similar to: Problems with overlap dial and Xorcom Astribank BRI

Hello, I'm in the process of building an Asterisk machine and need 5 or 6 BRI-Channels. I was looking for the beroNet and Junghans cards and stumbled upon the Xorcom Astribank xBRI products. Has anybody tried out the Astribank xBRI-Channel Banks? Are they production ready or should I go with a beroNet BN8S0 or JUNGHANNS.NET octoBRI ISDN? Thanks in advance, Lars -- We don't like their

Hello Mailinglist, we have a Xorcom Astribank which is installed at our client and working since 10/2008. Suddenly last week there where odd behaviors, no calls going through (it is used to route calls to GSM Gateways). Upon testing (reinitialization of the complete gear) it turned out that one call get's through and then nothing more. I have nothing in the logs, indicating that Asterisk

Hi, I have upgraded an Asterisk installation with a Xorcom BRI Astribank that was working under Ubuntu 8.04 to Ubuntu 9.10 and the device is no longer initialized. When I reload the udev rules, I see that the rules seems to be correctly loaded: udevd[452]: reading '/etc/udev/rules.d/40-xpp.rules' as rules file However, these rules do not seem to be triggered: # udevadm trigger

Hi, I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and wondered if anyone is able to offer any advice. In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk box. e.g: HiCom user dials access code and can call Asterisk extension or establish SIP call over Internet. Likewise, I'd like Asterisk to be able to present a call to the Hicom, either

Hi, I have a problem with people that are calling from analog lines. We have a block of numbers 12345 - 0 to -99. Most calls are transmitting the whole number including the extension. There's no problem with that. But people calling from analog lines are connected to our asterisk box as soon as they finish dialing 12345. They don't get a chance to dial an extension. Just inserting a

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

Hi * :-) I found in the online WiKi docs some information on how to integrate Asterisk with "old PBX"... http://www.voip-info.org/wiki-Asterisk+legacy+integration ...but I couldn't find anything on integration with a Siemens HiCom 150E. Later on we'll migrate to a HiPath 3750 so information covering this model would be nice too... Do you know if any of the PBX listed

Hi, I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches. So I'm running zaptel-1.2.17.1 and asterisk-1.2.18. The problem I'm having is that for one I get no LEDs showing if the unit is in TE and NT mode (not a issue for me but may have some impact on things) I have no errors in any logs I

Hello List, it is a very long time since I wrote here.... It has been still in Zaptel times.... Today I am run into a related problem: I can't get a DAHDI setup to work 100%. I am configuring an Astribank XR00013 (BRI, two ISDN ports). At some degree the installation (latest DAHDI drivers, Asterisk 1.6.0.18 on Centos 5.x) works. I can get incoming calls to the dialplan context that is

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

Hi, Is is possible to implement this: Hicom150 --- BRI (QSIG) ---- Asterisk I've been reading Siemens documentation and they say: "Digital nailed connections Corporate communication networks can be implemented over digital S0 or S2M nailed connections between several Hicom systems using the CorNet N protocol and between Hicom and non-Siemens systems using the QSig protocol. The

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

We are using MFC/R2 driver successfully in at least three places in Brazil. I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite softfones, but SIPURAS and Linksys get a garbled audio, something like a "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and the Linksys

Greetings, asterisk list and community, I have a problem in how our telefon switch (Siemens HiCOM) "talks" with my new configured Asterisk server (V.11.18.0) without my Asterisks server in the middle.... <phone> <--> Siemens HiCOM <-ISDN-> NTBA <-...-> PBX Telekom A phone connected to the switch requests an "Outgoing" line by dialing "0".

Hi, Xorcom astribanks get initialized straight on when using Ubuntu 11.10 packages but I am having a hard time to get the same result running in a qemu/libvirt image. The first difficulty is that astribanks devices get different usb device ids during their initialisation process, requiring hot plug support. I have figured out how to solve this issue using the technique described in this post :

Hello, I recently got a problem I have never seen during my previous installations. I have a Xorcom device with 4 BRI and 8 FXS ports. I use ubuntu Linux and asterisk 1.4.22. Everything works fine but after some time zap channels disappear in asterisk. root at pbx1:~# cat /proc/zaptel/* Span 1: XBUS-00/XPD-00 "Xorcom XPD #00/00: BRI_TE" AMI/CCS NOTOPEN 1 XPP_BRI_TE/00/00/0 Clear

Hi guys, This has been bugging me for days and I can't figure out what's happening. Using the same channel on DAHDI, I can't make a call after previous one being hung up by remote side. However, it works like it should if my end hangs up the call. To make things interesting, DAHDI, or astribank rather, seems to "reset" itself after receiving an incoming call on that DAHDI

Hello all, I really checked voip-info.org but it still seems to be not very easy and I just hope that there is anybody with a simular config. We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper than a channel bank :-) Carrier ----S2M------ * -----S2M------Siemens | |

Hi I've bought the Wildcard TE110 some days ago but I'm unable to get it to work with Siemens HiCom 300. I've tried this so far: 1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4 and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard takes a few seconds and sets the link to green (OK). 2. I've tried to connect our running

I have recently updated from Centos/*1.2 to Ubuntu Server and FreePBX 2.8.0.2. We have an Astribank with 4 T1 ports and 16 FXS ports. After updating, we had it working for a while with one NT PRI and one TE PRI and, in the process of trying to configure another PRI, I ran into a couple problems. (1) As my configuration changes didn't seem to affect the Astribank, I power-cycled it. I