search for: reinvites

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends

...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 it seems there is nothing similar in PJSIP. any help would be much appreciated! -- Jöran Vinzens - vinzens at sipgate.de sipgate Gm...

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 through the asterisk box in the middle. Using sip debug I even can't find

...ge 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 it seems there is > > nothing similar in PJSIP. > > The "direct_media" optio...

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 immedi...

...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 remember Cisco Phones and ATA's have some reinvite issues, wondering if the same applies to the CCM and how it handles reinvites? In the wiki example of integrating CCM with Asterisk, the SIP context shows canreinvite=yes, so should this be working ok, maybe I'm doing something wrong? Thanks JR JR Richardson Engineering for the Masses -------------- next part -------------- An HTML attachment was scr...

...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 connection information of phone B Phone A responds to this with a 488 Not acceptable here. Asterisk ACK's and sends another set of reinvites to both phones with the correct codecs and connection information (the asterisk box itself). Both phones reply 200 OK. Asterisk ACK's and immediately sends BYE to both phones. Is this normal behaviour? I would expect one of two things: - Asterisk intelligently does not even try reinvite becau...

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

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 able to use

I'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

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 ranges of

...all is transfered back, there will be one way audio. It seems this is Always how it is, over and over.. The Originator Cannot transfer the call first. I THINK if the destination transfers first, THEN the originator can transfer.. I've checked netmasks, ips, gateways, etc, etc.. The SDP on the reinvites looks ok.. No Nat, no funny business here.. just IP routing.. Any ideas? -Brett

Hello, I know there's a problem with Asterisk at the moment in that while it's easy for one caller to dial another (using the dial command), it's tricky to connect two calls that are already in progress. I've been using MeetMe to achieve this (with each caller's call being directed to a dynamically created conference room programatically), and this is working - kind of -

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

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 quite well. However, even with this option set, Asterisk would reINVITE itself back into the au...

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?

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 ent...

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