similar to: Is there a way to get inserted into an LEC's CLIDB?

----------------------------------------------------- "Yeah, we rocked the vote all right. Those little bastards betrayed us again." - Hunter S. Thompson on the 2004 election. ---------- Forwarded message ---------- Date: Tue, 22 Mar 2005 19:16:09 -0800 (PST) From: Matt Klein <mklein@nmedia.net> To: Asterisk Users Mailing List - Non-Commercial Discussion

Does anyone know if there's a service out there to -- for a fee -- inject our DID into the LEC's CLI database so a called party gets our associated name? /rg

I know a SIP client behind a NAT trying to peer with Asterisk behind another NAT is troublesome. Has anyone had any luck doing this by interfacing Asterisk to the WAN using 1:1 NAT translation to give it a public IP while still firewalled? In my instance I'm using m0n0wall, but this is a hardware-neutral question. Thanks. -- Robert Goodyear Brand Up LLC http://www.brand-up.com

Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

Does anyone know how to suppress the "Missed Calls" indication -- perhaps on a per-line basis -- on the 7960 running SIP? Reason: I've configured a group of extensions to ring for inbound calls and it seems pointless to accrue missed calls on those line presentations. /rg

Anyone know how to bypass the CONFIRMATION of the user announcement recording in MeetMe? While I like the "please say your name" to announce a user into a conference, I find it confusing and time consuming to make the user to press 1 to accept a recording they haven't even previewed. I'm not a coder, but I'd be happy to comment out the confirmation loop if someone

TDMoE was it. Thank you!!!! Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Friday, June 24, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Exposing Zap Channels on Server A to be UsedByServer B

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

I've been using wine for several years to run the Lotus Notes client at work and found how to get Notes to launch mozilla and openoffice. So now I can open M$ format files (well, most of them) without having to resort to using windows terminal server or a windows based pc. However, I'm having a problem with acroread 5.0. I can get it to launch, but it claims it can find the file or

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

I googled on this for about an hour and the most relevant hit I got was, of course, the first hit: http://www.sowerbutts.com/linux-mac-mini/#support In it, he indicates that the stock Bluetooth module "should work, but untested" - he doesn't qualify the statement with anything. Has anyone tried chan_bluetooth or even the Bluz stack on a Mini or a G5? If so, under Linux or OSX?

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

Robert, Essentually I want to be able to have Server B dial the extensions connected to server A as well as route calls to the outbound route on Server A. Server B will have little to no knowledge of what is on Server A. I just want it to dump the calls off. For some reason I keep thinking this was a PRI type of thing. Like there was a module that loaded up as a fake PRI that your

We have implemented * in one of our branch offices and recently ran up against a very strange issue. On random occasions, when we would dial out using our trunk lines, we would get a message stating "you do not have to dial a 1 or 0 when calling this number" even if we didn't dial a 1 or 0 in the dial sequence at all. After much troubleshooting, we found users with similar issues

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

I live in New Brunswick Canada. The phone company is Aliant. When you set up business service here, you can go with either analog or digital lines. This isn't a T1 or ISDN. They are talking individual lines direct to handsets that they provide. They offer the digital option with even very small ( 2 - 4) number of lines. What technology could this be? Is there any way to connect such a