similar to: Nationalprefix chan_dahdi option

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:

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

Hi I have done everything richard told to do ECT . below is my trace, anyone can help ? -- DAHDI/i1/09123278669-4 answered DAHDI/i1/88050048-3 -- Native bridging DAHDI/i1/88050048-3 and DAHDI/i1/09123278669-4 PRI Span: 1 Adding facility ie contents to send in FACILITY message: PRI Span: 1 ASN.1 dump PRI Span: 1 Context Specific/C [1 0x01] <A1> Len:11 <0B> PRI Span: 1

Hi, Starting in Asterisk 1.8.0, Asterisk supports connected line updates. This is fantastic for SIP. How can I prevent them from being sent to a PRI channel? I'm having problems when a call is answered by an internal SIP extension, then transferred (blind or attended) to another internal SIP extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform APDU and drops the

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All

I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing. Below is a dialplan snippet and the resulting CLI output. This is running in an 'h' extension. Noop(DIALSTATUS=${DIALSTATUS}) Noop(CDR(disposition)=${CDR(disposition)}) -- Executing [h at pbxmax-dial-simple:1] NoOp("SIP/msx_01-0000005b", "DIALSTATUS=ANSWER") in new stack

handled. So....what do I do? Thanks, MD =1=================================================== !! Invalid Protocol Profile field 0x11 -- Accepting call from '2004000' to '111' on channel 0/23, span 1 -- Executing NoOp("Zap/23-1", "Incoming call from Meridian1") in new stack -- Executing NoOp("Zap/23-1", " From number: 2004000|

picture. I know the firmware on the Nortel is old, so I'm guessing that libpri is sending something that the Nortel does not know how to handle. Is there a way to dumb down what libpri sends? From everything I've read PRI is an evolving standard - and older devices may struggle with newer extensions/developments. (This might be very handy for users trying to talk to old pbx's.) Is

Hi, I tried to use Early Media: exten => 1,1,Playback(demo-thanks,noanswer) same => n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using "Progress()" in the first priority. I tried "pri set debug on span 1" and got the

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.

I have a working system with asterisk 1.4.26.2 libpri 1.4.1 and zaptel 1.4.12.1 With a digium TE205p. I am trying to update to libpri 1.4.10.2. When I do, incoming calls work but outgoing does not. When I do this I "rm /usr/lib/libpri*" then just install libpri-1.4.10.2 as normal. I then do a make clean in asterisk and make distclean ,then configure, make and make install. I do

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]

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

I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk setup with outgoing calls not completing and requiring an Asterisk reset to 'unstick' span 1. Sorry this is a bit long but I'm completely out of my depth :-( This system has been in use for some while and I recently upgraded it to asterisk 1.4.24, zaptel 1.4.11 and libpri 1.4.9. I didn't change

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

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

Hello, Configuration: Asterisk CVS HEAD 20050730 on RH EL3+ DIGIUM TE110P PRI card + euroISDN E1 I am trying to sort out the problem: 1. Provider's switch sends "SETUP"; 2. Asterisk receives "SETUP", rings allocated extension but does not send "Setup acknowledge" (or any other messages) to switch; 3. After 4 seconds of waiting of *'s response switch sends

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

Hi everybody, I've connected Asterisk 1.2.5 (libpri 1.2.2, zaptel 1.2.4, Linux 2.6.13.2) to an Avaya-Tenovis PBX via a PRI/E1-line. Calls from SIP-phones via * to the PBX work fine. However, incoming calls to * only result in: -- XXX Missing handling for mandatory IE 24 (cs0, Channel Identification) XXX -- which seems to be an * problem, because a Windows-fax-machine works fine on a PRI