similar to: problem : undefined symbol.

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

hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able

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! 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.

Hello all, as good? It would like to make a question, asterisk supports the protocol qsig, for interconnections in ISDN with equipment Siemens HiPath 4000 or same Ericsson MD110, so that it could identify to the name and the number of hosts and also to use some features of asterisk in the Siemens/Ericsson equipment. Greetings Josu? -------------- next part -------------- An HTML attachment was

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 Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josu?

I have read most of Eicons information on Q.SIG, and I am able to load the Q.SIG protocol (instead of ETSI for example). No strange logging in divacrtl mlog. But how do I tell Asterisk to understand Q.SIG? My PBX is configured for QSIG, but I cannot see anything on my trace when trying to make a call via the S0(Q.SIG) Janne > -----Ursprungligt meddelande----- > Fr?n:

> > Great service. Hope it catches on. > > > > In time there will probably have to be some adjustments: > > > > - ratio 10/1 - this means everyone can make 10 calls for every call > donated. > > This doesn't add up in the long run. > > - I get credited 10 calls for every call I make to my own dialplan. This > > should be fixed. > > -

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

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

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

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 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, 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"

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)

Guys, I need to make asterisk connect to another device over a pri using QSIG signalling (master/slave). Is this feasible and has it been done before? Thank you! al. _________________________________________________________________ Get ready for school! Find articles, homework help and more in the Back to School Guide! http://special.msn.com/network/04backtoschool.armx

Hello. I am going through the documentation and trying to find if asterisk can help me in my case. It is quite difficult to find answer because I do not know the exact question. I have two location. Each in different country. Both locations have Siemens HiPath - different type and software. I can not use card that would allow me to connect those PBXs using SIP. But I have some free ISDN and