search for: incominglimit

Hi all, I have some peers configured in SIP.CONF file with parameters incominglimit and outgoinglimit set up to 10. By doing that, I expect that this peer will not be allowed to handle more than 10 incoming calls and 10 outgoing calls at the same time. However, since I upgraded to Asterisk 1.2.17, I started to face a problem. Sometimes, calls to those peers are not connected. Whe...

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...

...oned 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 can't remember your name) with regard to "incominglimit" and "outgoinglimit" to prevent that horrible call waiting in your ear for Grand Stream phones. It worked only if you received a call and only once. I've tested this on my system between GS and X-ten, using normal extension and queue calling, and it seems to work ok for me. No c...

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.

For those that don't read every line of source code here's something I found out today... -------- Deprecated incominglimit and outgoinglimit Incominglimit = number of calls the local extension can originate to Asterisk. Outgoinglimit = number of calls Asterisk will terminate to local extension. End of Life for these commands announced**, please use setgroup and checkgroup, that will also be helpful with cross channel...

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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-use...

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...

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

...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 qualifyfreq=300 insecure=port,invite nat=yes outgoinglimit=4 incominglimit=4 [mypeer](peer) host=10.0.138.226 defaultuser=2155551941 fromuser=2155551941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IP (tos 0x0, ttl 64, id 9242, offset 0, flags [none], proto UDP (17), length 555) 10.0.1.3.5060 > 10.0.138.226.5060: SIP, length: 527 OPTIONS sip...

...nd call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I would like to limit the number of calls sent to each phone to 1 call only; otherwise respond as being busy. I have looked at trying to accomplish this in the sip.conf by using the 'incominglimit' and 'outgoinglimit' parameters, however, the only one that *seems* to work is the 'incominglimit'. This prevents further calls from reaching the phones, rings busy, but does not allow our phones to initiate a 2nd call OR transfer their existing call. The 'outgoinglimit'...

...ddr=0.0.0.0 trunkdfreq=40 amaflags=billing disallow=all allow=gsm allow=g729 ;allow=g723.1 ;allow=ulaw tos=lowdelay jitterbuffer=yes [cornjundiai] type=friend username=corncea context=default secret=spojundiai host=133.1.150.1 peercontext=default qualify=yes trunk=yes insecure=very canreinvite=no incominglimit=30 disallow=all allow=gsm ;allow=g729 ;allow=g723.1 ;allow=ulaw [gatewaycea] type=friend username=sipcea context=default secret=ceaiax host=200.XXX.XXX.XXX peercontext=default qualify=yes trunk=yes insecure=very canreinvite=no incominglimit=30 disallow=all allow=gsm ;allow=g729 ;allow=g729 ;allow=...

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". Also you don't need to limit it to one, just having a call-limit at all works. (Tried with call-...

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 consultative transfer using sip refer commands) I could start to use the agents app with agentcallbacklogin to (almost) emulate the current behaviour and use app_groupcount - I can automate the login usin...

Hi All Has anyone managed to get the hint priority with Swissvoice IP10S phones working? I have 2 phones: a Snom 360, setup as the reception phone on extension 11, and a Swissvoice IP10S on extension 12. When calling each other (tested both ways) I can only ever see the Snom 360 in the Active State from 'show hints'. The Swissvoice stubbornly remains in the Idle State when on a 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...

...nd 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 host=192.168.1.210 context=outbound incominglimit=2 disallow=all allow=g723.1 However, when asterisk receives call from this box, it does not send it to context "outbound" b ut the "general" context and call fails because it does not have instructions in this context. Did anyone successfully get incoming h323 call? What is...

...on and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten => s,1,Dial(SIP/5000&SIP5001&SIP5002&SIP5003,20,tr) exten => s,2,Dial(SIP/5004&SIP5005&SIP5006&SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Robe...

...long should be at least 13 chars long amaflag- Categorization for CDR records. Choices are default, omit, billing, documentation and of course Null though not stated. >>> canreinvite this looks to be 3 chars not 1 (Null, no, yes) >>> context See name above >>> incominglimit/outgoinglimit these have been depreciated and probably should be removed unless there is come Realtime coding that requires these fields to be present. >>> restrictcid currently 1 char long, should be 3 chars for values (Null, no, yes) >>> pickupgroup Since callgroup was s...

...6004,6004@internal device => ATA0016c75645ec ; This is an example config with multiple phones ; 08/2005 Stefan Gofferje [lines] id = 6000 pin = 1234 label = 6000 description = Office context = client_int_unrestricted callwaiting = 1 incominglimit = 2 mailbox = 1000 vmnum = 8500 cid_name = Office cid_num = 6000 line => 6000 id = 6001 pin = 1234 label = 6001 description = LivingRoom context = client_int_unrestricted callwaiting = 1 incominglimit = 2...