similar to: call waiting disable in sip

Hello, Right now I have app queue working with incominglimit=1, there is no call waiting signal, but after a while( like couple of hours) some phones randomly get stuck. The * thinks that they are in use and doesnt ring them, when they are infact not in use. sip show inuse, shows that they are inuse. typing reload on the console resets this and they are again available for working. anybody

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk

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

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

Hello, Anybody has it working with asterisk? Could you share your experience ( good/bad) Thank you -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

When I have an ip hardphone username setup in my sip.conf : [109] type=friend username=ipphone9 secret=bla-la host=dynamic dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info defaultip=172.20.0.139 mailbox=109 ; Mailbox for message waiting indicator callerid=ipphone9 <109> callgroup=1 pickupgroup=1 and this user has a wrong password then calls are denied, but

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

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

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.

Hi, This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents is that when they make an outgoing call they are not regarded as "busy" by asterisk and it sends more calls to the agent if it has call waiting enabled. This behaviour is totally senseless since the whole purouse

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

Let's say you have a 256 Kbps Internet connection and you're using it for voice calls. With mu-law (G.711), each call uses about 80 kbps, so you really can't have more than 3 calls active at one time. Does Asterisk support any kind of Call Admission Control where it would prevent you from originating a call if it would exceed your Internet bandwidth? For example, in this case, ideally,

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 just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension

Hi All, This is the first time I'm submitting a patch, and I hope it fixes more than it breaks. I'm putting it here, since John Todd mentioned a while ago about the heavy load Mark and crew have at Digium (doing such good work), so I thought all of us could test this first, and if ok submit for inclusion in CVS later if appropriate. This is an extension to work done earlier (sorry I

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

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,

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. -