similar to: Connecting Siemens HiCom PBX with Asterisk through E1

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 all, I have this topology: telco_company>ISND30/PRI/>siemens_hicom_150>classic_analog_users_with_extensions_100-499 and I want to integration asterisk PBX on linux redhat 8 for cca. 4 users. so, my first question is, which hardware I need in linux server and which in hicom 150? and my second... it is possible to connect asterisk PBX directly to PSTN? in this case I'll have this

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

Hello, I would like to make sure I am not going to shoot myself in the head, so I would appreciate if you could tell me if what I am willing to do is realistic (and will work) or not. I am willing to replace our office "London" phone system when a new E1 (G703) is installed later this month and I have the choice between using asterisk or a more classical hardware PBX. The planned

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

Has anyone implemented an outcall notification when there is a voice message waiting? I would like to have the system notify me of awaiting voice messages by a telephone call rather than an email notification. I would imagine that a call could be dumped into the asterisk spool directory, but I'm not sure how I would monitor for messages waiting. Has anyone implemented such a feature for

I have an Asterisk system with a TE110P installed and connected to an ISDN E1 PRI that is delivered via a 2mb SHDSL connection. I am experiencing delays (the type of delay you would get on an international call) during calls. I am wondering if anyone could advise, would the problem be with any part of the Asterisk system or is the problem with the fact that the ISDN is delivered over the internet?

Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am trying to setup an E100P for use on Swisscom E1-PRI here in Switzerland. Swisscom seems to use Siemens hardware. Here are my configs (cvs from a few hours ago) : zaptel.conf loadzone=fr ; tried "de" but got warning at modprobe defaultzone=fr span=1,1,0,cas,hdb3,crc4,yellow ; Seems to be right, as the Siemens SDSL-G703 modem switches its LEDs off ... not sure

Hi I try to connect my asterisk with a Siemens Hicom pbx. I have a PCI cologne Chip card wich support NT mode. I have compiled mISDN driver, and I use chn_misdn from debian package. The card wotk fine in TE mode but mot in NT mode. for informations : routeur*CLI> misdn show stacks BEGIN STACK_LIST: * Stack Addr: Uid 40200001 Port 1 Type NT Prot. PMP Link DOWN --> bchan: addr 0 channel

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, 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

> Subject: [LARTC] Small ISP problems (CBQ) > First of all, this is what we want (in network priority order): > 1: SSH - to be realtime always. I don''t think you want this to always be high prio - that includes scp. > 2: HTTP to be fast, always. Clearly can''t be done if you have more http requests than your bandwidth can handle. > 3-> ftp, direct-connect,

Hi! I want to test Asterisk<-->Siemens HiCom integration using Q.SIG instead of ISDN. I did not find any documentation about Asterisk und Q.SIG. Thus, I wonder is it sufficient to set "switchtype" from "euroisdn" to "qsig" or are there any other things which I have to take care of? Thanks Klaus

This is partly an llvm question, and maybe partly C++. I am trying to make calls on code in the llvm infrastructure.(3.4.2) I appear to be missing some include files, but don't know how to track them down. I get compile errors on existing llvm header files, like the following two examples: -------------------------------------------------------------------------------- In file included

If someone could explain me this (added some extra debugging code Mar 10 00:47:13 NOTICE[7561]: chan_sip.c:8448 handle_request: Registration from '<sip:a7@domain>' failed for '172.20.41.7' Mar 10 00:47:13 WARNING[7561]: res_config_odbc.c:97 realtime_odbc: SQL param name is a7 Mar 10 00:47:13 WARNING[7561]: res_odbc.c:90 odbc_smart_execute: SQL Execute error! Attempting a

Sorry for the long descritpion of the problem, I''d like to know If I misunderstand something or if I meet an intrinsic limit of my setup. 217.58.51.162 HDSL eth1 - SRV_XP: 192.168.254.10 eth0: 192.168.254.1 -----+------------------+------- 81.121.243.250 ADSL eth3 - I want to allow incoming pptp request (port 1723) to be forwarded to srv_xp

Perfect, thanks very much hth. I just set it to unknown, but it doesn't work. Have I to use also prilocaldialplan ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 29 settembre 2005 16.22 A: 'Asterisk Users Mailing List -

Hello. I have a pretty strange problem with routing and iptables mark. My firewall has a classic 3 NIC config: one nic connected to the ISP routers, one network for DMZ and the third network for my private network. Here is the schema: HUB HDSL router----+----ADSL router | | HDSL IP---eth0---ADSL IP | | +-------+------+ | | | | | | | | |

> Unless you're going for some kind of distance record, standard Cat5 will > work > without any issue on any modern installation. As I said, I'm pretty sure > (not 100%, but close) that the T1 specification is only Cat3, since it's > standard BellCore wire and they don't run your T1 loops (which aren't T1 > anymore, they're DS1 over HDSL or HDSL2) in