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I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

...arketer screening system. But I am not sure of the syntax here, but don't want to add another line for each 8xx toll free npa either. Looking for suggestion on syntax here and not even sure how to pattern match on only the area code(first three digits) here. Thanks for any suggestions! Lyle Giese LCR Computer Services, Inc.

...nat=never dtmfmode=rfc2833 Otherwise, I stayed with the standard Uniden provided config files served up via tftp and only made the minimum required changes to config files in Asterisk. I am running firmware 4.77(also tried downgrading firmware on phones to 4.63). Any suggestions? Thanks, Lyle Giese

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is "dropped" until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to

Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here

I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) But

The orginal did not make it to the list... Spam filter issue??? No repeat of the lockup yet. Lyle -------- Original Message -------- Subject: voicemail locked up Asterisk 1.4.13 Date: Thu, 01 Nov 2007 20:57:27 -0500 From: Lyle Giese <lyle at lcrcomputer.net> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> I am running Asterisk 1.4.13 with libpri 1.4.2 and zaptel 1.4.6 on openSuSE 10.2 (64bit kernel) with an AMD dual core 64 bit processor at 2ghz and 1g of ram. M...

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

...at > ? ? ? ?asterisk-users-owner at lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > ? 1. Re: use dahdi for local terminal modem access? (Lyle Giese) > ? 2. dialplan pattern help (Armand Fumal) > ? 3. Re: Securing Asterisk - How to avoid sending, "SIP/2.0 603 > ? ? ?Declined" (Patrick Lists) > ? 4. Re: Securing Asterisk - How to avoid sending, "SIP/2.0 603 > ? ? ?Declined" (Paul Belanger) > > > ------...

I am setting up my * for at home office and still have analog phones attached and answer from those analog phones and not necessarily through the pbx. I found that with the X100P cards, they see the 2nd ring and will be ready to answer the line. I used a Wait to pause and allow another 2 rings before * answers. But found that if we answer the line after the 2nd ring and before the 4th, * still

I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x "restart gracefully" However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks

Hello everyone.... Searching the archives and google always comes up with entries regarding the "dyn" dns option in the 7960, but I can't find answers to my specific question.... My 7960 is connected via cable modem and is NAT'ed (everything is working fine). On the 7960 under SIP configuration\NAT Address I have the public IP of my cable connection. Comcast gives me a

Not even sure how important this is considering the state of many of the online docs... I have doxygen installed as is noted for the requirements for 'make progdocs', but the make doesn't find dot. I have no idea where dot went, is or should have been... I am installing und Suse 9.0 and it's rough. If you forget something duringthe initial install, adding the package later

I am putting together my first *. I had it running with two other pc's running xlite and setup voicemail and a couple of menus and submenus and had that running well. I had order a couple of oem x100p cards from digitnetworks. I installed them as they said with their voicepet2.2.zip drivers and did the modprobe on zaptel and wcfxo and then ran ztcfg -vv and got this: Zaptel Configuration

I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in

...t horwits.co.uk> > Subject: Re: [asterisk-users] Telemarketer Torture.... > To: asterisk-users at lists.digium.com > Message-ID: <20080316143700.2e1952af.g.stewart at horwits.co.uk> > Content-Type: text/plain; charset=US-ASCII > > On Sun, 16 Mar 2008 08:50:50 -0500, Lyle Giese <lyle at lcrcomputer.net> > wrote: > > > I just forward them to one of those two extensions. If callerid > worked > > more reliably I would automate it. But I get a lot of caller id > failures > > on my incoming POTS lines, esp when calling in from my cell phone...

Hi Which is the best interface card to connect PSTN line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj -------------- next part -------------- An HTML attachment was scrubbed... URL:

If anyone has any info on this it'd be much appreciated - haven't found much about this topic anywhere. We are setting up sip probe monitor to make sure that our Asterisk boxes are up and functional (or at least responding to the sip protocol) and we need to determine the appropriate probe syntax for the probe requests to the Asterisk boxes. These boxes are running on various platforms and