch, the 'n' flag to Dial
   got changed to be used as part of the privacy features,
   instead of being the 'dont jump to +101' flag. That flag is
   now 'j'.
 o *o*: Restore the Asterisk v1.0 CallerId behaviour (send the
   original caller's ID) in Asterisk v1.2 (default: send this
   extension's number)
 o *j*: Asterisk 1.2 and later: Jump to priority n+101 if all
   of the requested channels were busy (just like behaviour in
   Asterisk 1.0.x)
 o *M(*/x/*)*: Executes the macro (x) upon connect of the call
   (i.e. when the called party answers)
 o *h*: Allow the callee to hang up by dialing ***
 o *H*: Allow the caller to hang up by dialing ***
 o *C*: Reset the CDR (Call Detail Record) for this call. This
   is like using the NoCDR
   
   command
 o *P(*/x/*)*: Use the PrivacyManager
   
,
   using /x/ as the database (/x/ is optional)
 o *g*: When the called party hangs up, exit to execute more
   commands in the current context.
 o *G(context^exten^pri)*: If the call is answered, transfer
   both parties to the specified context and extension. The
   calling party is transferred to priority x, and the called
   party to priority x+1. This allows the dialplan to
   distinguish between the calling and called legs of the call
   (new in v1.2).
 o *A(*/x/*)*: Play an announcement (/x/.gsm) to the called party.
 o *S(*/n/*)*: Hangup the call /n/ seconds AFTER called party
   picks up.
 o *d*: This flag trumps the 'H' flag and intercepts any dtmf
   while waiting for the call to be answered and returns that
   value on the spot. This allows you to dial a 1-digit exit
   extension while waiting for the call to be answered - see
   also RetryDial
   

 o *D(*/digits/*)*: After the called party answers, send
   /digits/ as a DTMF stream, then connect the call to the
   originating channel.
 o *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y'
   ms are left, repeated every 'z' ms) Only 'x' is required,
   'y' and 'z' are optional. The following special variables
   are optional for limit calls: (pasted from app_dial.c)
   + *LIMIT_PLAYAUDIO_CALLER* - yes|no (default yes) - Play
 sounds to the caller.
   + *LIMIT_PLAYAUDIO_CALLEE* - yes|no - Play sounds to the
 callee.
   + *LIMIT_TIMEOUT_FILE* - File to play when time is up.
   + *LIMIT_CONNECT_FILE* - File to play when call begins.
   + *LIMIT_WARNING_FILE* - File to play as warning if 'y'
 is defined. If *LIMIT_WARNING_FILE* is not defined,
  

[Asterisk-Users] canreinvite=no and codecs.

[Nikolay Pavlov]

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?


-- 

= Best regards, Nikolay Pavlov. <<<--- =

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[asterisk-users] canreinvite option - gona have problems?

[Andy Smith]

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 enable mediaproxy RTP proxy 
on my OpenSER box to interoperate correctly
with Asterisk,


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[asterisk-users] Canreinvite and remotely registered devices

[Alistair Cunningham]

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 case?


--
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Re: [asterisk-users] Canreinvite and remotely registered devices

[Patrick]

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 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 not let
go if you have ie the "T" or "t" option in your Dial statement. Remove
those for starters. If Asterisk is not involved at all I guess you need
to find out what the equivalent of "canreinvite=yes" is in SER country.

Regards,
Patrick

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Re: [asterisk-users] Canreinvite and remotely registered devices

[Alistair Cunningham]

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 not let
go if you have ie the "T" or "t" option in your Dial statement. Remove
those for starters. If Asterisk is not involved at all I guess you need
to find out what the equivalent of "canreinvite=yes" is in SER country.


Patrick,

Yes, Asterisk is handling the call setup for billing purposes. There is 
no t or T in the dial.


SER does not have an equivalent of canreinvite as it is a SIP proxy not 
an end point.


Alistair Cunningham,
Integrics Ltd,
+44 20 799 39 799
http://integrics.com/
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Re: [asterisk-users] Canreinvite and remotely registered devices

[Joshua Colp]

- 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 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 case?

What do you mean by not used? Even if going through SER it should still be used.
 
> -- 
> Alistair Cunningham,
> Integrics Ltd,
> +44 20 799 39 799
> http://integrics.com/

Joshua Colp
Digium
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Re: [asterisk-users] Canreinvite and remotely registered devices

[Alistair Cunningham]

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 re-invites in this case?


What do you mean by not used? Even if going through SER it should still be used.


Joshua,

That's what I would have expected, but Asterisk is not issuing re-invites.

