similar to: Pass-through Call Recording Transfer Information

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, I have a connection between a te120p and a siemens hipath which was recently set up. The connection uses etsi / e1 with no crc. Every few hours the asterisk shows the pri as in alarm asterisk01*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN and I have to run ztcfg to clear the alarms asterisk01*CLI> pri show span 1

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

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 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, 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

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, I just received a couple SX440isdn phones and was wondering if they can be plugged into a Diva 4BRI port in NT mode and work with asterisk+chan_capi? I know they probably work fine with mutliHFC cards with either bristuff of chan_misdn but since I have some spare Divas, I was curious about their potential as phone ports. The Diva's 3 and 4 ports are already set to NT mode at boot

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)

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, On a Asterisk 1.8.12 system working OK for months (>100k calls proceed), users are complaining for bad audio. My setup is: PSTN <--E1/PRI ---> Asterisk <--- E1/PRI---> Siemens HiPath <---E1/PRI ---> PSTN asterisk -rx "dahdi show version" DAHDI Version: SVN-trunk-r10414 Echo Canceller: HWEC asterisk -rx "pri show version" libpri version: 1.4.12 A

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 have successfully configured DIGIUM card and successfully communicated through it to the another E1 card running application. Can anybody tell me does TE120P support MFC/R2 protocol. Thanks and Regards sanchal singh

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 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've bought a R2Adapter to convert R2Digital to ISDN. I am trying to config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue? Daniel

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"

Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P (same problem with various previous versions; same problem with different TE120P cards). The customer has a partial (10 B-Channel) PRI that when it is busy (eight or more B channels in use), tends to fail as shown below... [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown error 500 [Jan 26 23:00:31]

Hi, Our potentiel next phone provider ask me a question i can't answer for sure, maybe someone here knows ? He says that is equipement only support VN4 protocol or more, or ETSI, however i can't find matching terms in the digium documentation or the chan_dahdi/dahdi/system.conf files... Any idea ? regards