similar to: Using ChanIsAvail and SIP

I am trying to use ChanIsAvail to detect the best route for a call. I am testing by dialing an extension that is then forwarded to the DID. Normally it will be an incoming PSTN call that is forwarded. When I try it, I get put on hold for a few seconds and miss the beginning of the recorded message. Any ideas what is going on? -- Executing ChanIsAvail("SIP/501-304d",

I am trying to construct a macro for long distance dialling. I have two internet feeds, I have all routes including Teliax on Internet A and a static route to Voxee on Internet B. I thought I could use the dialplan entry below which uses the ChanIsAvail() command to check the connection, but this returns the provider but not the username, so I don't understand how to use this for real

I have incoming PSTN lines on an Adtran 750 channel bank. Calls are evaluated by an agi script based on callerid and forwarded to an international DID through Voxee. There is an IVR at that number that asked to user to enter a selection. When the user presses a key, my pbx puts the call on hold and tries to start music on hold. What's doing this? I have no backgrounds, no listen, the call

I sent this earlier under "Editting variable contents" but no-one has responded. So, the subject is now more to the problem, instead of the solution I was trying to implement. ChanIsAvail returns the channel ID plus "-<session>". How can I edit ${AVAILCHAN} to remove this session ID, so I can use its contents in a subsequent Dial statement? Dialing on Zap just gives a

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 I am trying to setup ChanIsAvail function in the extensions.conf file so that user should use the available channel to call out, but immediately after the function like, zap channel hangup. Here is the copy of my extensions.conf file and messages display on consol while making the call. Please help me to fingure out this issue. Thanks Deepak Extension.conf : exten =>

I'm testing ChanIsAvail context and it is not working for me. exten => 55,1,ChanIsAvail(SIP/11&SIP/21) exten => 55,2,Cut(theChannel=AVAILCHAN,,1) exten => 55,3,Dial(${theChannel},r) exten => 55,4,Hangup exten => 55,102,Goto(s,4) It is not dialing SIP/21 when I'm talking on SIP/11, it execute Hangup instruction instruction. According to notes: The channels are checked

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,

One might also conclude that during the outage the support people were focusing on getting the system back up and were not near phones. At least that is what I would bet on. Just a thought considering how most of the smaller ITSPs seem to work. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Hi, So I've done my research on Chanisavail, read the wiki, checked the archive but can't seem to find anything to suit my scenario. I've played around with it a lot, but I'm still scratching my head on what I need to do. What I need is to be able to accept a call by SIP and ring all telephones that are not in use (which just so happen to be on Zap interfaces, but might be SIP

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 am copying the Master.csv file to another server and importing to mysql. I am looking for a simple billing application that will produce a bill for a give account code for a give period, based on a rate table. Is this available? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM:

Hello all, I am trying to find out if anyone has a provider that is good with dtmf playback using a Sipura 2100? I've just dialed with voxee and the call goes through but when I press 1 my dialer does not " hear" it. My dialer is making the call using a Dialogic d/4PCI connected to the Sipura 2100 through voxee and I am calling my landline. When I pick up the landline

What is the function of the snare daemon? I just found it on a machine and never heard of it before. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla at yahoo.com -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be

I have a Astlinux system that can't run mysql. I want to bill calls by extension but all I have is the raw CDR files. Is there any software that cal produce a simple call accounting from text CDR files? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com --

I need to make changes to the kernel for a firewall application I am running. Can anyone point me to a howto on downloading the rpm, building the source, patching the source, and building the new kernel rpm? I am a bit confused by changes in teh way things are done and not sure of the correct procedure. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax:

Where could I get a source rpm for this kernel version? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla at yahoo.com -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Is there a way to configure Apache and WebDAV so that uploading requires a password but anyone can read from the directory using a browser? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla at yahoo.com -- This message has been scanned for viruses and dangerous content by

Hi, all. I'm making my first foray into AEL. I'm starting with a simple macro, but I've already encountered an odd behaviour. I'm wondering if someone can shed some insight: Asterisk 1.2.9.1 macro newPlaceCallPSTN { s => { TIMEOUT(absolute)=7200; NoOp(Analog Out List: ${ANALOGOUT}); ChanIsAvail(${ANALOGOUT}); NoOp(Available Out List:

I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband