similar to: question about reinvite

Hi I am trying to setup asterisk at home, I have 1 in bound VSP (I have a register cmd setup for that in asterisk). At home I have a cordless phone with 2 line capability - I currently have 2 spa3102's in place to handle the 2 lines ( I am in the process of buying tdm410 to handle to handle this and the backup pstn line). I also have 2 laptops setup with soft sip phones. What I would like

I've been wondering for a long time why my reinvite option is not working with Broadvoice anymore. It happend during the time Broadvoice was having a lot of issues, so I decided to wait. Recently I decided to test the same sip.conf with another VSP (SIPphone) and it worked fine! No issues on the reinvite. Note: clients and server using ULAW (only), no NAT or firewalls, public ip address and

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.

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

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

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

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 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, 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'm using Asterisk a SIP Server for a lot of GrandStream HandyTone ATA's. Each one of them is configured in sip.conf as: [1234567] type=friend username=1234567 secret=1234567 callerid="ATA 1234567" host=dynamic nat=yes qualify=yes disallow=all allow=g729 canreinvite is set globally to YES. When one ATA calls another, I see the next traffic on Ethereal (just shown the sequence

Hello all, I am allowing a reinvite between a snom 320 phone and a SIP gateway to take load off my Asterisk server. When I put the caller on hold, for example, Asterisk successfully reinserts itself into the rtp stream to play music on hold to the caller, but when I do a chanspy Asterisk does not seem to pull the call back. If I am spying on a channel when the call build up happens the reinvite

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

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've been playing with T.38. I observed that mostly but not always, it's the "calling endpoint" that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the "standardized" or most common, way to start a T.38 session ? Shall it come from callee or

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 everyone, I've got the same problem described in http://bugs.digium.com/view.php?id=10647 (unfortunately, the bug is closed and I could not find the way to reopen it). Wiki says, " When options t, T", "h", "H", "w", "W" or "L" (with multiple arguments) are applied, Asterisk will remain in the media path, even if

Hello, I've just set up DISA on my * server. I'm using it to avoid cellular overseas calling charges from support staff in the field at our customer sites. Support staff often spend hours on the phone to our UK factory. However, I'm not sure about the implications of reinvite in this arrangement. A support engineer calls in to a DID that I have from VoicePulse Connect. They match

Hi, We're routing SIP calls through Asterisk and we want to be able to reinvite calls without Asterisk performing codec conversion. We've performed the following test: Asterisk has license for G.729 installed sip.conf [general] context=default autocreatepeer=yes disallow=all allow=alaw allow=g729 canreinvite=yes nat=no We have configured two endpoints: EP1, preferred codec order

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 -