similar to: Parsing incoming extension till first @

Hi I do have asterisk running in real time I do want to add voicemail to real time. I did follow : http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail However when I do try to make a voicemail I do get : [Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! -- Executing [999alijawad at a2billing:1]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Guys Ive got an internet cafe on which I have a debian sarge box running. The Debian box acts as a gateway and it has masquerading on. I have 40 client PC and i do not want to assign more than 64k per pc for upload and the same is true for download too. Ive done alot of research and Ive read tutorials about CBQ and HTB. I found that CBQ.init is

oops, i forgot to reply to the list :-/ Début du message réexpédié : > De : Vincent Dautremont <vdautrem@ulb.ac.be> > Date : 30 août 2007 16:58:26 GMT+02:00 > À : Ali Jawad <alijawad1@gmail.com> > Objet : Rép : [LARTC] Priotirize SSH Traffic > > try that > #tc qdisc add dev eth0 root handle1: prio > # tc filter add dev eth0 protocol ip parent 1: prio 1 u32

Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN

https://bugs.freedesktop.org/show_bug.cgi?id=109876 Bug ID: 109876 Summary: JIYE BHUTTO Product: xorg Version: unspecified Hardware: x86 (IA32) OS: Windows (All) Status: NEW Severity: critical Priority: medium Component: Driver/nouveau Assignee: nouveau at

fileUrl <- "https://data.baltimorecity.gov/api/views/dz54-2aru/rows.csv?accessType=DOWNLOAD"download.file(fileUrl,destfile="./data/Cameras.csv",method="curl") I tried it after installing package "RCurl" but it give error message: Error in download.file(fileUrl, destfile = "Cameras.csv") : unsupported URL schemeI can you help me to solve this

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on "pause". I'm using chan_agent for the agents, so agents are

I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get:

Yell at me if you will, but I hate func_realtime - it's not very usable nor is it change-friendly (update your database and your dialplan completely breaks). I'm getting a new 1.6 box built out and working, and wanted to emulate the functionality of APP_realtime somehow, so I started digging around in the func_realtime source - here's what I came up with: For 1.6.0, look at line 86

I have two ISP connections, and am having some issues. I can connect to any services on the firewall, like the smtp gateway, but anything on the internal server only works from one connection. The lartc guide has a good example for what to do for services on the box, but leaves it open for how to handle services on an internal host. I''ve tried using iptables to mark the packets

hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --------------------------------------- Marek Cervenka =======================================

Hi guys! I'm having one or two issues with queues hosted by an Asterisk machine where the clients are on a legacy PABX - at least for the interim. I fully expect most of these issues to be non-resolvable, but thought I'd at least ask to find out if there is some way of working around the issues. The legacy PABX is an NEC 7400 ICS connected to Asterisk via an E1 ISDN link. Calls are

Hello users, Can you share with me your experiences with Asterisk 1.6? Is it stable enough for commercial service? Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080901/d849d412/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi guys, I've been working on a little dialplan fragment for roaming extensions, however the customer wants us to set the MWI indicator for the roaming extension that has just logged in. We're using MySQL realtime, so I've figured out that RealTimeUpdate will happily update the realtime database with the correct mailbox. My problem

Greetings, I''d like to mark incoming traffic based on TOS to use the mar for backtraffic routing. I have two gateways on the same net and incoming traffic may arrive from any of them. I want the return packets to go the same way. My plan is: Normal traffic goes through default gw. Traffic from the other has TOS 0x08 set. I''d like to mark traffic with TOS and use fwmark

Dera All, I have the following scenario, A customer dial a DID number...The call is routed to a PSTN GW that send the call to asterisk... On asterisk I created an AGI Script that send the call to an extension registered on OpenSIPS server... The extension is ringing successfully, but as soon as I accept the call on OpenSIPS side the call is hangd up... I checked rhe SIP debug and it seems that I

Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I need to conect an Asterisk with an Alcatel OmniPBX 4400 using an E1 port. It is the first time I make this kind of connection and I do not know exactly how to get it working. Someone has experience with this kind of connection? Could you paste a zapata.con and zaptel.conf files with QSIG configuration? Any clue will be wellcomed. Thanks Voipcrazy -------------- next part

Hello, I am trying to make a call from our PABX to Asterisk on PRI interface. How can i configure Asterisk to enter the overlap receiving state if the complete number is not obtained in setup message. Looking forward to any help in this regard Regards Nauman Bin Ali __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection