similar to: Pri problems

Hi, I am currently using asterisk to record all incoming calls. My setup is as follows, the asterisk server has a two TE120P cards one of which sends/receives calls from the carrier and the other is connected to a Siemens HiPath 3000. All calls that come into asterisk use MixMonitor to record calls and this works fine, but if a call gets transferred the transfer information is not sent back to my

Hi, I have an IBM server running latest asterisk 1.4.x connected to a Siemens hi-path user a TE120P single-span. Approx every 8 hours (although not every 8 hours and sometimes 2 in a row) at exactly the same time I see the following errors Does anyone have any suggestions / ideas ? Thanks Ruairi Sep 24 11:56:30 asterisk01 -- MARK -- Sep 24 11:59:02 asterisk01 ntpd[3615]: synchronized to

Hi All, I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue? Daniel

hello, We encountered signaling problem with a french national carrier. They ask us, which signaling is configured on our single E1. I need to know if it's ETSI, VN4 or VN6. I know what ccs, and hdb3 mean but I do not succeed to make the link between the signaling type. I searched through RFC Q.921 and Q.931 It would be great to obtain some help. cedric

Hello List! We have problems setting the right caller id on outgoing calls. The Asterisk Pbx is located in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the local telefon number 40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID numbers available. The telco is aspecting a 3 digit long Callerid from us, for example like "710", for the extension 10.

Hi, As mentioned we have the problem that sometimes (could be up to a view times a day) for the calling party (SIP Device) you here ringing. The called party however answered the phone, but hears nothing. The calling party keeps ringing until the phone is hangup. First I thought maybe the card or the server has a problem, so I changed from a PCI beronet 4bri to a Junghanns 4bri PCIexpress and

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

Previously incorrectly sent to asterisk-dev list, sorry. I tried today while connected to a Jtec QSIG E1 card, with DAHDISendCallreroutingFacility with the following test dialplan: Extension 4888 is on the Fujitsu [incoming] exten => 8688,1,Answer() exten => 8688,n,Playback(connecting) exten => 8688,n,DAHDISendCallreroutingFacility(4888,8688) exten => 8688,n,Playback(goodbye)

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

Hi, I've connected Asterisk with 4 PRI to a Siemens HiPath 4000. For CALLERID(name) feature I wanna use Q.SIG as switchtype. Cause Siemens PBX orders Channels logical I need the parameter qsigchannelmapping=logical. Here is my chan_dahdi.conf trunkgroups] [channels] language=de context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes

Hi, I'm trying to connect to Telewest/Virgin Media with a TE110P using asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always appears as PRI span 1/0: Provisioned, Down, Active My zapata.conf is currently ----------------------------------- [channels] echocancel=yes echocancelwhenbridged=no echotraining=yes switchtype=euroisdn contect=from-pri signalling=pri_cpe group=1 channel

Hi, I am able to dial through asterisk PBX having TE120P card to E1 card running application. Communication was established successfully Now, I want to do the reverse way out. I am using the following configurations 1)zaptel.conf span=1,1,0,ccs,hdb3,crc4 defaultzone=us bchan=1-15,17-31 dchan=16 2)zapata.conf group=1 signalling=pri_net switchtype=euroisdn context=incoming

Hi, I am facing problem in configuring D-channel for TE120P card.I did the following things /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe

I am trying to connect two machines to each other with an T1 crossover cable. The first machine has two TE120P cards - one connecting to the telco on an ISDN PRI. The second to a crossover T1 cable to a second machine which has one TE120P card. Telco <-cA-> Machine1 <-cB-> Machine2 Machine1: Two TE120P cards Machine2: One TE120P card cA: Standard T1 Cable cB: Crossover T1

Hi Everyone, Got a setup as follows: Telco ----> Siemens HiCom 300E <----> Asterisk1 <----IAX2 Trunk----> Asterisk2 <----> Siemens HiPath 4xxx The solution works except for one problem. Incoming calls from the telco get redirected to the Asterisk1 box with the correct extention, only if there is a callerid set on the call, the Asterisk1 box drops the call (it doesn't

Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause

Hi all, I have a connect between a siemens hipath & Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting "The number you have dialed is not in service" In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)

Hi, We did an interconnection with our carrier few days ago. But, I noticed that there was a signaling problem on our trnuk. In fact, Asterisk indicates that the call is answered when we received ALTERTING message from our carrier. This is PRI debug logs : -- Executing Dial("IAX2/0w955897@0w955897/5", "ZAP/g1/0130450836") in new stack -- Making new call for cr 32784

Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed

Thanks in advance root@asterisk01:~# modprobe wcfxo /lib/modules/2.4.20/misc/wcfxo.o: init_module: No such device /lib/modules/2.4.20/misc/wcfxo.o: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20/misc/wcfxo.o: insmod /lib/modules/2.4.20/misc/wcfxo.o