similar to: Re: CallerID name inbound from PRI

I switched PRI vendors recently, and one of my questions was "do you provide caller ID name in addition to number?" AT&T Local did not, But XO communications said they did. Before I call to complain, is there an setting to turn this on in asterisk? I want to make sure that I have my side covered before I call XO. My current zaptel.conf is: context=from-pstn switchtype=national

Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the

XO CAN supply callerid NAME on a NI2 PRI connection. We have three of them and they work great. Its takes a little doing to get to someone at XO that knows what they are doing but XO does have some VERY good tech support people that know how to get things done. It just takes a bit of work to find them. Outgoing CNAM is a different beast however. They can't take it via IE. You need to get

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

Reason I ask is I may have a non-voice T-1 replacement project going on and I'm investigating my various options. Costs may be about the same for turn-key and DIY. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, February

Define non-Voice T1 porject? You do know that TDMoE does not travel over long distances, You can not route or otherwise take it off of a single ethernet segment. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Mike Hammett > Sent: Thursday, February 09, 2006 1:20 AM > To:

I have an issue with call parking and hope there is some undocumented feature for this. ;-) We are replacing our legacy PBX with asterisk, but to save money over time (handsets and network), I am trying to maintain the use of our legacy PBX. Asterisk extensions can not use the call parking features (not usable over trunk cards) of the old PBX, so I have to get the old PBX to use asterisk's.

>From WIKI: D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) When I use the D option to send a call to my paging system and pick a zone, the Tone is too early. I have tried the 'w' option, but it does not appear to work. No matter how many 'w's

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added

I''ve been kicking this around for months and I thought I would finally ask the list and see if I can get some opinions. I want a database table that holds configuration options for my system. Think exmail server address, port, maybe a path for a banner image, things liike that. The way i''ve implemented it is a single column, each column is a value. This maps very nicely to a

hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud "i use elastix" Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

Announce option for meetme - is it used? It makes a caller record their name, but I do not see where this name recording is ever used. ? Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Fax. 248-836-5101 www.hirotecamerica.com Board member of www.glimasoutheast.org Our company name has changed to HIROTEC AMERICA www.hirotecamerica.com Please update any

My Dell 2800 Dual 64bit Proc. machine came in with hyperthreading enabled. (they call it virtual processor??) I have been intending for a month to disable it. Tonight, I rebooted, turned it off, and let the system come up. zaptel loaded and asterisk loaded, but both of my t1s were red. (it is a TE411P) /proc/interrupts looked OK, zttest gave OK numbers. I doublechecked all of the files in case

Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject:

hot to asterisk multiport...??? example 5060, 5061, 5080 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070815/8a515309/attachment.htm

I will be on vacation from <22/06/06> to <30/06/06>. I will not be reachable on my mobile. I will have limited access to mails, and please expect a delayed response. In my absence, please contact the following: Ray Richard or Safeer Mohammed Thanks H.Gireesh

change asterisk.conf: mkdir /var/run/asterisk chown it to your asterisk user. change astrundir => /var/run to astrundir => /var/run/asterisk My guess would be that you are running asterisk as a non-root user and that this user can not write to /var/run . if so, the ctl and PID files are not created. -- -- Steven http://www.glimasoutheast.org "Tom Castleman"

Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. >You have to compile and install Zaptel first, for asterisk to build meetme. > >-- >-- >Steven > >http://www.glimasoutheast.org > > > >"??" <lijun820311@163.com> wrote in message news:45C1B35E.0037E8.32263@m5-81.163.com... >> asterisk-users@lists.digium.com

Bill Gibbs,hello Thank you so much. According to this method , I get the "app_meetme.so" . ======= 2007-02-01 22:49:43 ????????======= >Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. > >-----Original Message----- >From: