similar to: Call External Program When SIP Message Arrives

I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in

I am beating my head against a problem where queue calls offered by Agent channel to a SIP UA cannot be REFER transferred if the target UA/extension hasn't accepted the call. If the members of the queue are SIP channels, this is not a problem. I suspect chan_agent isn't flagging the bridge from Zap/n -> SIP/n properly, or this is by design. The following line is what is spoken before

Hey gang, I'm trying to work out all possible scenarios using SER & Asterisk in our upcomming deployment. The example scenario is 50 different customers, all with different numbers of SIP UAs. All UAs would register with SER; This will help keep any inter-office conversations off our bandwidth since SER doesn't handle the RTP stream. Calls from PSTN to UA are easy to handle.

Hi all, I have the following setup: UAs ------------SER ------------------------ ASTERISK ---------------------GNUGK --------------- GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out

Hi all, Can someone help with with Asterisk, SER, and Asterisks Queues? I have three servers: Server A: Asterisk with TE410 connected to PSTN Server B: Asterisk connected to Server A via IAX2 trunk Server C: SER where SIP agents register/connect to What I wanted to do is configure Server A so that it would route certain DIDs to specific UA that are registered in Server C. I don't think

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am getting a "Got SIP response 481

Hi, I downloaded the stable branch of asterisk a couple of month ago, and I'm using it as a SIP UA voicemail server with SER, and my setup works fine. I do have a list of phones defined in voicemail.conf, in the sip.conf file I only have the setup of asterisk as a peer registering to ser. The extensions.conf file contain the extensions that link to the voicemail application. This setup is

lqbal, I do plan on having alot of users. Two markets I'm trying to get some volume users from are: residential consumers and business users. Residential consumers should get basic line services such as their own DID, voicemail, caller-id, call-waiting, three-way calling, and basically, all the standard features you get from companies like Vonage, etc. This particular market base

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail.

Hi all, i'm tring ro use sip with an external sip proxy as vocal or ser. My scenario is Vocal or SER ----> Asterisk with cnah_oh323 -----> Gatekeeper I would like that sip termial register themself to Vocal or ser and the h.323 terminal to gatekeeper. When i place a call from h323 side to sip side all work.... When a try to place a call form sip to h323 nothing happen Does

Waldo, How do you let your customers manage 'their' PBX. I too have a setup like you. However, I installed a * server for each customer, via vserver. I'd like to now what kind of software/webbased package you use for this. I also have SER installed as a front-end server for the * servers. But, as I'm still not very into SER, don't know exactly how this fits in. Should I use

Hello i am trying to use this exmple with SER-0.9.3 but still NATED Clients are not working any other requirement http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper ----------------------------------------------------------- # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters

Hi all, Hope somebody can help-I really am stumped as to why this won't work. I realise that this isnt an Asterisk problem (Please dont bash me on the list) and I have emailed the SER list but I havent received a reply and maybe someone on this list can help...Once this problem is solved I am going to use Asterisk for voicemail etc with SER (I have it set up) I currently have SER set up and

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there

Hello, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

I've been reading a bit about Enum and DUNDi and still have something not very clear to me. This is a HYPOTHETICAL scenario: I have 4 asterisk servers. All of them are handling registrations of both SIP and IAX2 UAs. SIP agents are being load balanced by something like SER. I have another server in charge of load balancing IAX2 UAs registration (some sort of dynamic firewall telling

Using Asterisk 1.0.9 When I try to make an outgoing call with SIP I get the message " 513 Message too big" back from SER. Any ideas what I am doing wrong? Debug below. SER and Asterisk are running on the same Server SER is on port 5060 Asterisk is on port 5061 In my extension.conf I have the line SERADDRESS=192.219.85.57:5060 in Globals and am using exten