similar to: priority -1

Many built-in applications jump to current priority + 101 upon unfavourable output. Does this apply to macros and AGI's also? Precisely, when an AGI returns non-zero, will Asterisk automatically take the call to +101? And when a macro hits an invalid extension, will Asterisk automatically take the call to +101? (So far the "i" extension could not be reached from within a

I've setup my IAX2 over FWD and it is working I can receive a test call and I can call out. Though I cannot figure out how to dial 1-800 numbers over FWD When I dial 1-800 it hangs up on me. Here is a typical session: Called xxxxx:xxxxxxx@iax2.fwdnet.net/18007425877 -- Call accepted by 65.39.205.121 (format ULAW) -- Format for call is ULAW -- Hungup

Hi John, First, I have not much experience dealing with complex dial plans. But since you asked, thought of some feedback. In my opinion .conf files should be kept as simple as possible. It should provide straight forward and simple manipulations to simple & common applications. If more complex manipulations are necessary, then those scenarios could be built using scripts. Therefore, I think

What am I doing wrong? Goif dialstatus: busy CONGESTION not working. exten => _7NXXXXXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr) exten => _7NXXXXXX,n,GotoIf($[$["${DIALSTATUS}" = "BUSY"] | $["${DIALSTATUS}" = "CONGESTION"]]?line2) exten => _7NXXXXXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr) exten => _7NXXXXXX,n,Hangup() When I try to

I'm testing Teliax tall free number line and I'm experiencing long delay about 1sec. during conversation. When I call myself over FWD the response is normal no delay or cut messages. When I call my number over FWD the is a long delay, welcome message usually cuts off few first words and during conversation my voice arrives after about 1sec. delay. Since, the 800-number is only accessible

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

Is it possible to do nested dial() command on one line, Dial number, wait new seconds, dial another number etc. or dial number and jump to another line to continue dialing. D(ww) doesn't work as it sends DTMF but before the call is bridged, and I need to send numbers after the call is bridged. -- #Joseph

I would like to implement enum lookup in my dial plan but searching for solution / implementation I'm getting confused what is current standard. On some pages I read that the ENUMLOOKUP is not in development anymore and suggesting on using Enumlookup.agi scrip , some are saying that Asterisk 1.2.0 comes with a new powerful ENUMLOOKUP. So there is probably no need to use this script anymore;

I need to execute account number, device number after dialing main number; what is the best solution? Is it possible to pause during dialing. Dial 1-800-number press 1 for English wait 5sec enter device number wait 5sec enter device ID What are my best options? -- #Joseph

(Yes, today is the "John has an idea day") So, in my now fairly extensive coding of extension priority lists, I'm getting very bored of re-numbering my priority lists every time I add something at the top of the list. If I have a 7 step priority list, and I need to add something in at priority 2, then I have to re-number five other priorities, and if my Dial statement is

hello everyone! im new here and i just would like to post my problem about the dial application. when i'm calling from an outside line to our pstn which is channel 4 in our digium card. the card recognizes the call and when i dial a number i.e., 1877-LINUX-ME, channel recognizes the call and the call is bridged from 4 to 2, my problem is that the dial application doesnt seem to apply because

This week has been very productive and has shown a huge leap forward in Asterisk development. The creation of the new concepts of an "unstable" branch of the code will, I believe, make for a better development environment in the long run. With that in mind, I'm going to do something I only infrequently do, which is to re-post something in it's entirety and look for comments

I just changed the from DSL to Shaw Cable (static IP) configure the firewall but now asterisk I can not register with FWD nor VoipJet calls going out. I am using IAX with FWD Did I missed to change a setting? I don't think there is any though. I am on shaw extreme connection; I just talked shaw tech. and they are not blocking any port - I was told. So why IAX will not register with FWD and

How do I solve the problem with between patterns: _1800 _1NXX I would like all numbers 1800, 1877 etc to go through iaxtel but all other numbers 1xxx via voipjet Example in my extension.conf I have: [iaxtel] exten => _1700NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten => _1888NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten =>

Can not check voice mail-box externally. I'm trying to log-in externally (from PSTN line) to check my "voice-mail" so I created context to authenticate log-in ... exten => 7,4,Authenticate(01894546) exten => 7,5,DISA(4789|disa-access) Authentication works OK, I get inside dial none enter my mailbox extension but it doesn't accept my mailbox password even though it is the

Is there a way somehow to implement Asterisk with Credit Card Processing (IVR system)? -- #Joseph

On Sat, 2005-06-18 at 13:17 -0700, Matthew Asham wrote: > For what it's worth I use a residential Shaw connection and have no > problems with IAX registration nor SIP to our Asterisk PBX elsewhere on > the 'net. When I had a dynapic IP connection with shaw, there was not problem either. I just changed to static and IAX can not connect. How can I test it? iax2 show registry Host

After somebody records a message asterisk notifies me and encloses the WAV file. Though I'm not sure if this is a WAV format. I can not play it. According to the file specification it is: msg0000.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz How to play received message? -- #Joseph

When I try to call out to FWD over IAX2 I get: Call rejected by 65.39.205.121: Unable to negotiate codec I'm using asterisk-1.0.5 (the same settings works fine with *0.9) I've standard settings in iax.conf [general] bindport=4569 register => xxxxx:xxxxxx@iax2.fwdnet.net [iaxfwd] type=user context=fromiaxfwd auth=rsa inkeys=freeworlddialup disallow=all allow=ulaw -- #Joseph

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph