search for: callflow

Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace

...ll flow is agent -> SIP -> callcenter asterisk -> NEW call over IAX -> routing server -> PSTN Then agent hangs up - so that the original caller and the new call will get connected - and - it is working But - the call will not get released on the callcenter asterisk machine So the callflow after the transfer is Original call PSTN -> routing server -> callcenter asterisk -> routing server -> PSTN But it should be Original call PTN -> routing server -> PSTN I have transfer = yes and mediaonly both tested on my connection routing server to asterisk callcenter - doe...

Does anyone know how to do this? Just curious, ie SIP callflow A -- Asterisk -- B, RTP goes directly from A to B .. Matt

...possible if you have a central registration server that handles all of your dialplan routing and several asterisk PSTN gateways that it routes calls to for an outbound SIP conversation using reinvites and NOT have the registrar box try and send ANY RTP traffic back to the client? It looks like the callflow goes like this, currently: Client invites, registrar contacts PSTN-GW, registrar sends invite back phone to setup RTP between phone and registrar, registrar then sends invite back to phone to setup RTP between phone and PSTN-GW. what I want to do: Client invites, registrar parses dialplan and fo...

Hi, I'm using Asterisk 1.2.3 with the asterisk-oh323 channel driver, version 0.7.3. Pwlib is V1.8.7 an OpenH323 is V1.15.6. Following CallFlow: SIP-UA -> OpenSER -> * -> CCM OpenSER routes all calls with prefix 60 to Asterisk, where I've configured following extension: exten => _60.,1,Dial(OH323/${EXTEN:2}@v.w.x.y) v.w.x.y is a Cisco Callmanager where Asterisk is configured as a H.323 Gateway. When a call comes in fro...

Hello Users, My Setup is like this openser --Registrar asterisk --Callflow using asterisk-b2bua + radius for accounting My Intention was to generate a Acct-Stop Packet when there is a failure of RTP media from one of the UAC's( callee or caller) who is in dialog. so that the Caller will not be charged for Meaning less network problems Because there is no way...

Hello all. Has anyone had any success using ChanIsAvail with only SIP channels? Is there another, better way to check if an extension is busy without dialing it? Thanks, B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040109/48ac2c3e/attachment.htm

...is is a seminal release for us as the product is now more stable, powerful, and easy to use than ever. We've also added a new ActionStep "CallWSByWSDL" that allows you to easily consume Web Services from your Saflet, providing you with even more integration possibilities for your IVR/Callflow applications. The release of this ActionStep is merely the first of many non-telephony-specific ActionSteps that are in the works, representing an overall commitment to making SafiWorkshop and SafiServer the most flexible and powerful development and integration tools for Asterisk on the market (f...

    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do:

Ok, here's a weird one. I have an AGI script where one user calls another. The call is answered. Everything is peachy. If the call is terminated by the CALLEE hanging up the call, then Asterisk returns control back to where the Dial() command left off, and I can check the return code of Dial(), ${DIALSTATUS} etc. That's all great. HOWEVER, if the CALLER hangs up the call, it seems as if