similar to: Dial Command M(x) Option

I have officially engaged in a pissing contest with the local Telco over PRI calling name delivery. The telco publishes their calling name delivery over PRI feature as being bellcore gr-1367-core compliant. The gr-1367-core spec states that the calling name is to be included as a facility IE in the setup message, or sent in a subsequent facility IE message with an indicator in the setup message

David, I guess I'm a little confused here. Are you asking me to provide a "pencil copy" of an LOA for your review? I don't understand why you need an LOA from us. We need an LOA from you to order circuits that will be billed to us that will be attached to your CFA. It was also my understanding that you had an LOA ready to be given to us, which had already been reviewed by

FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching

I am having my colo set up 2 PRI's for my new asterisk implementation. They asked the following... ##SNIP## What type (NI2, NTI, 4ESS, or 5ESS) and whether they want to be USER or NETWORK. If the equipment is flexible, NI2, with us as NETWORK is preferred. ##SNIP## We are using a digium quad span T1 card. What is the recommended setup? Thanks, Chris

Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks,

Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco.

Hello! What kind of sms text messaging capabilities does Asterisk have? I do not know very much about about SMS technology, but I am looking for the following features: 1. mobile SIP devices can send and receive SMS messages 2. Asterisk server be able to accept and send SMS messages through PRI lines and Internet connections. I noticed that Asterisk has an SMS function, but I am not farmiliar

Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of

I have an NI2 PRI connection to a 5ESS switch. I am explicitly told to use NI2 as the protocol, not 5ESS. I am running Asterisk 1.2.4. When making calls to the PSTN, Asterisk sends the calling name as part of the display information element in the SETUP message. I need to be able to send the calling name as part of the facility information element in the SETUP message. I've seen that work

Hi Folks, I'm trying to send CallerID Name information out to the PSTN via a PRI with Bell Canada with no success. With inbound calls (originating from the PSTN) CNAM is received successfully, and we've not had any similar problems with other Telco PRIs, so I'm stumped.

Hi, Does anyone know of any projects that would allow you to use Redis in place of AstDB? By in place of I don't mean for what Asterisk needs but to store values. For instance for CNAM currently we need to use an AGI to connect to redis to pull CNAM. So in place of: Set(CALLERID(name)=${DB(CNAM/${CALLERID(num)})} it would be done with redis for example:

Hi, I'm having trouble configuring and E1 link , I know the E1 is a PRI and the switchtype is 5ess. The problem is that everytime I try to dial I got and error saying that it was unable to open the zap channel . Devices are created in the /dev/zap directory and I can open them with a cat /dev/zap/1. I can also see the channels in /proc/zaptel/1. Thanks in advance [zaptel.conf]

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > brett-asterisk@worldcall.net > Sent: Tuesday, November 30, 2004 2:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] National (US) callerid name > resolution for yourasterisk box >

I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great !

Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows:

-----BEGIN PGP SIGNED MESSAGE----- Does anyone have any working example GR-303 configurations for zaptel and zapata conf? The information available on the wiki as well as in the sample configurations just doesn't seem to be enough to bridge the gap for me. In Zaptel.conf, Do you set up a GR-303 circuit like a PRI with b and d channels or do you set fxo or fxs, ks signalling? How do you

Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office:

Dear all, I have a time serie dataset such as the following with data acquired every 15 minutes: Date Heure Profondeur Temp?rature Salinit? Turbidit? Chloration 1 2012-07-06 08:47:22 -0.144 22.469 0.011 0.000 0 2 2012-07-06 09:02:21 -0.147 22.476 0.011 0.000 0 3 2012-07-06 09:17:21 -0.139 22.498 0.011 19.323 0 4 2012-07-06

Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot

Hello! I am having an odd problem in that when the caller dials extension "2" in a dialplan, the system waits 3 to 4 seconds before proceeding. This doesn't happen when any other other extensions are dialed, including an identical dialplan on other another extension! Is this a bug? Later, Elliot