similar to: Connecting * to a Ericsson BP250

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

Hi, sorry for the newbie hardware questions but here it goes scenario - our telco is feeding us e1 thru coax connection (unbalanced) - so the coax feed rx-tx goes to our old pabx using ericsson bp250 - what we wanted to do is to install asterisk in between hence telco<-->asterisk<-->bp250 using asterisk to power up the voip portion the problem is the we are getting crackling sound

hi, i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. The problem is, that the Bearer Capability (BC) together with the High Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the Service Indicator (SIN). The SIN is used to determine if the call is voice, fax or data. It's essential to set

hi, after several problems getting the right callerid on a E1-PRI there is (so far) only one problem left: when receiving calls over the telephone network from another E1-PRI that has a "Caller ID no screen" capability (e.g. a bank and a customer of us), asterisk does not get the callerid that is set up by the calling PBX, but the callerid of the trunk of the calling PRI. no matter

Hi, Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram. For some odd reason now that I have the asterisk box almost to the stage I want it, I hit a problem. I have a te405p in the system, Zap/g1 is connected to the telco as an ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250 phone system. When calls come in on g1 they go straight through instantaneously to the

Dear Telcotype Braniacs, I have tried doing a google search to find out what this switch looks like, what the physical interface is, but havn't been successful. I am quite new to the ISDN world so I'm not sure what to expect when I see this switch. Does anyone have any experience connecting to this switch? The following is basically all I have been given for info: 1) Ericsson, AXT121,

Hi all, I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX. I need to use Asterisk as E1 line for the Ericsson PBX. How do I have to connect them? I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions? Thanks -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am trying to install my E1 card to make a conection with an Ericsson MD-110 PBX. I installed dahdi drivers as: dahdi_hardware pci:0000:04:08.0 wcte12xp- d161:8000 Wildcard TE121 ran dahdi_genconf and it created all my e1 ports. On the other side i also configured the pbx to communicate with TE121. On ericsson side, i have no error messages. On asterisk side, no error messages. But

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Hi, Anybody using this Ericcson analog phone with Asterisk: Ericsson dialog 4187? I was told some functionalities like CLID will only work with an Ericsson PABX but other than that I would like to hear from anybody using this phone on a FXS port. Thanks, Jose Limeres -------------- next part -------------- An HTML attachment was scrubbed... URL:

My asterisk server has an external IP. Also my Ericsson DRG22 has an external IP. Tried setting up the Ericsson DRG22 device. *SIP Server IP (primary): MYIPNUMBER User name: drg22-2 password: password Caller ID name: drg22 Telephone number: 12345 *Telephone domain name: ?? (I only have stighost.someotherdomain.com ) Do I have to run my own dns on the Asterisk server? Port: 5060

Hello all. This is my very first message to the list. I have a TDM400P card, It has 2 FXO channels which are connected to extensions of my PBX (Ericsson BP250), so I can dial from any SIP softphone directly to physical (analog and digital) extensions on my company. My PBX is configured so when I dial 8 on any extension, it will redirect to the first free FXO channel on my TDM400P card.

I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk. Has anyone gotten a Webswitch running? When I try to connect asterisk thinks everything works fine, while the webswitch just rings. I belive chan_h323 is picking the wrong port to talk at the webswitch on, however I'm not sure, nor am I sure how to fix it. Any clues/hints? A tcpdump is attached to show the session.

Does anyone have info on using the Ericsson Webswitch 100 Model G4 with *? The documentation and brochures make it sound like it's some kind of specialized FXO device purely to provide legacy PBX with access to VOIP, and not a general purpose FXO-VOIP gateway. Is it possible to use it to route incoming PSTN calls into * with full callerid, and establish outgoing calls from * to the PSTN

I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I consistently get one of the following errors: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 or PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 My zaptel.conf file: -------------------------------- span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us

I have been asked by a potential customer whether we can connect an Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR. They are unable or unwilling to upgrade their E1 port to QSIG. Has anyone here had experience of successfully making such a connection? I have found a couple of hits on Google that suggest it "should" work, but I'm after something a little more

Hi, Has anyone on this list tested out the new SE P1 phones ( http://www.uncrate.com/men/gear/cell-phones/sony-ericsson-p1/). It says it supports VOIP, wonder if it is working with asterisk. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070517/467cbea1/attachment.htm

Hello, This is a little off-topic. I have an Ericsson FCT f251m, according to the specs it supports call signalling through polarity reversals and loop break, but it's currently disabled. On my PSTN line, my TelCo does send polarity switchs to signal answer and hangup (answeronpolarityswitch=yes and hanguponpolarityswitch=yes in zapata.conf), and asterisk detects it alright. I've been

Hello fellows!!! I'm having problems with Ericsson Dialog 3221 phone and DISA. I've configured an extension to test DISA and it work properly with all other phones, but freeze with the mentioned phone. Here is my extension: exten => 105,1,Answer exten => 105,2,Set(TIMEOUT(digit)tting =5) exten => 105,3,Set(TIMEOUT(response)=10) exten =>