similar to: SIP Caller ID Number vs. Caller ID Name

On Tue, 15 Feb 2005 03:02:45 +0100, Stefan Gofferje <stefan@gofferje.homelinux.org> wrote: > Hi folks, > > I have some trouble with the FOP and would appreciate if anyone could > point me into the right direction. There is a FOP user list, although not too active. http://www.asternic.org/ > Is there a way to define a button like Zap/g1/6000 and have it light up > when

I am thinking of having a pots line with multiple numbers on it, and having Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring another desk if the person called xxx-xxx-xxx2, etc. Can Asterisk do this? -- Jeff Ramsey MIS Administrator Tubafor Mill, Inc.

This is super easy to do. All you need to do is to put that announcement in a MP3 and set a musiconhold class for that incoming Zap channel. Then basically when ever that PSTN number rings, Asterisk will play the MP3 stream "Your call may be monitored or recorded, please hangup if you do not agree...etc" in a loop until the line is answered. Caller doesn't pay a single dime to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

Hi, We have to servers set up in two different networks. We are able to connect calls via IAX and they work perfectly. We do not see caller ID from clients on either side. Our Grandstream phones say Eri and our XTen phones say Asterisk. We did a debug and I am pasting the output from both servers below. We tried setCallerId in several different ways. We see the value get passed to the

After I made the following changes to get my Win2k machine to be able to mount my Unix disk, now my Win95 machine is not able to mount the Unix disk (using 'map network drive'). It asks me for a password but says the password is invalid, it is the same password that is used on the Win95 and Unix machines... help > -----Original Message----- > From: Barker, Brian W. > Sent:

I have a ALPHA version of my new ping AGI demo available. Access via: IAXTel 1-700-923-3645 or Dial(IAX2/guest@ext.fnords.org) When asked for an extension, enter 2101. This will bring you to the System Services menu. The Cepstral version of the ping is option 28, the Festival version of the ping is option 32. Please report problems and/or issues directly to me. I'm trying to get

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and the From: should be the party doing the attended transfer. This seems like a bug. - Doug.

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line

Okay, now I am really confused. I have two PRI's coming in from two different Carriers (QWEST and ELI), both of them are supposed to be setup to pass name and number on incoming calls. Problem that I am having is that I am not receiving inbound caller id name on either PRI, the only thing that both carriers have in common is that I am terminating into a DMS switch at the carrier.

Hi, This is a message I already posted on the chan_sccp mailing list, but since this list has a lot of active members, I'm hoping someone might be able to help (And my problem is * related, so I guess it's ok if I post it here also ;) ). I'm trying to get SCCP ATA188s to run with Asterisk. The Asterisk box uses the latest Asterisk@Home image (Version 2.6). I have compiled and

I've helped a lot of people on the mailing lists and on IRC #asterisk. and wanted to let people know that I will be in Europe between May 19 and June 21. Stockholm (VON 2005), Brussels (holiday/vacation), Amsterdam (holiday/vacation), and Madrid (Astricon). There are several weeks during my trip that I have no current plans for and may add other cities to my itinerary. I'm looking

Hi I want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Thanks Joseph ___________________________________________________________ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.

Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 => jumps into a context which redials until callresult is not busy Maybe like this: [autoredial] exten => s,1,Set(number=${CHANNEL(lastdialed)}) exten => s,2,Dial(SIP/${number}@account,60,g) exten => s,3,Wait(15) exten

Hi, in case, anyone is interested... I have started compiling a blacklist of hosts and networks from which SIP fraud attempts occur. My criteria currently are: To block an IP: - Minimum 3 attacks within one week from the same IP To block a network: - Attacks from minimum 3 IPs from that network within 2 weeks Common criteria: - Provider does not react to complaints OR - Provider sends autoreply

http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetGroup -Matthew ----- Original Message ----- From: "Stefan Gofferje" <stefan@gofferje.homelinux.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, January 31, 2005 6:43 PM Subject: [Asterisk-Users] Limiting no. of calls on one channel

Hmm. By the way, please don't post bugs to asterisk-dev as I've been told :> That list if for on-going development. That sounds like a bug I encountered in 1.0.5. There is a division by zero bug in chan_iax2.c introduced somewhere after 1.0.4 I believe and currently fixed in HEAD. (They've given me enough shit for posting the bug while it was fixed in HEAD already. No need to

I am impressed, I have been trying this for sometime using the SIP image and the only difference I can create is a 'single' and a 'double' ring on the phone. I use the 'single' ring for phone calls and the 'double' ring for the doorbell. I would love to be able to choose a ring tone based on the incoming msn or callerID. The idea of the phone shouting 'Its the

Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL: