similar to: Q: How to get the preset callerid from a CLID-no-screen E1-PRI

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 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, 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, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission

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 List, This is somewhat off-topic since the problem itself isn?t asterisk but the Ericsson BP250 I want to connect to. But since there have been a couple of posts relating in part to that system I am hoping someone can help me out. What we want to do: PRI <---> BP250 <---> Asterisk Currently the BP250 handles extension 000 - 899. We now would like to forward a chunk of 100

Hey guys, When a call comes in via the PSTN to our Call Manager 3.2 and is forwarded (via unity and H323), the caller id is set to our Unity Voicemail instead of the caller id from the PSTN. We're using the oh323 channel in this case. Has anyone experienced this issue before? Any solutions? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>

hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on the analog cards does not work or i get cracking noise or even hangups on my BRI lines, due to bit slips.

Hello, A question on channel originating (call files and AMI Originate): How can I change the CALLERID(num) var (because of the E1 provider needs), but having another n?mber (the original one) stored on the "clid" CDR field on the database? A channel agnostic solution would be the best one, without having to deal with the problem based on what type of Tech used for the outgoing

Hi all, I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. the customer is currently using a paging system (small receivers which display a callback number and a base station (transmitter) with several antennas at the site) the problem is, that the currently operative base station uses 4 ISDN BRI interfaces. But the protocol is old germany 1TR6

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

Hi, I'm facing a strange problem. My setup is: Alice cellphone <--GSM--><--ISDN--> Asterisk <-- ISDN --><--GSM--> Bob cellphone When Alice calls Asterisk which forwards the incoming call to Bob, sometimes Bob sees Alice's number, sometimes he sees a default CallerID (which happens to match the dialed number and the ANI). For various reasons, Bob really needs to

hi, since my latest libpri update i get these messages: !! Unable to handle ROSE operation 36 !! Unable to handle ROSE operation 30 i searched through ITU X.219 and X.229 but can't find any values for the Remote Operations Service Elements. are these AOC-E messages? regards frank

I have a client who needs an application for there field techs to call in when they arrive on site and when they leave. The logic behind it seems pretty simple. I am going to write something in AGI to capture some DTMF tones and update this data into MySQL to run some reports from. But here's my initial problem. I have started to create a simple AGI script to capture the CallerID, but I

I'd be grateful if anyone here could comment knowledgeably on an idea that I have had, as to whether it should be possible or not. Consider two Asterisk boxes, each with one or more E1s on EuroISDN. Each box has a different telephone number that hunts across all its E1 channels. In addition there is another number that hunts across all the channels on all the boxes. A call comes in to one of

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

hello everybody, one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco <---pri---> asterisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with

Dear All, I have a setup where three errision PBX's (head office & 2 branch offices ) are @ three different location. Head of has a errision BP250 where the branch officers has Errision BP50's. I wana connect all three PBX's through IP link's. My idea is to connect BP250 ( head office ) to asterisk. This has to be done using analog interfacing coz there is no E1 / or T1