One thing I should have mentioned is that Asterisk, SER, and the phones 
are all on an RFC1918 network (but there is no NAT between them). Maybe 
Asterisk is seeing that:


1. The phones are offering an SDP address that is in RFC1918.

2. The SDP and SIP headers are not the address the SIP is coming from, 
as SER is relaying them.


and therefore incorrectly concluding that there's NAT between it and the 
phones?


Alistair Cunningham,
Integrics Ltd,
+44 20 799 39 799
http://integrics.com/
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[asterisk-users] canreinvite yes or no for PBX

[satish patel]

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 

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[Asterisk-Users] canreinvite=yes not working with sipura device.

[Jon Gabrielson]

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?


Thanks,


Jon.



p.s.  it's a sipura3000, but it should be the same for any sip device.
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[Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk

[Zen Kato]

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
 
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Re: [Asterisk-Users] canreinvite=yes not working with sipura device.

[Luki]

> 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 the call, and I think
when both ends are behind the same NAT or not NAT'ed... but don't
quote me on the last one. For the Sipura 3000 the last one is not an
issue since it's the same physical device for both ports.

Try sip debug and look for line like "Found Peer" or "Found User" on
an incoming call to make sure the right one is used, unless you have
canreinvite turned on globally.

--Luki
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[Asterisk-Users] canreinvite always =no * no matter what we try :-(

[Steve Gladden]

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?

Any pointers greatly appreciated!

Steve


Pretty simple extensions, on lan no nat


[4785]

type=friend
username=4785
secret=test
host=dynamic
canreinvite=yes

[4786]

type=friend
username=4786
secret=tesst
host=dynamic
canreinvite=yes


exten => 4785,1,Dial(SIP/4785,66)
exten => 4785,3,hangup

exten => 4786,1,Dial(SIP/4786,66)
exten => 4786,3,hangup

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[asterisk-users] canreinvite=yes and RTP dropping in and out

[Gary Richardson]

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 wait 20 seconds or more, the stream will resume. Flipping the person on and off hold won't resume the stream.
We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem to happen all of the time. There are no sip messages being exchanged when the stream stops or restarts.Any suggestions?Thanks.

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RE: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk

[Girish Gopinath]

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 a Dial statement that has "t" or "T" in it?
This will force the media stream to pass through Asterisk.
Regards, Girish

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Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk

[Zen Kato]

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 user, respectively? If we don't have these
't' and 'T', can't we do transfer?

Regards,

Zen

"Girish Gopinath" <[EMAIL PROTECTED]> wrote  :

> 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 a Dial statement that has "t" or "T" in it?
> This will force the media stream to pass through Asterisk.
> 
> Regards, Girish
> 
> _
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Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk

[Eric Wieling]

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. 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 user, respectively? If we don't have these
> 't' and 'T', can't we do transfer?
> 
> Regards,
> 
> Zen
> 
> "Girish Gopinath" <[EMAIL PROTECTED]> wrote  :
> 
> > 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 a Dial statement that has "t" or "T" in it?
> > This will force the media stream to pass through Asterisk.
> > 
> > Regards, Girish
> > 
> > _
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"Unofficial Links" section also see
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Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk

[Girish Gopinath]

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 transfer the call.
't' allows the called user to transfer the call.
Andy Powell's guide to Asterisk http://www.automated.it/guidetoasterisk.htm 
has these details, It is simple, and contains some good basic things about 
Asterisk.

Regards, Girish

Regards,

Zen

"Girish Gopinath" <[EMAIL PROTECTED]> wrote  :

> 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 a Dial statement that has "t" or "T" in it?
> This will force the media stream to pass through Asterisk.
>
> Regards, Girish
>
> _
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[Asterisk-Users] Re: [Asterisk-Users Canreinvite=[yes|no] explained (new subject)

[Olle E. Johansson]

