similar to: Advanced Ring All Hunt Group

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)

Calling from PSTN let extension 601 ring twice, hang up and starts over again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SIP/102,20) exten => s,3,Voicemail(u102) exten =>

The problem I am running into is that when the call to my cellphone is made, it appears as though the call "completes" so it never rolls to asterisk voicemail. Here is my current config: exten => 102,1,Dial(${sipura},10,) exten => 102,n,playback(pls-wait-connect-call) exten => 102,n,Dial(IAX2/asterisk1/9139275900,10,r) exten => 102,n,VoiceMail(u102@default) exten =>

Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hi all. I am new to the list and new to asterisk. I have asterisk installed and running. I am using it as a voicemail server only. What I would like to do is send users to a general mailbox that will be addressed as <companyname>@asterisk and give them the option to wait for the tone and leave a message, or press 9 to dial by name. My questions are: 1. What is the best way to do

Hi, I am trying to use the attended transfer. So I put this in my feature.conf: [general] [featuremap] atxfer => *0 blindxfer => #0 I completly restart asterik, and not just make a RELOAD. But during a call, when I press # it runs a blind transfer and if I press * I am disconnected. I am using the CVS version of * get as explain here

Hello again dear list. Could you please help with this? Thank you for all support, you are great, and i am now at a late stage in the setup and tweaking this server, So I hope you can help me again. I Can't make include the context nighttime. Just to demonstrate if it works, I have a playback function there. But CLI reports: CLI [Jun 18 14:20:22] WARNING[2287]: pbx.c:9542

I manage the PBX system for a medium sized call center. Where all calls are distributed via a few call Queues. However I am having an issue where reps are being distributed calls regardless of wether they are on a call. I have looked into using SetGroup but I don't think this works with Call Queues. I have also looked into incomingcalllimit and that seems to no longer work. Any sugestions?

Folks, Apologies ahead of time if this has already been asked (read the list for the last month looking for something similar). I have been trying to get the Background command to work with no joy yet. Here is what I am trying to do: 1. Answer the call. 2. Play the message in the background, while waiting on DTMF from user. 3. If I get a "1", then interrupt the message and dial the

I've been beating myself up over this script but clearly I'm missing something. If I enter an extension like 101 it rings through fine, but if I pick 2 for sales it hangs up with this message: == Spawn extension (sales, s, 1) exited non-zero on `Zap/1-1' Since I'm not sure what that exacly means I cannot take appropriate action. Any help would be appreciated. [default]

Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum

how would one dial multiple multiline sip phones (cisco 7960) and making sure that all the phones ring on the next available line appearance? I'm currently using the local channel to accomplish this but I'm having some trouble. Here is the configs: each cisco 7960 phone has six registrations in sip.conf, 1XX1 thru 1XX6, normaly when an extension is dialed the following happens: exten

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

Paddy, Why not use local extensions? You can do something like this. Exten => s,1,Dial(Local/set1 at call_all&Local/set2 at call_all &Local/set3 at call_all) [call_all] Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 Exten =>

Hello fellow Samba users, I have a problem setting permissions using smbmount on my Linux /GNU computer I have read/write access rights on NT Server computers, and on a Solaris 2.5 machine, running Samba. My Linux computer has Redhat 5.0, and runs kernel revision 2.0.33. SMBFS support is installed as a loadable kernel module. The smbfs client software is version 2.0.1, and was installed with

Apologies if this is a repeat: I trawled through the archives and couldn't find a reasonable answer, so I'm asking here. I have an Asterisk install connecting from behind a NAT device (DSL modem) to a SIP proxy (in my case, Broadvoice). I have an sjphone softphone on a Windows PC also behind the NAT device that connects to the Asterisk install, and using this setup I've been pretty

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

Hi, I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone. Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution. Here I am sending my configuration file values: Contents of