similar to: Sending 2 CallingNumbers

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 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 --

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

Hello all, I have a question concerning the calling number with an incoming PSTN call through a E100P : Here is what I see with a pri debug : < Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) < Presentation: Presentation permitted, user number not screened (0) 'XXXX333007' ] < [6c 0b 20

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

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

The latest libpri is supposed to handle this properly, but doesn't seem to. Here's the debug info. CALLERID(name) is set to empty. < Protocol Discriminator: Q.931 (8) len=66 < TEI=0 Call Ref: len= 2 (reference 256/0x100) (Sent from originator) < Message Type: SETUP (5) < [04 03 80 90 a2] < Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability:

Hi all, I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on a openSUSE 10.2, i586 host. The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the destination FAX devices are in Germany too, but in different areas, so I have to use a city prefix. I did set the pri device in debug mode, below are two calls, to two different FAX numbers, the first is

After sometime I got my E1 PRI configured correctly in /etc/zaptel.conf and I now don't see any alarms on the E1, but I can't still dialout correctly, I enable every debug that I could though of and this is what I see: At the end the D-channel is down and I cannot even try to connect another call because it tell me that the zap channel is unavailable ,the other this that I notice is that I