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.
If one device is on a different side of a NAT device than another phone
or Asterisk, you can't allow re-invites for that device.
Let me explain:
Phone A -> Asterisk -> phone B
Phone A calls phone B. The first thing you have to remember is that Asterisk
is *not* a SIP proxy, it's a SIP PBX. So phone A sends an invite to Asterisk.
Asterisk starts a *new* SIP dialog and sends an invite to Phone B.
If Phone B accepts the call ("answers"), it sends an "200 OK" sip message
to Asterisk. Asterisk sends a "200 OK" SIP message to phone A. We now have
*two* SIP calls, or dialogs, that are bridged through Asterisk.
The natural behaviour for the Asterisk SIP channel in this case is
to get a feeling of "Hey, these two phones are using me just for transport
of bits between themselves. Why not get out of the loop and let them
exchange these bits directly with each other? I can use the spare CPU
cycles for something more meaningful than shipping these bits..."
So, in order to get out of this meaningless position, Asterisk checks if
the phones are compatible. If they support the same codec, if they
can talk to each other. If *one* of them have canreinvite=no
or something else that stops a direct audio relationship from phone A to B,
Asterisk stays in the middle of things, shipping bits between the phones
(the audio stream).
If they are compatible, Asterisk sends a new SIP INVITE to both of the
phones, redirecting the media streams for the current calls to each
other. This is called a Re-INVITE, an INVITE where we change media
for an existing SIP dialog, not trying to start a new call.
This works perfectly well if all phones are compatible and we have
the network configured so that they can talk full duplex to each other.
If one is behind a NAT, they can't. If that's the case, we'll only
get one audio stream, from the phone on the inside to the one on the
outside. That's why we want to set canreinvite=no to make sure
Asterisk doesn't even try to re-INVITE.
--
Executive Summary:
canreinvite=no in a [peer] configuration makes Asterisk stay
in the middle of the media stream. This is the "safe" setting,
but remember it makes your CPU always handle the media stream.
Regards,
/Olle
PS. Yes, the Asteriskdocs.org project is allowed to use this text if they
find it meaningful :-)
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Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(

[Moises Silva]

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 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?
>
> Any pointers greatly appreciated!
>
> Steve
>
>
> Pretty simple extensions, on lan no nat
>
> 
> [4785]
>
> type=friend
> username=4785
> secret=test
> host=dynamic
> canreinvite=yes
>
> [4786]
>
> type=friend
> username=4786
> secret=tesst
> host=dynamic
> canreinvite=yes
>
> 
> exten => 4785,1,Dial(SIP/4785,66)
> exten => 4785,3,hangup
>
> exten => 4786,1,Dial(SIP/4786,66)
> exten => 4786,3,hangup
>
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Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(

[Luki]

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 don't show if you allow reinvites globally? What
about the nat= setting?

Couple pointers I can give you to get you excited:
1) Reinvites work quite reliably, I use them between the PTSN gateway
and the end user's ATA, all the way across the Internet -- nicely
reduces latency.

2) If you use RFC2833 for DTMF you can issue an reinvite and still use
t/T for transfer. NOTE that you have to modify the source to make
asterisk reinvite even when it needs to listen to DTMFs. I give no
guarantees how well it will work for you but it does work.

See "AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1" in rtp.c.

3) Reinvites *can* work even if both ends are behind NAT. It really
depends on the NATing router and the ATA. Sipura's and good NAT
routers work, but I would not call it "reliable" -- it's really
pushing it a bit...

So if you really want to see why your Reinvites do not work, then you
probably will have to make your hands dirty and analyze where
ast_rtp_bridge() in rtp.c bails out. But since you are on a LAN it
makes the situation a lot easier.

--Luki
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Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(

[Steve Gladden]

> 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/established call unplugging the
ethernet cable from the asterisk box
audio instantly dies on both phones when this occurs.
plug asterisk box back into it's ethernet termination
audio comes right back.

Seems odd that these reinvites are supposed to magically occur
(from what I gather) and it only happens when the sun is shining
and everything is just right...

I'd like a way to force it or KNOW that it should be occuring
versus just expecting it to 'possibly' occur automatically if
all conditions are met and automatically detected.

Or maybe I have this all worng :-)

Thanks!
Steve









> 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 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?
>>
>> Any pointers greatly appreciated!
>>
>> Steve
>>
>>
>> Pretty simple extensions, on lan no nat
>>
>> 
>> [4785]
>>
>> type=friend
>> username=4785
>> secret=test
>> host=dynamic
>> canreinvite=yes
>>
>> [4786]
>>
>> type=friend
>> username=4786
>> secret=tesst
>> host=dynamic
>> canreinvite=yes
>>
>> 
>> exten => 4785,1,Dial(SIP/4785,66)
>> exten => 4785,3,hangup
>>
>> exten => 4786,1,Dial(SIP/4786,66)
>> exten => 4786,3,hangup
>>
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>
>
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> http://www.gnu.org";
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Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(

[steve]

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 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?
> 
> Any pointers greatly appreciated!


Remove from your Dial command all options that require Asterisk to hear 
the media stream.  (T, t etc)

Steve

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Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(

[Steve Gladden]

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 that it does not take anything special.

I've tried what seems to be a simple arrangement,
no nat two phones on the lan same codec,
lack of canreinvite line and also tried
canreinvite=yes

I am not using a global nat=yes statement.
also tried nat=no on each phone just in case of a default
option.


