similar to: SIP reinvite code negotiation

Using R 2.0.1 on Windows XP, I am getting an error msg: Error in .Fortran("conic", nxy = nxy, npt = npt, CP = cp, EP1 = ep1, EP2 = ep2, : Fortran function name not in load table I am wondering if there is a way to see what function names are in the load table? Maybe the function name has been altered? The first thing I do in my analysis script is to load a DLL, conic.dll,

Hello, I wonder if anybody is using PJSIP realtime in production environment? I've started to play with it and encountered many problems. Here's my config: sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints extconfig.conf: [settings] ps_endpoints => pgsql,users,pjsip_endpoints_v pjsip_endpoints_v is postgresql view. 1. The biggest problem: if I have small number of endpoints

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

Dear all, I have a function to optimize for a set of parameters and want to set a constraint on only one parameter. Here is my function. What I want to do is estimate the parameters of a bivariate normal distribution where the correlation has to be between -1 and 1. Would you please advise how to revise it? ex=function(s,prob,theta1,theta,xa,xb,xc,xd,t,delta) { expo1=

asterisk-users-bounces@lists.digium.com wrote: > Isn't it possible to use T.38 for interconnecting hardware gates > supporting T.38 with asterisk using SIP REINVITE? > I'm not shure but but think its's might be possible because after > reinvite traffic goes directly from one gate to anotger, not over > Asterisk We've seen a problem here with asterisk. Wehn

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

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

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

I'm having issues with T.38 on Asterisk 1.6.2.8. A few lines are received OK and then I get only garbage. I'm using ReceiveFAX() provided by app_fax to receive the faxes. After talking to the engineers on the telco, they said Asterisk is sending a REINVITE to alaw after the T.38 reception is complete, and that could be the cause of the problems. I personally am not totally convinced of

Hi Although canreinvite option is yes, the asterix doesn't send reinvite and the media is going through the asterix instead of between the two sip phones. Both sip phones (handytone 486) don't use NAT and are configure with canreinvite option yes and use the same codec G.729. And Dial() command don't contains t or T. Any suggestion on what could be the problem ?

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

Greetings- Working with the T.38 gateway functionality that is sparsely documented [1] , I'm attempting to get the following functional: Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box in question) -> SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from

Hi, I've finally given up on trying to fax over my Digium TDM400 card. I've found that fax over VoIP is quite more reliable (at least I can receive the faxes). My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes everyday (just ocasionally), i pretend on using g729, unless a fax is detected. Is there any way to force asterisk to make a reinvite, and swap the codec on

For a sip-only application, what exactly is required to ensure that calls completed via followme are reinvited? Can it at all? The code after outbound = findmeexec(targs, chan) calls ast_bridge_ call(). I don't see anything there which can cause a reinvite, yes? When the same peer is used for both the incoming and outgoing legs, it is a bit of a waste to proxy the rtp. And even when the

Steve, My Problem is not a problem, with the codec negotiation between end points. But when asterisk does it with canreinvite=no, * do not do it right. I replied with a lengthy discussion about my findings here, This behavior can be reproduced. But '*' do not seem to do the negotiation correctly. http://lists.digium.com/pipermail/asterisk-users/2004-January/032197.html

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

Hi, In a VoIP only environment, Asterisk has to use ztdummy to have any chance of playing back understandable audio files (without drops, hickups etc). I have been using ztdummy to some degree of success, but I also have a "Wildcard TDM400P REV E/F Board 1" in the Asterisk machine I'm using. I'm not using this card for anything at all, but I'm wondering how to set it

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

It is not clear how exactly g729 pass-through can be enabled. I have a SIP call off a gateway come into an Asterisk menu, and then I send the SIP call to another SIP gateway using Dial(). Even though codec preferences have g729 listed first, it never gets used. Both gateways have separate peer entries in sip.conf, and both have canreinvite=yes set. Can Asterisk change the media type during

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