search for: kibeki

Hi, at the moment we have in Avaya Integral PBX with german pri (30 lines). We want to smouthly migrate to an Asterisk server. For this reason: Is it possible to route the external german pri (E1) through Asterisk server to that Avaya PBX? I think at first we need a Digium e1 card 4-Port. But how do we have to configure the routing of the whole PRI? I really would appreciate any sample

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

Hi *, I want to integrate the Eicon Diva 4Bri Card to Asterisk. Eicon drivers and capi is installed. I use the latest dev version from eicon compiled and installed for my fedora 2 system. I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. I changed the Makefile to include the capi20.h from eicon:

Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306

Hi, some days ago i found a list with default extensions for things like 'echo test = *44' . But i can't the point where they have been. So, maybe someone of knows what i looking for. thanks, kib

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first rings at my phone and in the second i can see the callers number. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

Hi, i try again to ask this. When i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Thanks, Kib

Hi, the following from extension.conf does not work correctly: exten => 301, 1, Dial(SIP/455&SIP/456, 15) That is the console output: -- Executing Dial("mISDN/1/105", "SIP/455&SIP/456&SIP/456| 10") in new stack -- Called 455 -- Called 456 -- SIP/455-46a8 is ringing == Spawn extension (incoming, 301, 1) exited non-zero on

Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley could be produce hangups callwaitingcallerid=yes ; show

Hi, i need an recommandation for an ATA adapter to use with an anlog fax maschine. I would appreciate any hints. Regards!

Hi, the option c for the announce of the user count does not work me in * 1.0.9. exten => 9999,1,Wait(1) exten => 9999,2,MeetMe(|Mdcs) And how to handel the marked mode with option A? I can't find any sample config for this. Regards

Hi, I tried so many but can't find the right setting for my problem. What do i have to configure so that the complete number including extension is displayed at the called party. At the moment the called party only sees the number 7837-0 not the 7837-134. Everything works fine. Incoming and Outgoing calls. Is there someone who configured a german pri with that digium card? I really

Hi, I don't if was yet an issue. It really would be nice if each user is able to active/deactivate the mail forwarding of his voicemail via the VoiceMailMenu. Regard

Hi, can anybody tell me what the errors mean and why my asterisk server falls from time to time. From time to time means several hours, not regularly. I also can provide a core if someone can debug? Thanks and regards Jan 11 14:34:59 NOTICE[13573] chan_zap.c: Hangup, did not find cref 83, tei 64 Jan 11 14:34:59 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:03

Hi, we use Polycom SP IP 501 phones. We use the standard key/softkey configuration to do attended transfers. The only thing we miss is the CID info of the original caller after the call is transfered. This behaviour is different from the blind/direct transfer. With blind transfer method the original CID info is displayed. We already opened a call (in 2006) with Polycom JIRA. This is what they

Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as caller number. 2. A call made from a SIP