> correct? Does SIP debug give you any info (i.e., does it match the
> right peer) -- you don't show if you allow reinvites globally? What
> about the nat= setting?

I've not set nat= or canreinvite= globally just on each phone

I can certainly try that but having specific settings on the
phones seems to almost guarantee I know where I stand with
those two :-)

I've not torn apart the sip debug on this yet as I am quite new to SIP

but will do so if need be.

Was just trying the simple approach first.



>
> Couple pointers I can give you to get you excited:
> 1) Reinvites work quite reliably, I use them between the PTSN gateway
> and the end user's ATA, all the way across the Internet -- nicely
> reduces latency.
>
> 2) If you use RFC2833 for DTMF you can issue an reinvite and still use
> t/T for transfer. NOTE that you have to modify the source to make
> asterisk reinvite even when it needs to listen to DTMFs. I give no
> guarantees how well it will work for you but it does work.
>
> See "AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1" in rtp.c.
>
> 3) Reinvites *can* work even if both ends are behind NAT. It really
> depends on the NATing router and the ATA. Sipura's and good NAT
> routers work, but I would not call it "reliable" -- it's really
> pushing it a bit...

Yep I will eventually go there but right now still just trying to get it
to work for a test on the lan and have not seen it fly yet.

asterisk always creates a 'native bridge' and seems to hold on for dear
life so far as I have seen :-)


>
> So if you really want to see why your Reinvites do not work, then you
> probably will have to make your hands dirty and analyze where
> ast_rtp_bridge() in rtp.c bails out. But since you are on a LAN it
> makes the situation a lot easier.

Yep!

Still new at this but enjoy getting hands dirty.
Thanks for your time!

Steve




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Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(

[Luki]

> 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
verified that no transcoding is needed (i.e. both ends use the same
codec). Well, then it should work!

Once you get it to work, you can individualize the accounts and no
longer use a global setting. But that's down the line.

> asterisk always creates a 'native bridge' and seems to hold on for dear
> life so far as I have seen :-)
It says "Attempting Native Bridge" but it doesn't tell if you if it
succeeded or not; there was once a notice saying the the bridge could
not be established (failed?) but it caused even more confusion. You
could add some statements to the ast_rtp_bridge() code in rtp.c and
give yourself some feedback -- succeeded, failed because X / Y / Z.

Hope that helps...

--Luki
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Re: [asterisk-users] canreinvite=yes and RTP dropping in and out

[Joshua Colp]

- 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,
> 
> 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 wait 20
> seconds or more, the stream will resume. Flipping the person on and off hold
> won't resume the stream.
> 
> We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem
> to happen all of the time. There are no sip messages being exchanged when
> the stream stops or restarts.
> 
> Any suggestions?

If the audio is going directly there's not too much you can do to examine it. 
There may be software out there to sniff the data on your network and examine 
the RTP stream, maybe even see when it drops out (if it really does drop out, 
ie: stream actually stops). I know there's some Windows software out there 
capable of this as I picked a copy up while at Spring VON but you might need to 
look around. OH - can you also send a sip debug with the reinvites? I'm just 
curious to see the RTP information in the SDP.

> Thanks.

Joshua Colp
Digium
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Re: [asterisk-users] canreinvite=yes and RTP dropping in and out

[Gary Richardson]

My next attempt at this is going to be putting a hub in between the path to the switch. I'm hoping to be able to sniff the packets to see what's going on.Also, using the network status page on the hard phones, the transmit and receive counters for the direction of the channel slows way down as if almost no data is being transmitted. 
How do I send a sip debug?Thanks.On 8/2/06, Joshua Colp <[EMAIL PROTECTED]> wrote:
- Original Message -From: Gary Richardson[mailto:[EMAIL PROTECTED]]To: Asterisk Users Mailing List -Non-Commercial Discussion [mailto:
asterisk-users@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 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 wait 20
> seconds or more, the stream will resume. Flipping the person on and off hold> won't resume the stream.>> We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem> to happen all of the time. There are no sip messages being exchanged when
> the stream stops or restarts.>> Any suggestions?If the audio is going directly there's not too much you can do to examine it. There may be software out there to sniff the data on your network and examine the RTP stream, maybe even see when it drops out (if it really does drop out, ie: stream actually stops). I know there's some Windows software out there capable of this as I picked a copy up while at Spring VON but you might need to look around. OH - can you also send a sip debug with the reinvites? I'm just curious to see the RTP information in the SDP.
> Thanks.Joshua ColpDigium___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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Re: [asterisk-users] canreinvite=yes and RTP dropping in and out

[Joshua Colp]

- 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 at this is going to be putting a hub in between the path to
> the switch. I'm hoping to be able to sniff the packets to see what's going
> on.
> 
> Also, using the network status page on the hard phones, the transmit and
> receive counters for the direction of the channel slows way down as if
> almost no data is being transmitted.
> 
> How do I send a sip debug?

