similar to: How to check PRI status from dialplan

Hi, I would like to know if there is a way to detect the status of a span prior of sendig a call across it from the dialplan. I was asked to set up an * server with 2 spans connected to the telco and use the second as failover for the first. I checked that dialing a failed span (for instance: cable disconnected or no line) results in "congestion" for ${DIALSTATUS}, but message is too

Alright I've been having an issue when trying to dial out locally when coming from SIP. This used to work no problem, now it doesn't. Now the local PRI to Bell Is working fine I have calls coming in and out of it constantly right now. BUT if I try and make a local call from SIP (from X-Lite or one of our Linksys SPA2102s) It fails every time with errors like these == Using SIP RTP

I am trying to setup Asterisk to act like a PBX connected via a PRI gateway to a voice netowrk where Asterisk is doing outbound overlap dialing for calls that terminate via that PRI. AFter researching through the archives and online dcocs, I thought I had everyting setup right, dialplan configured for '_X!' and the 'overlapdial=yes' in the chan_dahdi.conf file, but when I try and

Did anybody figure out how to make dial detect a busy on a zaptel channel on a pri interface when using overlap dialing? According to the documentation dial should return to priority n+101, if the called party is found to be busy. I can see a DISCONNECT message with "user busy" coming from the network when I turn on pri debugging, but the dial application does not seem to notice.

Steve Casto escribi?: >/ I am trying to retrieve the cause code of a outgoing call over a PRI />/ where the number called is out of service. When an out service number is />/ called I get a recording that the number dialed is not a working />/ number. I see cause code 1 in in the CLI as soon as the call is dialed />/ the Telco recording goes on for 30 sec. then hangs up. Any

Hi - We're moving up in the world to a PRI (Verizon), and I'm having some problems with it. I'm new to this PRI thing, though, so maybe I've just screwed up a simple config detail. I've got a TE410P on a Dell PE1600SC (ServerWorks Chipset). The card itself has a green light for the PRI, and Zttool shows Span1 as "OK". All I'm getting when I call into any

Hi, I've a busy server where one E1/PRI span has been behaving oddly. Here's the span: Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 Looks good, eh? But every

Hi - Thanks for the fast response! > Try changing your dial statment to use a group instead such as > Dial(Zap/g2/3055551212) Same result: Spawn extension (from-sip, 000, 1) exited non-zero on 'SIP/2000-8dca' -- Executing Dial("SIP/2000-a4f1", "Zap/g1/19142246402") in new stack Dec 14 22:11:09 NOTICE[1506]: app_dial.c:800 dial_exec: Unable to create

The originating PRI system passes the entire dialed number in the d-channel setup frame, thus the concept of a wait time for additional digits is meaningless. Progressive digit gathering implies that the signalling is occuring 'in-band' as would be the case with DTMF signalling on analog lines. You need to look in the Ascom and find the configuration table that lays out the dialplan for

Hello List! We have problems setting the right caller id on outgoing calls. The Asterisk Pbx is located in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the local telefon number 40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID numbers available. The telco is aspecting a 3 digit long Callerid from us, for example like "710", for the extension 10.

Well, it works for me .. YMMV. Yesterday I had a problem where I had a meridian talking to * via a PRI card, and from * to the pstn via an isdn30 link. The problem was that if the number was bad, or engaged then the meridian line simply dropped, not giving the operator any indication of what occurred. With much help from this list, I managed to construct a dialplan which solved our issues.

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten

I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten => 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [0 at default:1] Dial("SIP/5166-b76004f8",

I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on a CentOS 5 server. The server has a single TE110 card connected to a provider called Alestra in Monterrey, Mexico. Since we installed everything we have been having problems dialing certain numbers, those numbers always fail when dialed from Asterisk but if you dial from your cell phone they always go through. I once has a

Hi This is the output from show dialplan dial-sipmnf-sippt-pstn [ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ] 's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config] 2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config] 3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

I am trying to retrieve the cause code of a outgoing call over a PRI where the number called is out of service. When an out service number is called I get a recording that the number dialed is not a working number. I see cause code 1 in in the CLI as soon as the call is dialed the Telco recording goes on for 30 sec. then hangs up. Any idea on how retrieve info that the called number is is

Hi, I tried so many but can't find the right setting for my problem. What do i have to configure so that the complete number including extension is displayed at the called party. At the moment the called party only sees the number 7837-0 not the 7837-134. Everything works fine. Incoming and Outgoing calls. Is there someone who configured a german pri with that digium card? I really

Hello list, I need a hand to find the best dialplan failover solution when using two SIP Trunks. My reasons to do failover are: a) one of the two providers could be unreachable b) both providers may be UP but one of them could return a SIP error message (maybe caused by DOWN E1s) Googling I found a few possible solutions: 1.

I have a system with two 4 port T1 cards, with 5 PRI's configured. Each PRI is configured as an individual PRI and belongs to it's own group (groups 1-5) This system is handling roll-over from another system, where any error in processing the call on that system results in it being sent here. This mainly results in all calls resulting in a busy being sent for retry here. I then