similar to: siemens pbx what i ask techinician?

hi all, anybody have a siemens hipath 3500 with a sm2/pri card? because i need to connect to my box TE110P (e1) and i dont know how is the mode in the pbx to change it. thanks -- .- Pablo Allietti LACNIC

hi all i have an asterisk working and i need to add a mettme public service. for example i need to download a soft (sjphone) and without any configuration call to 509@asterisk.mydomain.com (meetme) and join a conference but when i do that i received an error saying nomber do not exist. but if i call a extension is work propperly. in the extensions.conf have exten => 411,1,Answer exten

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 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

Hi All, I was able to insert some extensions in Mysql DB and use them successfully. In Mysql extensions table the priority column is of type tinyint and when I give 's' value for it, it is not accepting that value as it takes only tinyints. Please tell how can I make that column accept values like t,s,i and make it work with asterisk in realtime without any problem? If I change the type

I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4. ------------------inbound call

After reading this article: http://www.theregister.co.uk/2005/08/31/blocking_chinese_ip_addresses/ I got to thinking that there is really no reason for *any* traffic to hit my servers that comes from anywhere outside North America. So I wrote the perl script at the end of this posting to extract selected IP ranges posted at iana.org and convert them into iptables rules blocking any traffic

I've just ordered a Siemens Gigaset C450 IP cordless IP/DECT phone, given that it's supported by asterisk http://www.voipuser.org/review_41.html However, I see that a slightly better Gigaset S450 IP is available for only a slight price premium. Are there any user experiences with the S450 IP? -- Eugen* Leitl <a href="http://leitl.org">leitl</a> http://leitl.org

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 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

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

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, Any recent experience to share when using OpenStage phones in SIP mode and Asterisk ? What about provisionning (and localization) ? BLF ? Audio quality ? User acceptance ? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101222/79c2b6bc/attachment.htm>

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

Hi folks, I wonder if any of you out there are using Siemens S685IP base station(s) (with S68H handsets) on Asterisk and experiencing problems with SIP registrations where the SIP extensions do not ring and peers become unreachable after a period of time. Symptoms are rather sporadic, but as described, SIP extensions being unreachable from Asterisk perspective. Also experienced 'not

Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the "soon-to-become-available-in-the-uk" S685IP. Both systems have great feature sets and, on-paper at least, look to be the bee's knees. Anyone got any skeletons on them? Thanks Alan -- The way out is open!

Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext.

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 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

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how