Actually since this happens randomly I doubt that will help. Is there any other 
traffic on the network too? Never know... or a faulty switch? Grasping at 
random things but nothing really comes to mind.
 
> Thanks.

Joshua Colp
Digium
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Re: [Asterisk-Users] Re: [Asterisk-Users Canreinvite=[yes|no] explained (new subject)

[Mike Machado]

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 with the private IP of phone A, but then it sent a
second reinvite with the visible IP of phone A, which I think would have
made the call work in some NAT environments, but then it sent a *3rd*
reinvite to phone B, back to the private IP of phone A, breaking the
audio from phone B to phone A.

What I think happened was when phone A send the 200 OK for its reinvite,
Asterisk saw the SDP info from that packet and triggered the 3rd
reinvite to phone B, but since nat=1 was on, it should have ignored that
SDP, or at least sent the visible IP in the 3rd reinvite and not the
private IP.

In case you don't have it handy, the call trace I am referring to is
here:

http://www.cheapnet.net/~mike/asterisk_excel_with_reinvite.log

In this log, only the 192.x network is nat'd. 10.x and 172.x have
straight routing between them. 10.10.11.77 is the visible IP to
192.168.222.197 according to 10.x and 172.x.



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[asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID

[jonas kellens]

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 displayed on the CLI :

-- Executing [...@intern:1] Dial("SIP/GXP1200-093900c8", "SIP/BT201|30")
in new stack 
-- Called BT201 
-- SIP/BT201-09395070 is ringing 
-- SIP/BT201-09395070 answered SIP/GXP1200-093900c8 
-- Native bridging SIP/GXP1200-093900c8 and SIP/BT201-09395070 

>From voip-info.org I understand that 'canreinvite' means that the
SIP-client will re-invite the other client, so that Asterisk is no
longer in the path...
This is indicated on the CLI with 'native bridging'.

Then why are there 2 sip-channels with a different Call-ID ? The output
shows that Asterisk is still in between !

asterisk*CLI> sip show channels 
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 
192.168.x.x GXP2020 4684b544470 00103/0 0x4 (ulaw) No Tx: ACK 
192.168.x.x BT201 1212e00ffa1 00102/43234 0x4 (ulaw) No Tx: ACK 
2 active SIP channels 

Is there something that I misunderstand here ??

Thanks for the feedback on this !

Greetingz,
Jonas.
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Re: [asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID

[Tom Moore]

Asterisk still controls the signalling, but the audio path should be going
through the phones directly.
Fire up a tcpdump on the Asterisk server to varify this.
 
 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
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=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 displayed on the CLI :

-- Executing [...@intern:1] Dial("SIP/GXP1200-093900c8", "SIP/BT201|30") in
new stack 
-- Called BT201 
-- SIP/BT201-09395070 is ringing 
-- SIP/BT201-09395070 answered SIP/GXP1200-093900c8 
-- Native bridging SIP/GXP1200-093900c8 and SIP/BT201-09395070 

>From voip-info.org I understand that 'canreinvite' means that the
SIP-client will re-invite the other client, so that Asterisk is no longer in
the path...
This is indicated on the CLI with 'native bridging'.

Then why are there 2 sip-channels with a different Call-ID ? The output
shows that Asterisk is still in between !

asterisk*CLI> sip show channels 
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 
192.168.x.x GXP2020 4684b544470 00103/0 0x4 (ulaw) No Tx: ACK 
192.168.x.x BT201 1212e00ffa1 00102/43234 0x4 (ulaw) No Tx: ACK 
2 active SIP channels 

Is there something that I misunderstand here ??

Thanks for the feedback on this !

Greetingz,
Jonas. 
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Re: [asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID

[jonas kellens]

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:
1060
14:38:01.271904 IP 192.168.4.248.sip > 192.168.4.240.sip: SIP, length:
433
14:38:01.272133 IP 192.168.4.248.sip > 192.168.4.242.sip: SIP, length:
861

is what I see... only SIP, no RTP/UDP...

I guess you're right...

Thank you, Tom.


On Sat, 2009-04-18 at 06:50 -0400, Tom Moore wrote:
> Asterisk still controls the signalling, but the audio path should be
> going through the phones directly.
> Fire up a tcpdump on the Asterisk server to varify this.


> 
> 
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