similar to: Problem with incominglimit and outgoinglimit

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. When I check the

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

Hello, I'd like to be able to see if a channel is use and handle the call differently if it is. The best I can find is the command ChanIsAvail(). The problem is, I have an snom200 phone which does call waiting, so even if it is engaged in a call, a second channel is still available on it. I would like to be able to differentiate between these two cases: no calls engages, or calls

Hello. I was hoping someone might be able to help me with the following problem: When an incoming call comes into to * I would like it to attempt to find the first extension in a group of extensions that isn't busy and to send the call to that extension. Should that extension not be picked up or if the phone at that extension is on DND I would then like * to send the call to voice mail

Hello all. Has anyone had any success using ChanIsAvail with only SIP channels? Is there another, better way to check if an extension is busy without dialing it? Thanks, B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040109/48ac2c3e/attachment.htm

Hello, Since outgoinglimit is EOL'd, I've implemented SetGroup/GetGroupCount to ensure that SIP clients will only have a single call at any time. Works perfectly for simple calls using Dial(). I'm now struggling to find a way to similarily limit 2nd calls to SIP clients that are Agents, who receive their calls from a Queue(). Is there any way to accomplish this (without writing

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,

Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com).

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

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.

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 am trying to use ChanIsAvail to decide if a particular extension is available in the sip channel I am using MySQL to hold my SIP friends. and wy cvs version shows Asterisk CVS-08/02/04 my intention is, that if the extension is not available in Sip channel, I will send the call somewhere else my extensions file contains the following: exten => _[123]XX,1,ChanIsAvail(sip/${EXTEN}) exten

Hello list. ChanIsAvail returns 20 for ${AVAILSTATUS}. What does this '20' mean ?? ... exten => 1,n,ChanIsAvail(SIP/sin10) exten => 1,n,NoOp(chanisavail == ${AVAILSTATUS}) ... [Mar 4 15:10:16] -- Executing [1 at sin:7] ChanIsAvail("IAX2/testlocal-14088", "SIP/sin10") in new stack [Mar 4 15:10:16] -- Executing [1 at sin:8]

Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro. The logic is simple; try Zap/g1 (a group of two E1s), and if that fails, try locating a channel via DUNDi. Here's a massively cut down version to illustrate the problem I'm having. macro dialout ( dest ) { ChanIsAvail(Zap/g1); noop(Value of AVAILCHAN is ${AVAILCHAN});

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

Is there a way to trigger an action when a certain number of zap channels are in use, or is there a variable that stores max used channels that can be read? I use PRI for inbound calls, but outbound goes out via SIP, so the simple solution does not work. I need to know when the potential exists for inbound calls via PRI/Wildcard to be blocked because there are no more channels. Obviously

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

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 list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)