similar to: Unrecognized prilocaldialplan NPI modifier

Hi list! What exactly is the meaning / function of the pridialplan & prilocaldialplan? I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? Thanks!

Hi, Is it possible to have MOH in early audio, while waiting for someone to pick up a Dial() call? (When using zap channels, I have early audio working with playback) H?kon Nessj?en Loopback Systems AS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/c901ff90/attachment.htm

Starting playing with asterisk 1.6.1.0 i found the following problems: In the cdr_pgsql, the sql statement is wrong: 2009-06-25 12:17:01 COT LOG: statement: INSERT INTO cdr

Hi, I have read that DAHDI automagically turns of echo cansellation when it sees that it is a FAX. So I checked this out. I have a fax call into asterisk which is immediately called out to an external fax machine via DAHDI again.. For example, the result is: DAHDI/1-1 = incoming call, DAHDI/2-1 outgoing call. Now, with the help of dahdi show channel, if I check channel 2: echo cancellation is

Hi, I'm having strange warning from asterisk when I try to dial GSM Gateway: -- Executing [1011501522xxx at sm:1] NoCDR("SIP/ibm-b2c52848", "") in new stack -- Executing [1011501522xxx at gsm:2] Dial("SIP/ibm-b2c52848", "Zap/R3/501522xxx") in new stack -- Requested transfer capability: 0x00 - SPEECH [Jul 13 11:58:50] WARNING[18208]:

Hi, This is the output from queue show 28: 47 (DAHDI/g0/12345678) (realtime) (Ringing) has taken no calls yet Why is the devicestate "Ringing" when no channels is calling this number, and the queue says "has taken no calls yet"? Is it picking up the general state of a random channel on g0 in dahdi? Or what is happening? It only seems to happen with this particular

Hi, I'm having problems with CDR's and Queues in Asterisk 1.6.1. Heres three examples: Normal call: User A calls in to asterisk, gets a PlayFile, and hangs up. This gives 1 CDR as expected. Call to a Queue and then a playfile afterwards: User A calls into asterisk, goes into a queue, asterisk calls user B. When user B hangs up a CDR for User A is generated. (no CDR for user B)

Perfect, thanks very much hth. I just set it to unknown, but it doesn't work. Have I to use also prilocaldialplan ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 29 settembre 2005 16.22 A: 'Asterisk Users Mailing List -

Hi, I'm now thinking of always dialing out to Local/xxx at outbound/n on all my queue members. The reason for this, is both to be able to limit the number of calls to one agent, and to have fail-over-lines on the agents. (for example, if dahdi fails, go sip) But for a few years ago, I did some testing with Local/ channels, and they seemed somewhat unstable in large quantity. Are they more

System have been working great for weeks, using an average 40 of 120 dahdi channels. But today, I suddenly see scary things like this: -- Moving call from channel 5 to channel 7 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing

Hi, I have two HFC-s boards I configured in NT and TE mode respectively. When I connect the two boards together, I can dial extensions and I see the correct called and caller ID numbers: -- Executing SetCallerID("Zap/2-1", "7516862") in new stack == CDR updated on Zap/2-1 -- Executing Dial("Zap/2-1", "Zap/g2/0795025602|30|r") in new stack --

Asterisk 1.6.0.6 with dahdi 2.1.0.4 is showing a strange "Unrecognized prilocaldialplan" error with the SIP username when a SIP call is dialed to a PRI trunk. The error shows up like this: Unrecognized prilocaldialplan TON modifier: a Unrecognized prilocaldialplan TON modifier: b Unrecognized prilocaldialplan TON modifier: c Where abc is the SIP username. Is this a bug

I want to create a script for IVR that compiles responses, aggregates them to a total number. Then, run an equation based on the result. Press 1 for X (X is a positive number 500) Press 2 for Y (Y is a positive number 200) Press 3 for Z (Z is a positive number 300) Press 20 to calculate the results = 500+200+300 =1000 then, exten => s,n,Read(NUMBER,,1000) exten => s,n,SayDigits(${NUMBER})

Hi all, I have Asterisk working great for a while now, and until now, only needed to forward calls which originate from one country (Holland, +31). We have a Wildcard TE410P card and configured it in zapata.conf as: switchtype: euroisdn pridialplan: unknown ...etc This was all fine, but now we want to also forward calls to the voice-switch with a calling-number with a different country code.

Hi group, anyone can explain me the exact difference between pri value in zapata.conf ? ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN If I use it, I also must use prilocaldialplan = local ? Thanks Giordano -------------- next

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Can someone explain what the parameters pridialplan and prilocaldialplan are? What do they do and do I need them? I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx. The pbx technican complains about the format of the nr asterisk sends. Asterisk sends all numbers in on piece the pbx expects the numbers devided into

Hi all, I've been batting my head against a brick wall for the best part of the day and still haven't got any further (apart from getting a big headache, that is). I've searched the Wiki and Googled the hours away, but I still can't find supportive documentation. I've just replaced my ISDN Fritz!/chan_capi setup with a HFC/Zap configuration and had the following problems: 1

I have connected asterisk 1.0.7 with Avaya Definity via E1 with a TE100P Digium Card. Inbound calls are working perfectly and I dont have any problem. But when I try to make an outgoing call with my softphone (xlite) I am getting the following messages. Hungup 'Zap/13-1' Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack Called g1/3118 Channel 0/1, span 1 got

Dear All, I am trying to setup an ISDN line from local telco on a digium card. The problem I am facing is that I am not getting any caller id from the telco. They say that they have enabled caller id. Please help me out. My zapata.conf -------------------------------------------------------------------------------------------------------------------- [trunkgroups] [channels]

Hi I have recently upgraded my Asterisk system (from 1.2 to 1.4) and I have started to notice the following messages when I recieve a call on my Zap channel :- [Mar 11 09:20:17] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC I have a single PRI ISDN 30 link to a Siemens Realitis DX. Here is my zapata.conf :- [channels] echocancel=no echocancelwhenbridged=no rxgain=-5.0