similar to: HiPath 4000 and Asterisk

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.

For all who are interested: A quick review of CeBIT 2005. :-) CeBIT was a very successfull event. Most of the time, the asterisk-booth was crowded with more people than we could talk to. We had with us a demo-installation including different IP-phones, digital and analog phones as well as a Siemens HiPATH PBX to which our Asterisk-server served as a VoIP-gateway, and many people were impressed

Hi, I am getting repeated messages in my logs with the following: ********************* Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: 70975D7A-CF70-4C05-8F21-56B06195995F@10.0.0.40 Feb 23 07:56:12 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23

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

I downloaded asterisk to use cvs to checkout the release version. After installing, I would like to load module chan_h323.so but there is some error : *CLI> load chan_h323.so Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource: /usr/lib/asterisk/m odules/chan_h323.so: undefined symbol: __use_ast_pthread_create_instead__ Unable to load module chan_h323.so *CLI> How can I solve

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:

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?

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everybody, after looking a bit around the Internet and checking which website style some users of webgen are using, I found that many people just stay at the default webgen website style (which is not really pretty ;-) So, I''m thinking about changing the default webgen website style to something more Web-2.0-ish, and I would like to know if there is a particular website style

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 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

Hello all, I am trying to connect asterisk to a Siemens Hipath 3750 PBX system. I have a physical connection issue. I know that I should use a crossover RJ48 cable to link the two systems. The problem however is that the physical interface of the Siemens system is very unfamiliar. From my digging around, I think that this is an S2M interface. http://www.mail-archive.com/asterisk-users at

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

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

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