similar to: IAX Extension Dialing Question

Hello, I'm having trouble getting this to work: I have a ring group that dials an extension and if no answer dials a cell phone. If the cell phone doesn't answer I want to go to voicemail or another extension. I have set the timeout to 15 seconds but it never actually works, it will just ring until the cell voice mail picks up. I'm using Asterisk@Home 2.8 and a TDM400P card.

I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question! How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss. I have tried the following

For some reason my voice mail stopped working properly. I was able to go in as a new user, set the password and options and now can never log back in using the password I assigned the mailbox. I can log in through the web interface with that password fine, and the voicemail.conf looks fine but every time I try to check messages I get "Password incorrect please try again" until it

I'm trying to figure out how to allow an extension to register more than once. For instance, I have all of these 4 line IP phones that I use with Asterisk and I would like to have a persons extension (say 101) ring at all four lines so that if the person is on the phone they can take another call, but it appears as though if you try to register the same extension more than once then the most

I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

I have been having a strange issue with my Asterisk 1.2.1 server. I have a TDM400 for the three POTS lines I have and I can receive calls without any problems. But sometimes (not everytime, but 70%) when I dial out of those lines it drops a number and of course gives me the telco error message. How do I know that it's dropping? Because I thought it was a line problem and the tech came out

hello I am running Samba 2.2.8 (latest) that I downloaded the binary from the Samba site. I have this problem in older versions of Samba as well. I have under Redhat 8.0 security level = user works fine under NT/2000/XP platforms, but when I try to login from a Windows 95/98/98se/Me machine, samba prompts me for a password to resource \\netbiosname\IPC$ how do I prevent this from happening in

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

Hello All, I have the latest Asterisk and am using the TDM400 card with 4 FXO ports. One problem I have is that I want one of my Queues to go to a cell phone if all agents for the queue don't answer. The problem is that once the TDM400 goes to a POTS line it does not time out after 15 seconds like I have configured (and like the other agents), it will ring to voicemail which doesn't

I have Asterisk 1.2 and a generic Wildcard single FXO card (a cheapo from eBay). I have read about many people who have used these cards without an issue and I'm just testing to work up a new system. The problem I have is that if I call the telephone number of the line attached to that card and have configured my incoming calls to, say, forward to an extension, then it will ring that

I'm not exactly sure how to phrase this, but I have two offices that I want to each have an * server. One office has two POTS lines coming in, the other has five. I'll call the two line office A and the other office B I want office A to be able to extension dial office B, which I think is straightforward enough because I'll just add a line in extensions.conf that points a particular

ok, I fixed the file permissions problem using chown... which is good. now, I am still having the problem in WIndows 98/Me. samba keeps prompting for a password to: //netbiosname//:IPC$ what in the heck is :IPC$???? it works fine in windows XP or 2000, so what am I doing wrong that it doesn't work in 98/Me? I tried reinstalling windows me on my daughters computer and running the windows

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming

I am trying to get a way to have * forward calls that are dialed to an extension, to end up at an extension on my old analog phone system. I will have 7 lines coming into * using the new Digium cards via PSTN, and then lines coming from * into the PSTN lines on the analog system. So that if for example someone dials extension 110: The system will call the analog system, the system will assume

Hi, I'm having a problem with the timeout option when dialing a ZAP channel. The goal is to ring a number for 15 seconds, if no one picks up, go to voicemail. The dial command is: exten => s,1,Dial(ZAP/1/6135551111,15) exten => s,2,VoiceMail(u1) exten => s,102,VoiceMail(b1) The call will continue to ring beyond 15 seconds. What's interesting is that a SIP channels does not have

Hi list, I am facing the following problem, i need to make the following scenerio work exten => _900.,1,Dial(Zap/g1/xxxxxxxxxx,25,r) exten => _900.,2,Wait(3) exten => _900.,3,SendDTMF(${EXTEN:1}) I am using PRI-ISDN with T400P card. Searching through archives, i found that we can add some 'w' s to the dial string. I tried that using on both PRI and Analogue channels but it

Hello, I'm trying to figure out how to get Asterisk to dial an extension when a call comes from the outside and contains the extension already. (Somebody wants to call a user of Asterisk with extension "111" from the outside) For example: I've hooked Asterisk to sipgate.de and received a landline phone number (say 0781205237). Now if you dial 0781205237 and and an extension

I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked.... Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten => 107,1,Dial(SIP/102&SIP/107,40|r) exten => 107,2,Voicemail(u102@pstn) exten => 107,3,Hangup exten => 107,102,Voicemail(b102@pstn)