similar to: bypass menu for certain numbers?

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

Hello, I have a rather odd problem with Asterisk detecting faxes. I have two POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2 is fof fax. When I set them up with channel => 1-2 in zapata.conf, all is fine, but as soon as I have two channel => definitions, Asterisk is unable to detect faxes. The fax line is not supposed to ring local phones, so the most obvious

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

1. It would be very helpful if the voicemail module included the caller-ID info (name and number) in the "Subject" line. Otherwise, looking at the list of messages is like looking at an INBOX without a "From" column. I patched app_voicemail.c quickly, but it would be even better if this were a configuration option (since some installations don't have caller-ID

Is it possible in any way to use an Outlook contacts database as the source for the internal Asterisk database that is used for callerid lookups? Thanks!

Most of the time i can find answers to my questions on the wiki, google, or searching the list now i am stuck . I have a small * box at my house running 1.0.9 stable and a devlite kit. Every thing is awesome VM, IVR, Echo canceling, and Meetme are all working great. But on Incoming caller id i need to add a 9 as a prefix to make it easier to return call from my cordless phone (cheap vtech

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.

Could someone with inside knowledge comment on that? If the source code says "deprecated" but the CLI help does not mention that - whom do I trust? -------- Original message -------- Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions From: Philipp Kempgen <philipp.kempgen@amooma.de> Thomas Kenyon wrote: > Philipp Kempgen wrote: >> You might use

I have found that I can make the phones display any one word on this second line by adding a fromuser=<word> in sip.conf. This really isn't good enough though. When you look at the received calls or missed calls directory, each item has two lines, the first is the CID name, and the 2nd is supposed to be the CID number. However, if it is asterisk, or some other word, when you hit the

Hello, I am trying to use * to handle anonymous ISDN callers. Something like exten => 5150/0,1,Congestion should work but doesn't. Apparently because the ISDN CAPI doesn't use 0 for callers who don't send their number. Is there a way to make * identify ISDN callers who use CLIR? -Walter -- Walter Doerr =*= wd@infodn.rmi.de =*= FAX: +49 2421 962001

The subject says it all. A couple of my sons have very annoying friends that tend to call ALOT. I usually don't like to answer the phone but these kids keep calling back with in 2 minutes of calling. I'm sure someone else has this problem and maybe using * to do a callerID match and block? Even add logic that if they called so many times in an hour? Or in my case, make it a

Hello When using LookupCIDName, Asterisk 1.4 says that it's deprecated, and we should use "${DB(cidname/${CALLERID(num)})}" instead, but I don't know how to use it: ;DEPRECATED exten => s,1,LookupCIDName ;ERROR exten => s,1,${DB(cidname/${CALLERID(num)})} I guess I should use this as a parameter to a function, but which one? Thank you.

Is there any way to take an incoming callerid string and remove the given "name" part of it and replace it w/ something arbitrary, or add to a blank name string (possibly by looking up the number in a database)? Thanks, John Lawler

I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a "call forward number" into a customer record. That way, I could automatically route calls to extensions to a forwarded

Hi Guys I have just installed a call center onto Suse 10. I have managed to do a DBget (astdb) and extract the DNID numbers to play a DNID specific greeting. We have installed Snom 320 and the customer would like us to Send the DNID(nam) to the phone screens so that the agent will be able to answer in the correct language and with the specific customer company name (ie. Agent says

I have been bugging by a telemarketer who does not take any cue at all. So I look up the Asterisk Handbook and send his call with the respect caller id to my voicemail. Has any one implemented any of this feature with database for more caller ids to be included?? David Kwok

Hi, Some background: I would like to call my girlfriend over the internet. We are both behind a nat router and I want to avoid portmapping. I've heard that you can call someone behind a firewall (nat router) with the IAX protocol, but I'm not sure. The questions: Do I have to set up my own PBX asterisk server or are there any other (free) servers where I can register on and connect

[This email is either empty or too large to be displayed at this time]

I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------