similar to: Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789

It looks like you are trying to dial out on your 'NT' instead of your 'TE'. Try changing Dial(DAHDI/g1/${EXTEN:1}); to Dial(DAHDI/G1/${EXTEN:1}); Oh, and I'd use mISDN for BRI as DAHDI always gave me problems. HTH -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Is it possible, for testing, to connect an cat5 straight patch cord between 2 ports of a Digium B410P card and use these 2 ports as a normal dahdi trunk ? I've tried this: One port is set as NT, the other as TE. I would expect timing to come for system hardware so I choose in /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami Results: - both ports lights are green -

Hi! I am searching for the description of the available dialstrin options for the DAHDI channel (and also other channel types). I am not looking for outdated voip-info links, but for the authoritative source, e.g. something like "core show application Dial" Does such thing exists? thanks Klaus

Hi, I've got the following use case where I want to simultaneously dial 2 endpoints that both need different CallerID presentation. How can I do that, from the dialplan preferably ? For instance, let say phone A needs to both dial B, an internal SIP phone and C, a cell phone reachable through a DAHDI span from a an Asterisk system where : 1. users can use 4-digits short numbers to reach

Dears; I am facing the following problem: Already we requested from the service provider to enable the auto jumping service for our analoge telephone lines, so because we have 4 telephone lines from the service provider, then if you called line # 1 and it was busy, then the call will be sent to any available line #2 or #3 or #4, and if you call line # 3 and it was busy then the call will be sent

Hi, I've always used dahdi-genconf to just create the dahdi-channels.conf and since our PRI is fairly simple (just dump all the channels into one group) it works with dialing with dahdi/g1/(number). I'm trying to understand the file though for my own reference. It seems the file looks like this: group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel => 1-23

Hi all, I was using dahdi 1.6.2.0.9 version for a long time. We decided to upgrade to 1.6.2.22 a few days ago. After that we started to have some problems with dahdi channels. PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2 We have 2 PRIs between Ericsson pbx and asterisk and a sip trunk for outside calls. At begining everything works fine but in a few hours, calls from asterisk to ericsson

Hi, can anybody please show me a valid dial plan for a dahdi card with a bri port? I can not get asterisk 1.6 to dial a number. I want to receive one call on the first b channel and dial another number on the second b channel of the same isdn port. I tried something like Dial(DAHDI/g1/123456789) but that does not work. It always says that the channel is busy. Kind regards,

Hi Listers, I'm running 1.4.25-rc1 on opensuse 11.0 with dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2. Incoming calls work fine. Outgoing calls made directly (exten => s,1,Dial(DAHDI/G1) then number work fine. The problem I have is trying to let Asterisk make the call (exten => s,1,Dial(DAHDI/G1/5551212,,r). If I use "m" (moh) the

i need your help regarding some issue related to the outband calls i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim with 2 ports when i try to call my phone number all time i receive message busy number this error just with g1. with g2 there is no problem i can call without issue can anyone see the CLI and tell me what is the problem thanks and regards == Parsing

Hi, I need to run ztdummy for Paging, but now that this is all become dahdi I don`t really know where to start. I did build dahdi before building asterisk, but that`s it. I find it hard to find any documentation referring to dadhi instead of zaptel. I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Regards,

Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a "Dial(DAHDI/g1/voicenumber,r)". The call is bridged and everything is fine. "dahdi show

I am current running on a production system zaptel 1.4.12.1 libxpri 1.4.1 asterisk 1.4.25 The above configuration works. I tried to update to dahdi 2.2.0, libpri 1.4.7 and asterisk 1.4.25 This did not work. calls came in but not out. I dropped back to the initial configuration. Today, I dried only updating libpri on the original configuration. I removed /usr/lib/libpri* installed libpri 1.4.7,

Hi, I tried again getting DTMF detection on my ISDN devices with dahdi going again. I used the channel debug to see whether asterisk sees the frames and detects them as DTMF. Interestingly here's what works: 1. GSM phone -> chan_dahdi g1 -> asterisk -> can_sip -> SIP phone Both the GSM phone and the SIP phone can issue DTMF that will be detected as features (transfer) 2.

Hello all. Please, I'd like to know if somebody can help me with this problem. I have successfully configured a PBX with Asterisk 1.4 and a Digium analog card with 4 ports. This PBX has a lot of incoming and outgoing calls, and works perfect in general, but there are some extrange cases where an incoming call is bridget with an outgoing call, and the caller that is calling TO the PBX can

Hi, I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and try to make a call I get the following error message: ======================================================================================== -- Executing [6781948 at default:1] Dial("IAX2/iaxy-7477", "DAHDI/g1/96781948") in new stack [Apr 25 13:00:10] WARNING[3772]: app_dial.c:1806

What's this: -- Attempting call on DAHDI/g1/9<removed> for application Wait(5) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- Channel 0/2, span 1 got hangup, cause 44 -- Forcing restart of channel 0/2 on span 1 since channel reported in use -- Hungup 'DAHDI/2-1' Where can I look up "cause 44". And if this is the sort of transient

I was running : asterisk 1.2.24 zaptel 1.2.21 libpri 1.2.6 I remove zaptel and compiled asterisk 1.4.22 libpri 1.4.7 dahdi 2.1.0 dahdi_cfg -vvv DAHDI Tools Version - 2.1.0 DAHDI Version: 2.1.0 Echo Canceller(s): Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel

Apart from you were dialling out on your inbound context and vice-versa. The best advice I can give now is to change to mISDN - as this is proven to work with v1.4 and v1.6. Actually - have you tried putting the 100ohm termination on for your NT port? I need to do that with mISDN as it only allows ptmp for ISDN extensions. Cheers Andy -------------- next part

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212