similar to: Connecting Asterisk to Siemens Hipath 3750

Hello! Is it possible to connect Siemens HiPath 4000 to Asterisk? What equipment required on Siemens side? I mean IP not E1. Sorry for asking here. Siemens-related websites use "salesperson language". There is no technical information.

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

Has anyone here successfully tried this? hipath 3750 --> hg1500 --> asterisk i'm not sure with the flowlines though. Thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060629/11f85674/attachment.htm

My new company is going to enter a bid competing with 3Com and Siemens solutions from others. Voice service is a PRI T-1 with a block of 100 DID numbers, a second T1 that will carry Integrated voice/data, 12 voice/768K for redundency. I asked this a week or two ago, now I have more details. Best/$ channel banks that work with Asterisk and Best deal on phones... 16 business, 34 basic IP

Hi Stephen, You said that PRI works great. We are using HiPath 3550 and Siemens digital phone which using *11, *97 etc for function keys. However Asterisk uses the the * key plus one or two 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

Hi all, I'm trying to setup Asterisk trunk to Siemens HiPath 4000 V2.01 What would be the best way to do so? I am a bit confused because as far as I've understand this PBX doesn't support H323, but I saw somewhere someone who created a cornet trunk and it worked using H323. So if anyone knows what I need to configure I would appreciate it. I've read some information

I am connecting to a siemens hipath 4000 with dahdi 2.1.0.4 and asterisk 1.4.23 using a Te210P card. the phone guy is saying that the lines are reporting always BUSY. however on my end the status shows OK. Anyone seen this? Is there something different about connecting PRI to siemens hipath? system.conf shows: loadzone=us defaultzone=us span=1,1,6,esf,b8zs bchan=1-5 dchan=24

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

Hi all, I have a connect between a siemens hipath & Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting "The number you have dialed is not in service" In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)

Hello, if somebody using this scenario in production successfully, please send me info, which ISDN card for asterisk server is usefull for me (Digium, Sangoma)? my crucial requirement is "caller id name" transfer/display between ISDN (Siemens PBX) and IP phone connected to asterisk I'm using PRI interface and Q.SIG signaling. thank you PJ

Hi, I saw your posting on Hipath and Asterisk.I have some doubts on the same.it would be really nice of you if you can help me out.My Doubt is as follows Currently I am using Hipath HG1500 V3.0 with Opticlient4.0. But i am not satisfied with the performance of Opticlient. I wanted to use SJPhone. Regarding this i had a talk with Seimens guys out here but they talk something ilogical. They told

Hi, I will like to know if anyone would guide me about how I can to interconnect one SIEMENS HiPATH 3700 with Asterisk. HiPATH have VoIP card and my idea is to do one un IP trunk between them so we would to transfer calls and services (voicemail, IVR,..) between both. We havent PRI ports unused in HiPATH so cheapest method of interconnection is one IP trunk. Any help or comment about will be

I am connecting 1.4.22 and dahdi 2.1.0.3+2.1.0.2 to a siemens hipath 300 and siemens hipath 4000. (2 channels to each switch) with a TE210p card setup as T1 with em_w. When the call is initiated to either switch the phone rings, when its answered then nothing... I hear no audio etc... After the timeout period the call is hung up. The phone switch 300 needs the T1 reset as the channel is not

Hi all, I want to use asteriks with my siemens hipath 1120 hardware. Is it possible? Thanks -- /** * @AUTHOR At?f CEYLAN * Software Developer & System Admin * http://www.atifceylan.com */ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101125/a5b9e9f6/attachment.htm

Hi there, our old pbx contract will expire soon and we have now 2 pbx companies trying hard to sell us a new VoIP one. I am usually a friend of open source software and read about the Asterisk poject, but I can not assess if Asterisk fulfills all our requirements. It would be great to receive some input from experienced users to be able to convince our management. My question is if the

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 am investigating possibility of using asterisk as an call center controller, i.e. Clients phone in, interact with IVR, if IVR is not enough get redirected to human consultant. There should be possibility for supervisors to connect to ongoing conversation. Expected traffic will not exceed 30 concurrent calls. Asterisk box should be connected to Siemens "communication platform"

Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted -- Called 2217008@Mediatrix Sep 6 16:54:54

Hello, I would like to use my existing HST Saphir V S2M PCI with Asterisk. Unfortunately I could not find any information about the usage with Asterisk. Does someone know if it's possible to use it? Do I need to do something special? Regards Julian Pawlowski