similar to: Rules about congestion

I am trying to get called back with a DISA dial tone when I call a trigger number. I got it to work almost the way I want, this is the callback context: [callback] exten=> 501,1,Congestion() exten=> 501,2,Hangup() exten =>h,1,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) exten =>h,2,Hangup() With the above, the call comes into the trigger number, then the call

I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card and an OpenVox A1200P card. Up to today everything was working perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo ports are used for a GSM adapter and for an ATA connected to Vonage. The problem we started noticing today was that the Vonage line will receive a call and then cannot connect to any of the SIP

I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could

Hi, I am trying to implement 911 funtionality in my PBX. A call should drop if all lines are busy. Here is my context nineoneone from extensions.conf [nineoneone] exten => s,1,Set(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,Set(EMERGENCY=1,g) exten => s,n,Set(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})

I'm creating extensions _*XX. But whenever I press *0 or *8, Asterisk throws out congestion and hangs up. I set verbose to 6 and debug to 6, but all Asterisk cares to display in console is -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' Are these combinations forbidden? Yuan Liu

I don't have a particular reason to upgrade, but I'm installing a new box, so I have the opportunity to go 1.4. On the other hand, I'm not familiar with 1.4, and relatively new to Asterisk. So instead of trying to keep up with two different versions, I want to tie my handful of boxes to one, before any of them grow too complex. Is there a document about the main motivations to

Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will appreciate your valuable help. Thanks Smir

I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the topology: VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension | FXO ___ PSTN extension When I call a VoIP extension on that box (from a VoIP extension), voice is good. But when this box tries to bridge the call with a

Hi, I'm using an half T1 line on a asterisk (obviously :)) 1.6.2.4 system, up to recently everything was fine but we are starting to experience the call limitation of the line (15). So as to warn user of the problem i attached a vocal notification to the CONGESTION status after a Dial(), but it looks like it also catch other congestion case (maybe on the receiver side). Should i / Could i

Is there a recommended way to find out the cause of DIALSTATUS = CONGESTION for PRI/BRI channels? Currently I am evaluating the DIALSTATUS variable and I also count the active ISDN channels for the ISDN trunk in question. Counting the active ISDN channels seems somewhat clumsy as the mapping to a specific trunk must be done by hand (or write even more code). I have a setup where outgoing calls

I'm trying to add an emergency dial to my context. However, when I try to dial it, I get caught in an endless loop. For debugging, I have pared out nearly all the control flow and just have ChanIsAvail() and Dial() called. Using two different extensions to call teh same number, I get two different actions by *. Here is the vvverbose output: -- Starting simple switch on

Hi, Yesterday I updated asterisk to the latest zaptel driver and today my congestion problems are gone... (see http://bugs.digium.com/view.php?id=6509), only to be replaced by: Feb 17 10:02:37 DEBUG[19225] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 26 Feb 17 10:03:08 DEBUG[19274] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 216 Feb

I read the wiki and looked at the examples, but I'm still having problems. I have a Digium 4 port card with POTS lines plugged into all four ports. How do I play the congestion tone the the caller when they try and dial out but all the lines are in use? should something like this work? [dial-trunklocal] ; Local calls ignorepat => 9 exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1}) exten

When I use DISA I get congestion when I try to reach 1-800-number: Here is the context: [disa] exten => 087,1,Answer exten => 087,2,DigitTimeout,8 exten => 087,3,ResponseTimeout,20 exten => 087,4,Authenticate(985) exten => 087,5,DISA(951|disa-access) [disa-access] include => tollfree include => outgoing-voipjet [tollfree] ; ; terminate toll-free no.'s via fwdnet ; US

I have an * installation which will refuse incoming callers once a max (5 callers) is reached. Caller 6 and up should be notified of congestion...without network load on my trunk. How would I do this? The voipinfo wiki shows playing a congestion tone to the caller, but that seems stupid since I'm consuming bandwidth to send a tone. I also tried just responding with the congestion

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Can the Dial() command tell the difference between busy and congestion? At the moment it seems to be treating them both the same on my server. I want to route the calls out via a SIP gateway unless that is congested, in which case dial out through my POTS line (using an X100P). It seems a bit pointless to try dialling the POTS line when the SIP

The SM56 MODEM manual says it does. But when used with zaptel 1.2.12, nothing shows up. Yuan Liu

Hi, I'm trying to compile ocfs2 1.2.6 on a 2.6.21 kernel (with rsbac and pax patches), but I can't get this to work .. In 2.6.20 there was an change in the definition of the INIT_WORK macro (http://lkml.org/lkml/2006/12/5/269) this seems to cause my problems (see below) but even after removing the third parameter of the INIT_WORK calls the compile fails (see second compile failure). Can

Hi, I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and try to make a call I get the following error message: ======================================================================================== -- Executing [6781948 at default:1] Dial("IAX2/iaxy-7477", "DAHDI/g1/96781948") in new stack [Apr 25 13:00:10] WARNING[3772]: app_dial.c:1806

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb