similar to: Question calling number

I use a E100P in France with a french operator E1. I can receive calls via the E1 and tranfer them to a VoIP phone, play IVR etc .... But outgoing calls doesn't work at all. I receive a RELEASE COMPLETE just after the SETUP. There is no pb with the operator (the E1 work well with an other Pbx). Here a call trace. Anyone have an idea ? (g1 is my group name for the 30 channels) -- Accepting

Hi.... Apologies for the length of this, but any help would be greatly appreciated ;-)... I have installed and configured a Digium E100P on my Asterisk PBX and I have connected this to an 8 port E1 VoIP gateway on a Cisco 6509. There is also an external E1 link from our Telco plugged into one of the other ports on this gateway and this gateway is in turn registered with a Cisco CallManager

Hi! I've a problem with caller presentation, when I call a number my presentation is set to "Presentation permitted, user number passed network screening", this works for outgoing call but not for incoming, numbers are always hidden and network provided number is always screened. I don't understand why? Anyone has an idea? My zapata options: usecallerid=yes

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension

Hi, is it possible to send 2 CallingNumbers with libpri/asterisk => wct4xxp? I see it while receiving Calls. < Calling Number (len= 6) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) < Presentation: Presentation permitted, user number not screened (0) '0712345678' ] < [6c 0d 01 83 30 32 31 31 37 37 30 35 39 38

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

Salve, I'm somewhat stuck on how to get DTMF working with my setup and googling didn't yield anything similar. My setup consists of one CAPI-capable board (AVM Fritz!DSL) connected to a BRI (T-ISDN), one HFC-S board running in NT-mode connected to an internal S0 bus with some ISDN devices (DECT stations, TA) and, of course, some ethernet interfaces. ISDN standard used is Euro-ISDN.

hello everyone, I have a trixbox server with an E1 card(not digium).It connects to an AVAYA pbx use E1. It works fine.But when i change the E1 card to digium TE220B,there is a problem. When sip extension A(on trixbox) call PSTN extension B(on avaya),A must wait longtime before B start to ring.From the log I find there are two times call. I don't know why the first request be rejected

Folks, (sorry for overlong lines) I have recently configured one port on my QuadBRI card to work in NT mode with NET signalling configured so that I can use an ISDN telephone on it. I have set up a separate group in zapata.conf and can call the phone and place calls from it like a charm. No problems at all. Problems came up when trying to hold a call and get it back. I turned on "pri debug

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 folks. My client is wanting to use call forwarding configured on their phones (Linksys SPA942), with a PRI from their provider. When we configure call forwarding, we invariably get a "The number you have dialed is not in service" message from the providers. Examining the detailed dial plan debugging as well as the PRI debugging, the number is dialed correctly. The only difference

Hi, When we make an outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup after answer. But when we place a full number before dialing everything is ok. Any help appriciated!! Thanks here is info with debug: == Primary D-Channel on span 1 up -- Executing Dial("SIP/200-164c", "zap/g1/|100|tc") in new stack -- Making new call for cr 192 --

Hi, I have a problem with dialling internationals numbers, and I don't now what is the cause. I have one asterisk with a e100p card connected to the Telco (spain/telefonica) and it can dial local and national numbers without problems but when I try to dial a international number it hangs-up. I call the Telco to ask if the E1 can do international calls and it said that it can. I have tried

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

Hello, we had a running configruation where asterisk passed the phone number and the ddi to the pstn (i.e. 595-431) Now only the rootnumber arrives: 5950 I do not know, what to do. I tried to use callingpres (now i am just hiding every number, because 595-0 is no valid extension..) but that did not worked. > Protocol Discriminator: Q.931 (8) len=44 > Call Ref: len= 2 (reference

hi all Attachd is a PRI DEBUG dumped while dialling out to a busy number among with zap(ata|tel).conf. asterisk did not flag busy, and I got a busy indicator going meeeeep-meep-meeeeep-meep-meeeeep-meep (never heard this before) Can someone help me out here? thanks roy -------------- next part -------------- A non-text attachment was scrubbed... Name: zapata.conf Type:

P.H.B. is insisting on having the ability to create a transparant SIP tunnel between old style ISDN telephony PBX with overlap dialing: PBX - ISDN - IAD - SIP - * - DAHDI - PRI The idea is that dialed numbers a the PBX are transmitted to the PRI as they are typed, whenever the PRI gets the signal that the number is complete the dialer instantly gets a ringing. This behavior is described in RFC

Hi all, I'm debugging a PRI problem, i can see the calling number but i get a busy all the time. From the output below, I guess asterisk hangs up immediately. Can anyone point out what the problem is? Thanks in advance. *CLI> < Protocol Discriminator: Q.931 (8) len=32 < Call Ref: len= 2 (reference 4865/0x1301) (Originator) < Message type: SETUP (5) < [04 03 80 90 a3] <

Hello, I have asterisk@home with zaphfc patch applied (http://dondisperato.blogspot.com/), but I can not make call to legacy PBX (Alcatel 4400). I can only accept incoming calls. I am dialing with this: exten => 202,1,Dial(Zap/g1/242) --- asterisk1*CLI> bri debug span 1 Enabled debugging on span 1 -- Executing Dial("SIP/201-4678", "Zap/g1/242") in new stack

I sent a question yesterday about having problems setting the caller ID. I turned on pri debug for both a good and bad call and I see this in the good call [2010-03-05 05:58:20.743] > [6c 0c 21 80 30 30 30 30 30 30 30 30 30 30] [2010-03-05 05:58:20.744] > Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [2010-03-05