similar to: Refer Method

I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways. I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites. Got SIP response 500 "Internal Server Error" back from 10.1.3.28 SIP/alma-1b77 is circuit-busy Everyone is

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone

For about 10 months I have been running a succession of Asterisk SVN trunk versions on an Athlon 64 X2 4400+ based machine with OpenSuSE 10.2 at my home. I have a variety of SIP phones (mostly Polycom) internally; my external connections are two POTS lines on a TDM400P (with HPEC) and an IAX2 link to a VoIP provider. I had Asterisk configured to allow G.722 and u-law on the Polycom phones,

Hi! I have some problems understanding the concept of REFER in Asterisk 1.4.23. I have the following scenario: Incoming SIP call (incoming leg) from a SIP trunk into Asterisk (handled in context fromTrunk), forwarded to the SIP Client (outgoing leg). Now, the SIP Client sends a REFER request (unattended transfer) to another extension. This terminates the outgoing leg and the incoming leg

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

I am currently using Asterisk with G.711 codecs and in-band DTMF for several Cisco 7960's and an Audiocodes GW. When allowing out-of-band DTMF, I could use voicemail menus and anything else on Asterisk that required DTMF but I could not get the DTMF relayed out of the GW. Has anyone verified that this works between 2 SIP devices? If so, I would be interested in your settings. Also, I would

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Hello, we are using Asterisk with Adhearsion as our application server, with another Asterisk box acting as the office PBX, where all office phones are registered. A REFER to transfer calls within the office results in the Adhearsion application call being dropped, because the leg between the PBX and the app server is terminated by the PBX following the REFER. Is there a way to configure

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

Hi. If I use a Cisco as a PSTN termination GW and need to route all incoming isdn calls to my asterisk and all outgoing calls from asterisk via the cisco out to pstn, how do I do that ? in the cisco I have this: dial-peer voice 1 pots destination-pattern [0-9]T no digit-strip direct-inward-dial port 3/0:D ! dial-peer voice 50 voip destination-pattern [0-9] voice-class codec 1 session

Hi people. I've been having some audio problems with some of my cisco 7940 phones using firmware 7.1. The sound gets gargoled up, I can't understand much of what is said (listing on the IP phone). My setup is the following: 7940 - * - Cisco 2621(GW) - T1 I'm using SIP and g729 for all stages of the communication. I was using ulaw for the leg between * and the Gw, but some guys on

Hello, Is it possible to use * as a gateway in the following setup: LAN (with Windows NT/Linux PCs) | Ethernet (IP) | Linux PC with * and AVM Fritz! ISDN Adapter | ISDN | Someone with a analog/digital phone (POTS) Basically, people sitting on their PCs will wear a headset, and whenever they want to call someone,

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

Hey all, Alright. So we decided to not go with Avaya for our next PBX and we are now full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP gateway and call center and Lync is our internal UC and IP-PBX server. I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all is beautiful (except for the Opt11 not receiving names from * but

For a simple call file like Channel: Zap/g1/XXXXXXX RetryTime: 60 WaitTime: 30 Context: from-file Extension: s Priority: 1 I noticed that s@from-file started to execute regardless of the state of the outgoing call. Is this supposed to be? So far I can only set a Wait() in the local leg and hope the remote party picks up soon enough. I thought call file extension will start execution only

Hi guys i upgraded to asterisk 16.10 and in most recordings you here only leg A in the recording sometimes you might hear a word of leg B Did any body hit this problem? Thanks, israel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200512/90ee8dc2/attachment.html>

Dermot MacSweeney pointed out to me that after my "fix" of legend(), points were no longer coming out placed in the middle of the lines, but at the right-hand edge. It turns out that naively swapping the order of point-drawing and line-drawing also messes up the bookkeeping that legend() does on the current x-location. Here's my patch, which fixes that bookkeeping (and incidentally

Hi, It isn't maybe the best place to ask the question, but I don't know better :( Does anyone could tell me if sending REFER request virtually ends current call? I mean if one sends or receives REFER request, he should stop rending RTP, just as it is required for BYE request. In other words is is more or less equivalent to the BYE request? Regards, Marcin Okraszewski

Hi to all, I am sending a SIP REFER message to Asterisk from a VoiceXML application using the <Transfer> element to do a Transfer through Asterisk. I need to know if Asterisk supports the full features of the SIP REFER message because if i set 'bridge=true' in the <transfer> element of the VoiceXML application to supervise the call, Asterisk sends a NOTIFY message with