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Is there a way to match the last 7 digits of an extension? So that 1008014445454 8014445454 4445454 Would all match? I have looked at extension matching and I can't figure out how to do this:-) Thanks in advance! Chris -------------- next part -------------- An HTML attachment was scrubbed... URL:

...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 retrieve next Via, don't know where to send responseSIP/2.0 180 Ringing From: "MODESITT,CHRIS " <sip:8013793000@200.200.200.200:5060;user=phone>;tag=4fdc9d0e-1e600f94-ed7e6 23f To: <sip:8014377860@67.137.28.10:5060;user=phone>;tag=as4fc8aa8a Call-ID: 3a8530f4-43cb1-1e600f94@200.200.200.200 CSeq: 5466974 INVITE User-Agent: Asterisk PBX I still can make outbound...

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

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.

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

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

> -----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 that I need > to but in some echo cancellers on my PRI's....

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

Is it possible to run NETHDLC and PRI signaling on the same 4 port T405P card? This is what I get when I try to start asterisk: == Parsing '/etc/asterisk/zapata.conf': Found Mar 24 10:48:53 ERROR[2368]: chan_zap.c:9435 setup_zap: Unknown signalling method 'pri_cpe' Mar 24 10:48:53 WARNING[2368]: chan_zap.c:9614 setup_zap: Ignoring switchtype Mar 24 10:48:53

Asterisk 1.09 works well with it however CVS head has major issues, wondering if anybody else has seen any RTP like issues and what they did to work around it. Thanks Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050917/fa2eda60/attachment.htm

I am attempting to unify how numbers come to me from a specific T1, this T1 acts as an ingress for about 4000 DIDS. About 98% of those DIDS come in as a 10-digit DNIS, what I would like to do is have asterisk log when a number comes in 7 or 11 digit so I can contact my upstream provider and have them translate those DIDS to 10 digit. I think I can achieve this by modifying my dial-plan I am