similar to: Extension based call forwarding using capiECT

Hi all, does anyone have capiECT working? I can't get it to work - please see my logfiles below. While using an * CVS version of late September and chan_capi-0.2.5 (I guess), it worked!!! (I know, never change a running system ... should've backuped ... etc.) Here's the setup: NT----AgfeoISDNPBX----AgfeoISDNPBX----AnalogPhones in between the two PBX I attached my * server with a

Hi, I'm trying to get capiECT working. I'd like to transfer call to another ISDN router connected extension and free channel from router to Asterisk. I get incoming call on CAPI and would liek to transfer it to dialed local extension - 400 in this case: [outbound-capi-local] exten => _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on msn CAPI/${CALLERIDNUM}) exten =>

Hi I use RH 9 + asterisk v1.0 stable + 2 PCI fritz cards + chan_capi 3.5 and it works fine, Since my users want fax fonctionnality and customers know 1 of the msm as fax number I wanted to try the chan_capi-0.3.5 patch if I patch chan_capi and run make, I get an error message , as you can read below, orginal chan_capi compile works, when patched I get an error, no CID ? Any idea ? anybody

I had unexpected hangups from my asterix box using the i4l driver. (SIP <-> SIP calls worked execellent, but SIP<->ISDN didn't.) Then I changed the i4l driver in modem.conf with the chan_capi from jungham. (http://www.junghanns.net/asterisk) I followed his instructions in the INSTALL file, and first encountered some errors compiling it. It help by deinstalling several

Dear list, I tried to install the CAPI support RPM from the SUSE 9.2 disks. YaST tells me it is happy with the process. The Asterisk release I am using is the one that comes packaged in RPM format, also included in the distribution. Still starting asterisk with the usual asterisk -vvvc I see that something goes wrong. [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242

Hi chan_capi users, this thing is awesome, no delays like in modem_i4l! Plus, it got those nice ISDN features. Here's my question: Does my service provider (Deutsche Telekom) have to provide me with these Services (CD, ECT)? (the Readme in 0.2.5 says "does not relay on service CD") I know, that I don't have CFU,CFNR,CFBS (which I would have to order seperately). How likely

hi all I'm being spammed by these messages in the console (see below) and sound doesn't work with today's cvs. I rolled back a week, and it works fine. In addition to the sound problems, I had to enable inband dtmf squelch on the dilnk mgcp phones. if not, each pressed key was counted twice NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process

Hi, I am trying to upgrade an Asterisk 1.0 with chan_capi 0.3.4 to a more recent version, but I cannot find any working combination of Asterisk an chan_capi any more: On "junghanns.net" there is a chan_capi 0.3.6, but this won't compile against any recent Asterisk (missing channel_pvt.h). The production version of BRIstuff comes with an old asterisk (1.0), the experimental version

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes

Hi, On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new Numbers. I changed the extensions in extension conf to match the new numbers. But i always get: Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel 'CAPI[contr1/89064934]/0' sent into invalid extension 's' in context 'default', but no invalid handler I only changed the

Ok I know this may have been covered and I did have a look back in the archives but didn't find anthing so I am asking it now.. Many moons ago the MySQL CDR functions and MySQL Voicemail functions had to be removed from the main asterisk code because of licensing issues.. Now there is new MySQL stuff like MySQL FRIENDS for SIP and IAX definitions.. So how is it that these options

I run an Apple OS X workstation and I've got a server on the same LAN that's both a webserver and an Asterisk PBX. I wanted to be able to originate calls in the OS X Address Book application, and have Asterisk dial them and connect them to the phone on my desk. I've assembled a system that uses AppleScript to connect, via XML-RPC, to a web application that, in turn, connects to

Hi All I am having this problem when setting up a H323 call. Can anybody tell me what is going on? Thanks Serge ------------------ NOTICE[245776]: File chan_oh323.c, Line 1293 (oh323_write): H323:1637: Format changed from 4 to 8. WARNING[245776]: File frame.c, Line 76 (ast_smoother_feed): Smoother was working on 4 format frames, now trying to feed 8? ERROR[245776]: File chan_oh323.c, Line 1380

Hi all, I get "Unknown RTP codec 72 received" message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console

I recently updated (fresh checkout) to the newest zaptel and Asterisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack NOTICE[245776]: File app_dial.c, Line 698

>>0.4.0? who is working on chan_capi? long time with no updates from. >>I'm rewriting the chan_capi logic. >> >> > >kapejod already did 0.4.0-PRE1, which I use as base for cleanup, fixing and >new features. > >What exactly are you rewriting? If someone else is working on chan_capi, we >should synchronize our work and ideas. > > yes

Hi, although setting up voicemail for busy and unavailable should be easy, things aren't working the way they should in my configuration (asterisk 1.2.14 bristuffed): Here's the relevant part of the extensions.conf: exten => 56830976,1,Answer() exten => 56830976,2,Dial(SIP/hbaumgart,20,tr) exten => 56830976,3,VoiceMail,u76 exten => 56830976,4,Hangup exten =>

Folks, I'm having asterisk connect to another device via an E1 PRI. However, using "pri intense debug span 1" I'm seeing asterisk ONLY sending SABME packets. No packets are received. The other end is saying ISDN layer 2 is okay. Can this imply RX-errors where the other device is ack'ing the SABME's (and hence considers layer 2 established), while asterisk isn't

On Wed, 30 Apr 2003 00:24:19 -0400, Uriel Carrasquilla wrote: > >Gary: >I just copied the content from chan->exten to chan->dnis. I am calling from How are you doing this coying ? >one extension to another. >Have you got DynExtenDB to work? nope, haven't got over the first problem yet. Gary .