search for: hicom

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 t...

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 on the link above are similar somehow to the Siemens I mention in terms of integration with Asterisk? Answers much appreciated. Martin

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...

I have a strange problem with overlap dialing. I installed an asterisk server between a Siemens HiCom PBX and our telephony provider. Everything is working fine except some strange problems with the dialing of the fax (connected to the HiCom PBX). It seems to me that if dialing takes too long Asterisk just hangs up the channel without recognizing that the fax machine is still dialing: (Fax gets H...

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 E1 line f...

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 ca...

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 recognized as faxes and * tries to forward the call to "fax". I do not answer this calls... == Spawn extension (Amt595xxx-In, 595164, 1) exited non-z...

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 => 022956388,1,Goto(callce...

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...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: --

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 work fine calling a URA menu on the Asterisk and can...

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 | | SIP Clients and many other features The Siemens is configured as if it would be directly connected to the carrier. Could you help me with exemples of zap...

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 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...

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

...n my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is connected. Fax passing thru this connection are not getting thru and always getting drop. FYI: all of the 8xE1s are currently up with two using mfcr2 and the rest is ISDN pri. Server is spec is Dual Xeon 3.00GHz 2 GIG RAM Load Average: less than 1...

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:

...digits for function keys as well(it is common key combination for functions). So is it any way to disable *11, *97 keys in HiPath system and pass this keys to Asterisk? Thanks and regards, Isaac >Hi Victor > >Looking for the same answers here too. We are regional distributors for >Hicom HiPath in this part of the world and until now we are still >waiting for chan_cornet to come around. So far we have successfully >interconnected via BRI (mISDN) and PRI (Zaptel) and it works great. > >Let's see if it's too good to be true soon. > >Best regards, >Ste...

Hi I've got an Asterisk box connected to a Siemens Hicom 300 using a Digium TDM400. The Hicom provides the calling extension as DTMF at the beginning of the call followed by two *, as in 3425** when 3425 calls my extension, I can hear all 6 tones if I have a handset connected but using Asterisk's Read application straight after Answer() Asterisk usua...

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