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Hi. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

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

Hello, I have an Asterisk system in the UK using ISDN service from BT. My problem is that the called number is always passed to the provider with the Type Of Number declared as ?national? despite pridialplan (and prilocaldialplan) is set to ?unknown?. My questions are: - How can I find out what the effective value is for the above two variables without actually making a call? (I have very

Asterisk 1.8.4-rc2 (and 1.8.3) DAHDI Version: 2.4.1.2 libpri version: 1.4.12-beta3 We are having a problem with getting the nationalprefix option of chan_dahdi.conf to work. National calls do not have a "1" added to them when nationalprefix=1. The PRI debug shows the call coming in as a National Call, but the dialplan sees the call without a 1. chan_dahdi.conf: <snip>

We use asterisk 1.6 with DAHDI and a PRI ISDN 30 line in the Netherlands. Can anyone help me sorting out this issue?? Thanks in advance! -- Executing [s at macro-transfer:25] NoOp("SIP/joostkuif-00000003", "gehe= im") in new stack -- Executing [s at macro-transfer:26] NoOp("SIP/joostkuif-00000003", "voor= de SET CALLERPRES() =3D

We use asterisk 1.6 with DAHDI and a PRI ISDN 30 line in the Netherlands. Can anyone help me sorting out this issue?? Thanks in advance! -- Executing [s at macro-transfer:25] NoOp("SIP/joostkuif-00000003", "gehe= im") in new stack -- Executing [s at macro-transfer:26] NoOp("SIP/joostkuif-00000003", "voor= de SET CALLERPRES() =3D

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

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 all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Hello, all. Could you tell me how to set the type of number in the outgoing SETUP message sent over PRI trunk? I need to have: Called Number (len=18) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '318989263037666' ] But always have: Called Number (len=18) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan

Hi all, using latest asterisk-svn I want to reflect an video call incoming via an PRI EuroISDN channel to another outgoing PRI channel, and I want the the outgoing channel to have the exact same bearer capability < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) < Ext: 1 Trans mode/rate:

Hi, I have the following setup: (sip clients) -internet- asterisk A -IAX- asterisk B -PRI- (pstn) This works fine for regular calls sip->pstn. the calls go through perfectly. However, when one of the sip clients (a snom320) is set to redirect to the pstn, then all I hear is congestion tones when I call that sip client. I am at a loss to see why the redirected call would fail, when the

When receiving multiple calling numbers via a PRI for a call setup, I cannot find the ability to select between either first or last. Is there a way to do this currently? If not, can anyone help me in getting this to work? Here is the dump of the PRI in intensive debugging. Please note that I want the first calling number, not the last. Any help would be much appreciated. Below the PRI trace

Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and

Hello, I have a problem, and I'm looking for you help. When I dial certain number my calls are getting dropped. I initiate the call, I hear IVR, then I am being transfered to operator, and then suddenly I get ISDN DISCONNECT message. I had this type of problem some time ago, and I thought it was a problem on the other end. But now this is a second time it occurred and I want an expert to

Hello, Quick Diagram: Telco-PRI -> Asterisk <- Norstar PRI -> Norstar PBX (DMS100) (TE405P) (DMS100) | | V Cisco 7960G (SIP) I'm trying to change the Origination Number on my outgoing PRI, and running into a weird problem. If I make a call from a SIP extension off asterisk using the following context:

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.

Hi, I'm facing a strange problem. My setup is: Alice cellphone <--GSM--><--ISDN--> Asterisk <-- ISDN --><--GSM--> Bob cellphone When Alice calls Asterisk which forwards the incoming call to Bob, sometimes Bob sees Alice's number, sometimes he sees a default CallerID (which happens to match the dialed number and the ANI). For various reasons, Bob really needs to

I there, I have some trouble to do working caller id name for outgoing calls on the PRI we just installed. Caller id name work on incoming calls only. Caller id number work on incoming and outgoing calls. The provider, Goup Telecom, said that's in what i'm sending. They said that I send the cid name in ascii code and to do it working, I need to send it in hex. So I take some traces

Hi all I wonder if anyone can give a little insight into this; asterisk@home 2.2, HP Proliant gl3, sangoma A101,Cable and Wireless, ISDN30. 10 zactive channels Incoming calls work fine no problems tested 43 DIDs all working. zaptel.conf # Global data loadzone = uk defaultzone = uk span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 zapata.conf [channels] language=en context=from-pstn