|Summary:||ASTERISK-03532: Disabling SIP-SIP RTP call bridging with reinvite|
|Date Opened:||2005-02-17 08:47:39.000-0600||Date Closed:||2011-06-07 14:05:16|
|Description:||Can I disable the SIP-SIP RTP Call bridging regardless of the following rules: -|
If one of the clients is configured with canreinvite=NO, Asterisk will not issue a re-invite at all.
If the clients use different codecs, Asterisk will not issue a re-invite.
If the Dial() command contains t or T Asterisk will not issue a re-invite
|Comments:||By: Mark Spencer (markster) 2005-02-17 08:58:06.000-0600|
What is your feature request? This is not a technical support forum.
By: amzaka (amzaka) 2005-02-17 09:01:13.000-0600
I require this feature. Since I think this is bug in asterisk that even local calls are being bridge. And you cannot use the asterisk as a PBX server on internet with sip endpoint clients connected on leased line.
By: amzaka (amzaka) 2005-02-17 09:04:09.000-0600
Only want to use Call bridging when it should be needed. Not in major number of cases.
By: Kevin P. Fleming (kpfleming) 2005-02-17 09:15:07.000-0600
Asterisk will not issue a reinvite if the peers are using different codecs.
Asterisk will not issue a reinvite if it needs to listen to DTMF during the call (for transfers or any other features).
Asterisk will not send reinvite to peer A if peer B is set to "canreinvite=no". Asterisk will not send reinvite to peer B if peer A is set to "canreinvite=no".
None of what you are asking for is new features, and there are no changes needed. If you are having trouble with "asterisk as a PBX server on internet with sip endpoint clients connected on leased line", then you have enabled reinvite for peers that should not have it enabled.
You say "Only want to use Call bridging when it should be needed", when in actual fact you mean direct RTP bridging. "Call bridging" is used every time two channels are talking to each other, it's always needed. "Native bridging" is used every time two channels are talking to each other that use the same media transport (RTP), it's not always needed but it's never harmful. Direct RTP bridgine (using reinvites) is only used when you allow it to be used.
If you are so concerned about this, set "canreinvite=no" for every one of your SIP peers/users and you will be done.
By: amzaka (amzaka) 2005-02-17 09:50:44.000-0600
Can you tell me how can I be done "If you are so concerned about this, set "canreinvite=no" for every one of your SIP peers/users and you will be done. " If using 300 active SIP channels simultaneously on 2Mb DSL connection. Since they would become 600 and bandwidth limit will be reached even for the local calls.
By: Kevin P. Fleming (kpfleming) 2005-02-17 09:55:03.000-0600
You are not making any sense here at all.
First you asked for Asterisk to do what it already does. Then you said you don't want to use media-path reinvites except when it's needed, and when I told you how to do that you changed your story.
Please explain _exactly_ what is not working the way you want it to, and how you want it to work. Ambigous requests are very hard to answer, since we cannot read your mind.
By: amzaka (amzaka) 2005-02-17 10:11:43.000-0600
OK!I want to setup the following infrastructure using asterisk. And want to use all feature of asterisk like call forward, transfer, conference call, music on hold and call queuing etc. And want to use the PSTN contivity via SIP provider.
And want to use PBX server on the Internet. With 300-1000 local SIP extensions connection over the 2MB DSL link.
I just want the direct RTP to and from media streams in between local SIP extensions. At the same time use the same extensions to use call features like transfer and forwarding etc.
By: Kevin P. Fleming (kpfleming) 2005-02-17 10:24:48.000-0600
I don't think you'll achieve that many extensions over a 2Mbit link, but it's possible.
What you want do to today is not possible; currently it's all-or-nothing. Either the media goes directly between the peers because you have _disallowed_ internal features, or it does not because you have _allowed_ them (whether they are being used or not).
Asterisk does not currently have a way to pull the media "back" because one of the parties in the conversation pressed a key that causes a feature to be invoked; however, in my spare time (<G>) I have been working on exactly this functionality. However, it will only be possible to use this if you are using SIP INFO for DTMF, because the other two methods transfer the DTMF via the media path, and if Asterisk is not in the media path it won't see the DTMF. With SIP INFO, the DTMF is sent via Asterisk even if the media is not, so it would be possible to pull the media path _back_ into Asterisk if one of the peers hits a key sequence to invoke a feature.
As far as mismatched codecs, Asterisk will never be able to send the media directly between those peers, because they cannot understand how to talk to each other. That is not an Asterisk limitation, it's just how SIP works. There _may_ be a way to accomplish that in the future, but it will be difficult to do.
(See how much more helpful the response can be when you actually state your problem clearly? Thanks!)
By: Olle Johansson (oej) 2005-02-17 10:28:47.000-0600
amazaka: You need to learn a bit more of Asterisk and SIP before you continue this dicussion. The bug tracker is *not* the forum to discuss this kind of features, as several of us has pointed out. Take this discussion to the mailing list or the #asterisk IRC channel.
You can build what you want with a SIP phone that supports native SIP transfer, DND and other features so you do not have to listen to DTMF and force the media through Asterisk. Canreinvite=yes is enough for that scenario.
With that said, I close this bug.