similar to: Extension Matching.

I have a question about how Asterisk Parses the Dial Plan. To create a hunt-group which would be the appropriate dial plan: [CompanyABC] exten => 7228888,1,Dial(SIP/8017228888,60,r) exten => 7228888,102,Dial(SIP/8014361234,60,r) exten => 7228888,103,Dial(SIP/8014362345,60,r) exten => 7228888,104,Dial(SIP/8014363456,60,r) exten => 7228888,105,Dial(SIP/8014364567,60,r) exten

I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server. When I use Asterisk version 10.0.10 everything works perfectly, however when I use 1.2.4 I lose the ability to receive calls from the PSTN. All I get is the following error in my SIP Proxies error logs: SIPSession::proxyResponseImmediately(): Failed to

Okay, now I am really confused. I have two PRI's coming in from two different Carriers (QWEST and ELI), both of them are supposed to be setup to pass name and number on incoming calls. Problem that I am having is that I am not receiving inbound caller id name on either PRI, the only thing that both carriers have in common is that I am terminating into a DMS switch at the carrier.

Question: I am PBX multi-hosting several customers on one of my * servers, what the best way to setup call parking to prevent company A from picking up Company B's parked calls ? Any basic examples would be greatly appreciated. Thanks Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Problem: Zap Channels Disappear @ random intervals. (Channels have disappeared on both gateways twice this week). My Setup: I have 2 Dell 1850 Power Edge Servers - Configured as. P4 2.8Ghz 512 ECC Memory SCSI Array (2 Drives Mirror) Configuration is really simple the boxes are setup as PSTN termination gateways: (SIP/ZAP) - We are running a SER/Asterisk Hybrid. My asterisk

I am terminating a PRI, NI2 signaling into a PBX (My company's PBX not the PSTN) from an Asterisk server. Caller ID number appears to be transmitted caller id name is not being transmitted is their a compile time flag on libpri that I need to un-comment to enable this feature or is this is unsupported for now? Thanks Chris. -------------- next part -------------- An HTML attachment

<http://voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf%20sorting> This page on voip-info.org describes how it is possible to affect the sort order of patterns in extensions.conf. What is doesn't explain is how asterisk really does sort patterns. How does this happen? Adi

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Hi, I am building PBX's for clients. I was thinking of using Dell computers. I was told that they do not work well with asterisk. Any one have any suggestions ? Any other brands that work well with asterisk ? Also any specific hardware to or not to use ? Finally does that Mac Mini work well with asterisk ? Thanks a lot. Dovid -------------- next part -------------- An HTML attachment was

I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the phone

hello, i've an asterisk box which is connected to an E1/PRI via a TE110P card. incoming calls from mobile phones where the number is transfered as a whole block work fine, but when dialing from an analog or ISDN line to the asterisk box there is a timeout of about 3-5 seconds. originally my incoming context looked like: exten => _X.,1,Dial(SIP/${EXTEN}@domain.tld) so i assumed that the

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Chris Modesitt > Sent: Tuesday, June 21, 2005 9:35 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] Best Echo Canceller. > > > I know this is slight OT however I have decided

Does Notes works well with last wine version or should I keep my good ol' 20041019 ? Dripple

Tobez: no disrespect intended, obviously you saw a problem with the master sites for perl 5.8.7 and did what you could to help, and with your position as a maintainer, I know that the trust we have in you and your patches is well earned, so don't take this question as anything but my well-earned paranoia rearing its ugly head: Yes, building perl5.8.7 did seem like it had a lot of problems

Hello, all - I've been throwing the question around on the kickstart-list for the last few days here, and can't quite get ahold of things. Please, allow me to explain. I am in the process of making a custom CentOS/RHEL kickstart install. It works well right now; however, it's a hackjob, and I am not comfortable with it as of yet. For the past few days, I've even gone as far as

Hello, We are looking for commercial solution SS7 with Asterisk. It does not need to be "build-in" with Asterisk. Could anybody suggest something? Thank you in advance. Bart

Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729

Sorry, i also tried this: exten => 6000,1,Answer exten => 6000,2,MusicOnHold(default) and i got this result: *CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack -- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class

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We have an old Telrad 128KSU PBX to which I am trying to connect asterisk in the following manner: Current Setup: Telco-> T1->PBX Desired Setup: Telco-> T1-> Asterisk-> T1-> PBX. I am first trying to setup the Asterisk -> T1->PBX part without disturbing existing setup so I can get asterisk to forward calls to PBX and once that is done, I would try to move the telco