similar to: dialing multiple extensions

hi all, was wondering if someone could assist with a slight problem i'm having. I have asterisk setup with extensions 101 to 109 and am using xlite, grandstream budgetone, polycom ip500 and a couple of other phones. the problem is: 1. only the xlite extension (107) can receive calls. 2. all extensions can dial into voicemail and get mwi when msgs are received. 3. when dialing a non-xlite

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, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

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

Hello Everyone, I need to setup a dialplan where if a incoming call is rec'd to a number, Asterisk needs to dial several SIP extensions at the same time. The SIP extensions are for Cisco 7960s and each have multiple line appearnces. For example, exten => 9043442342,1,DIAL(SIP/102&SIP/103&SIP/104&SIP/105,,20) exten => 9043442342,1,Voicemail(u102) The issue I have is

Hi world, I have a 10-disk RAID-Z2 system with 4 GB of DDR2 RAM and a 3 GHz Core 2 Duo. It''s exporting ~280 filesystems over NFS to about half a dozen machines. Under some loads (in particular, any attempts to rsync between another machine and this one over SSH), the machine''s load average sometimes goes insane (27+), and it appears to all be in kernel-land (as nothing in

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

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

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 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

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

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

This possibly isn't 100% asterisk related, but I'd like some opinions/feedback... A customer wanted different ring-tones to differentiate external and internal calls. No biggie once I'd worked out that details - they have 100% GXP2000 phones, so adding in the relevant SIP header and altering the phones to suit seems like it's going to be a solution... But I started to look at

That way we can ditch our gem_prime_res_obj implementation. Since ttm absolutely needs the right reservation object all the boilerplate is already there and we just have to wire it up correctly. Note that gem/prime doesn't care when we do this, as long as we do it before the bo is registered and someone can call the handle2fd ioctl on it. Aside: ttm_buffer_object.ttm_resv could probably be

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 =>

Hi all, Quick question. Is there a way to have multiple people have an extension, say 900, to their polycom 501 SIP phones on one of the blue buttons to where when a call comes in, I can have it simul-ring and folks can pick up the line on their phone? I'd like to set up a tech support extension for our techs and have a voice mail box assigned to it as well. Right now I just have

I am looking for clues on how to configure distinctive ring for a PolyCom SoundPoint 300. Does ALERT_INFO apply? If so, how? Thanks, David Koski david.nospham@kosmosisland.com