similar to: Linksys SPA941

Dear Group, I'm working on a call recording solution and would like to have the ability to initiate a 3 way call based on an incoming call. One party will be an AGI that I have other will be an outbound call via a second T1 interface. Does anyone have a working configuration for an Asterisk initiated 3 way call? Thanks and Regards Shad Mortazavi

Dear Group, Does any one have experience using SoundPoint(r) IP 300? I have one call center on Snom 200's I'm adding a second and was looking at the SoundPoint, but needed some input. Thanks Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was

Dear All, I'm running version 0.7.1 of Asterisk server for our global help desk. We have put together a comprehensive reporting package for static's from the CDR. I'm not able to calculate the time a call is in the queue before it goes to an agent? I would appreciate help with working this out. Warm Regards and Thanks Shad Mortazavi

Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 =>

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

Dear All, I have been running a successful and very stable call center PBX based on 0.7.1 release. I need to be on this release because of a number of features that I have complied from 3rd party patches, for the call center. I will not be able to upgrade to release 1 until the patches catch up and I have done the required testing. The system was very stable until two days ago. The changes made

Dear All, I have the following setup. Quad T1's<->Asterisk (PBX)<->(LAN<->DMZ)<->SER<->(Firewall)<->(Internet) | Local US Help Desk (Snom 200') This setup works well. I can pass calls from over the internet to the Asterisk PBX via SER using X-Ten Lit. I have a couple of questions; 1. How do I

Dear Group, I have been successful in setting up the Agents, queues and getting agents to log in. Is there a way that I could configure the system so that the agent is called back. i.e. the agent logs into the system, a call is destined for them and their phone rings. If some one has this setup I would be very interested in hearing from them. Warm Regards and Thanks --------------- Shad

Dear All, I have a number of call queues defined in Asterisk. I would like to program a system attendant that tells people; 1. Every 60 seconds 'Your call will be answered as soon as possible' 2. Tell the user how many calls are on the queue. I would then like them put back on hold music. Does someone have a configuration for this or something similar? Your help would be greatly

Dear Group, At the moment I use SJPhone as my soft phone with Asterisk. I prefer the look and feel of the x-ten lite. However, when ever I use my x-ten lite I get a lot of breakup in my communication. E.g. I will play some hold music, and every 5-6 seconds I drop some packets. I don't have the same issue with SJPhone. I'm sure this is a configuration issues, but I can work

Dear Group, I have my Asterisk box working with a Mediatrix 1204. I have 2 questions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi

Does anyone know how to set the distinctive ring on the Linksys SPA941? I want to be able to dial one extension and have the phone ring with a certain tone and then dial another and have the phone ring with a different tone. I have tried the following ------------------------------------------------------------------- exten => 802,1,SIPAddHeader(call_info=Classic-4) exten =>

Dear All, I seem to have stumbled across an AGI problem; I have written an AGI Script (bottom of this email); The script does the following; Makes a CDR entry when called Records the call Updates the CDR Finds a corresponding DNIS from the SMDR table (captured via a serial port logger) Matches up the record and updates the CDR. The script works perfectly in my test lab and has been doing so

Dear all, I have the following lines in my extentions.conf file; ;All US Calls exten => _9001XXXXXXXXXX,1,Dial(IAX2/dornoch:xxxx@10.xx.xx.xx/${EXTEN:1}@outbound) ;Dial 9 for outgoing numbers exten =>_9.,1,Dial(Zap/g1/${EXTEN:1}) ;include Brunswick switch => IAX2/dornoch:xxxx@xx.xx.xx.xx/sip What I'm trying to do is to send any calls starting with 9001 out through

Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi

Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the

Dear All, I'm having a problem with my Asterisk + E100P Installation in UK (BT PRI). The system functions as expected, and my dial plan works as expected. 30 minutes (or so) after starting the asterisk service I lose the PRI line, and only get this back after a service asterisk restart or reboot. During the failure there is no alarm on zttool, ztcfg show all 31 lines and there are no

Hi all, I'm new to this list, so forgive me if I'm not supposed to ask this: I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there any way to use TLS with this phone<--->asterisk (v 1.6.0.9)? It is said that is supports TLS/SRTP but I don't see any of these options in the configuration file or the admin (advanced) SIP conf panel. Am I missing something? Thnx

I have an a bunch of SPA941 Linksys phones for users in and out of the office. When the phones are in the office (and on the same network as the asterisk server) the WMI goes on when it should and off when it should. But when the phone is on another network and natted it fails to do so. The light always stays off. Has anybody had a similar problem (and hopefully a resolve)?

I have to agree. A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system. I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. There is always MSN. Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Netural Bay