similar to: app_queue and app_groupcount

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

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,

* Read the config sample files! (even if you're an Asterisk guru) ----------------------------------------------------------------- For those of you that have a working installation that you keep using, this is a reminder to check into the configs/ directory of the Asterisk source tree, regardless if you downloaded a tar ball or from CVS. As we add or change features in Asterisk, the sample

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

Hello all, I was wondering what will be the "proper" way to manage BUSY state notification in presence once call-limit, incominglimit and all those settings are gone. I'm using GROUP_COUNT for call limiting in Asterisk 1.4.13 but I have no idea how to set up the settings needed for BUSY notification to work as I want it to. Basically, I want to disable call waiting (this is

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

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

Playing with hints/presence/BLF on asterisk I've made the following "discoveries". 1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says: "If you add incominglimit=1 to your peer in sip.conf, the SIP channel will notify you when that extension is busy." As "incominglimit" is obsolete you can use "call-limit".

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 everybody, I'm running an asterisk box -cvs version since few monthes, updated it middle of may and a last one on thursday (24 june) Since this one, my H323 calls loose they audio, both sides. Calling directly from Gatekeeper is ok, so problem comes from h323 asterisk channel. I saw few people telling about similar problem begining of month, does they solve their problem? I also grab

Hello everyone, I have just noticed a fairly obvious feature that it looks like many people have been looking for... If you have a queue defined with strategy=ringall, members of the queue will still get incoming calls when they are already on a call (call waiting). The only solution that has ever worked is incominglimit=1 in sip.conf. The problem there is that it obviously disables call

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, I am using Polycom 501 and I came across a problem. As soon as I have incominglimit=1 in sip.conf, which is necessary for buddy watching, I cannot transfer calls. On the console it tells me: Call from user '3052' rejected due to usage limit of 1. Can someone please tell me how to get around this problem? (I don't know if this is relevant, but in the phone.cfg file, I have

Hi all, I have agents who are members of more than one queue. When an agent is busy with queue A, he is not considered busy by queue B, and receives call (since his EyeBeam Softphone has 6 channels). Besides that, I use a monitoring tool that connects through the manager interfaces and run "show queues" and "show agents" to know agents statuses. I need Asterisk to consider

Good morning How I use the described commands below to limit the number of simultaneous calls saw voip providers that they can be effected and be received in the trunk in the Freepbx? I verified the commands incominglimit and call-limit as I can use asterisk is version 1.4! It would like to restrict for I number it to four of calls that can be used in one trunk of a voip provider? thanks.

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,

Here's my situation. I have two receptionists that answer incoming lines. Each has a 7960G with 5 incoming lines each. I'm trying to set this up so each line on each phone doesn't utilize call waiting. My problem seems to be that ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always returns cisco1. Here are the sip.conf entries: (mind you,

Hello, I am moving topic from asterisk-dev list to asterisk-users list. Did anyone succeed receive incoming calls in h323 and orient them to right context based on "host" identification? To summarise, I have quintum Gateway sending call to Asterisk box, and I would like to use asterisk as a protocol converter h323 --> sip. in h323.conf, I have [quintum_gw1] type=user

Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3

Hello list, I have a Cisco 7960 with SIP Image 7.1. I can make calls outgoing through Asterisk, but I'm having problems with incoming calls from Asterisk. The phone is on a public IP address, no NAT, no firewall. The phone is registered and shows up in sip show peers. If I place a call to the phone, Asterisk sends invites to the phone in vain, and then gives up. I can use my soft phone