similar to: Call Admission Control

With rtcachefriends = yes in sip.conf, my SIP phone registered to Asterisk Server A cannot dial another SIP phone registered to Asterisk Server B. The error message is: "Cannot create channel of type SIP (Cause 3 - no route to destination)". The two phones _can_ call each other if I set rtcachefriends = no. The common extensions.conf simply uses Dial(SIP/extension) to dial extensions.

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing

I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I

Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find anything like it in the grandstream docs. Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

Hi, I seem to have a problem with chanisavail and the call limits on sip phones(incoming and outgoing) The problem seems to be that chanisavail when trying create to create channels and hanging them up afterwards screw up the current usage limit on the phones. Example with chanisavail: Phone A calls voicemail (usage now 1) Phone B tries to call Phone A and uses ChanIsAvail in the dialplan.

I would like to setup a hunt group, not a group ring, using sip phones. Anyone done this with sip devices? Comments suggestions? I have not had much luck with the outgoinglimit=1, incominglimit=1 stuff that I would need to get busy extinctions to work right, which is why I'm asking on the list.

It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql> describe

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

Hi folks, As none of the SIP softphones that I tested can disable more than one incoming call, I decided to implement it by software ;-) I'm attaching a patch that does it. To make it work, modify your sip.conf file and include callwaiting=[0|1] at the general section, or for each peer that you wish to control. Please note that I haven't tested it too much, and my source tree is quite

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

Not being an asterisk expert, but having been around the block once or twice when it comes to data and the like, I have made some observations based on the examples given on voip-info.org Sip configs. it appears there is an adjustment to be made in the sip_buddies example table: >>> name Although set to 30 characters, I don't see where it is limited in the text file. In theory,

Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a "beep beep" (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a "single line" user agent, like firefly, still

Hello, I am running version 1.4 with realtime support. I've set (for Snom phones 300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the database). - When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem was that when such a phone received a call and did attended transfer it was left "in use" and could not receive new calls. -

i am getting 403 Forbidden message from asterisk when it try to register my user agent. i am basically useing mysql through ODBC. i hvae checked ODBC connecteion with 'ODBC Show' command. ------------------------------------------------------ *CLI> odbc show Name: mysql1 DSN: asteriskdsn Connected: yes *CLI> ------------------------------------------------------ and user is added to

Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed for '204.194.36.138' The telephone LCD says "SIP registation

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim

Hi Akk, If this has been discussed/done then apologies be-4-hand. I did not find it in the Wiki or the Archives. Here's the question. We have incoming PRI lines, all on the same main number. An attendant is supposed to handle all incoming calls. Now, let's say I have a multi-line SIP phone. For argument's sake (and to keep it simple) say I only have two lines. We'll call them

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available