similar to: DiD for less then $4

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

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

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

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 immediately by asterisk (Status 100 / Status 200 - 0.02s). Media stream

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 think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

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

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 -

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

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

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

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 server 172.20.0.11 to a phone at 172.20.2.80 via server

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

Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat?d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this

Here is the setup: Phone A (in NYC) on own bandwidth. Phone B (in LA) on own bandwidth. Asterisk box in Houston,TX on own bandwidth. Both phones contact asterisk to register. Not much bandwidth used for this as it is a few packets every hour or so. Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk calls phone B. Both phones are connected and both people are talking.

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

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

Hello, as a variation of our issues with Adhearsion calls dropping when an INVITE comes in for a bridged call, I now have a new issue to contend with. Our call is in an AsyncAGI application, and has been bridged to another channel. The provider that supplies the DID sends a polling reINVITE every 15 minutes (it's a documented Metaswitch behavior amongst others). The reINVITE is seen as a new

Is anybody using Asterisk with a dual WAN router (Xincom XC-DPG602, Hawking H2WR54G, Fortinet FortiGate-60, SonicWall etc) ? -- #Joseph