search for: reinvite

....biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends up >> sending a reinvite which their side & they do not support us sending a >> reinvite. Ive tried: >> >> canreinvite=no which was supposedly replaced by: >> >> directmedia=no >> >> Can anyone shed any light on this matter? I'd love to get this fixed. >> > >...

Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is there any possibility to deactivate this kind of reInvite? We have some race conditions while have multiple asterisk in the call flow and the different asterisk systems are sending this reInvites out parallel. While an invite is pending on a system it is not accepting another incoming reInvite from peer. With chan_SIP canreinvite=no solved the issue. But...

Hi, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes...

Hi all, So the scenario is: A -> Asterisk -> B after B send back 200 OK Asterisk is answering the call to A. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. Here i do not understand why this could not be done in the 200OK to A? As far as i understood you Josh, there is no way to prohibit this kind of reInvite? It is not abou...

Hello! I'm facing ReInvites as caller from UAS despite configured session-timers=refuse (which can be seen in the SIP trace) always after 900s. (The behavior is the same if session-timers is set to accept). This just happens with one provider (German Telekom to callee at kabelbw). - The incoming ReInvite is answered immed...

Hi All, I haven't started sip traces or debug yet, but was wondering what the deal is with the CCM and reinvite, why it doesn't work with Asterisk (using 1.2.9.1). I can make calls back and forth all day with canreinvite=no, when I try to reinvite an inbound sip call from the CCM with Asterisk server 1 to Asterisk Server 2, I get one-way audio issues. All the RTP ports are configured the same. I remem...

Hi, I just ran into what seems to be an issue on re-invites. I'm not sure if it's a bug or as designed, so I thought I'd ask the question. Here's my setup: - Asterisk 1.8.13.0 - Phone A: Polycom ip331, only allowed to use ulaw, canreinvite=yes - Phone B: Polycom ip330, only allowed to use alaw, canreinvite=yes Phone A calls the extension of phone B. After the normal call setup asterisk tries the reinvite: - To phone B it sends an SDP which asks alaw and connection information of phone A - To phone A it sends and SDP with only the c...

Hi All, We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they need to go. We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity. Our side ends up sending a reinvite which their side & they do not support us sending a reinvite. Ive tried: canreinvite=no which was supposedly replaced by: directmedia=no Can anyone shed any light on this matter? I'd love to get this fixed. There is no firewall on this machine at all. Thanks --Tammy

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not...

...#39;ve an Asterisk server running in my office, which forwards all long distance calls to a third party SIP service using an extension rule: exten => _1XX0.,1,Dial(SIP/{EXTEN:4}@external_sip_server.com) (1XX0 is the international calls rule for Chile) Also, in my sip.conf, I've defined canreinvite=yes to decrease the network load to the server caused by the RTP. However, the external sip server seems to be buggy, because the REINVITE's against it only works for certain routes, and in others, it simply hang up the calls. Since I don't have control over that remote service (and I...

Hello, all. We are having a problem where audio for sip channels is dropping upon reinvite. Perhaps it reflects a misunderstanding of what reinvite does. We are running Asterisk 1.6.1.1 on CentOS 5.3. SIP is set to canreinvite=nonat. We have tried RTP with strictrtp set to both yes and no. We have also tried extending the Asterisk rtp port range to accommodate the differing default...

Hi All, I'm having a really weird can reinvite issue. I've been banging my head around on this for days now.. I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5 172.20.0.11 is a hosted box and serves multiple offices 172.20.2.5 is a box on site at a customer's office. A phone at 172.20.128.10 makes a call using serve...

...d this is working - kind of - but this results in a conference instead of a bridged call, so - we can't use the normal Dial parameters for transfer etc, - the other caller is not disconnected automatically when one party hangs up, and - (most importantly) we can't get SIP to reinvite. The SIP reinvite issue results in increased bandwidth costs, extra latency/echo and reduced call quality when compared with Dial (as the media path has to include Asterisk with MeetMe, but not with Dial). Does anybody know of any other way to bridge two existing calls with Asterisk, that will...

Howdy, How can you tell if RTP traffic has been reinvited/is bypassing an * server? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com

I have a setup with a number of customer Asterisks with T.38 enabled. This works quite well for each customer sending faxes between branch offices. They all have a SIP trunk to a central Asterisk, which connects them to the PSTN through various providers on dedicated lines. I cannot enable reinvite on those SIP trunks, because that would allow calls from the customer's phones to get reinvited and talk directly to the central Asterisk -- and there are firewall rules forbidding that. Is there a way to get ONLY T.38 reinvite without Asterisk trying to get out of the media path? /Benny

Hello all. I am currently using Asterisk 11.7.0 (also tried Asterisk 12, but same behavior) and is having an issue when it comes to reINVITE on BYEs. Apparently one of the SIP providers that I am using does not always process reINVITEs correctly, and would return a 500 Internal Server Error message on some (but not all) of these transactions. To get around this issue, I have been using directrtpsetup = yes in my sip.conf, and it worked...

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 pho...

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38 info in SDP with G711uLaw....and fax fails. How do I configure the host en...

Hello all! I can see a strange problem during invite in dialog in the context of timer handling. Given is the following incoming call from provider at 8.195.88.234 (2 at 2) to my asterisk at 28.19.57.152 (1 at 1): After 900s suddenly *asterisk* starts the timer reinvite - I would have expected the reinvite started by the provider as usual. The expected reinvite by the provider is started during authentication of the reinvite started by asterisk and is answered immediately by asterisk with sip 481. The answer of the provider after the resend of the reinvite came...