search for: sautter

...Macro | stdexten|443|SIP/zyx2000|187 | +----------+-----+------+-------------+------------------------------+ i also tried to put the parameter for the hint command into the 'appdata' column. my other problem is: how can includes of other contexts be accomplished? regards frank sautter

...over this bus), but i can't find any solution for digium cards. i've found a timing connector on the TE410P. can this somehow be utilized? is there another (software) solution? am i the only one with this problem (haven't found anything about this on the mailing list) thanks frank sautter

...able for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax feature (provided by Frank Sautter) - auto-config (compile and work with Asterisk CVS-HEAD and older versions) - use with ELinOS cross-toolbox and project handling For the versioning, I have decided to use the name extention 'cm' to avoid confusion with kapejod's version. This first release is 0.5 (not 0.1) because the...

hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission

hi, i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. The problem is, that the Bearer Capability (BC) together with the High Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the Service Indicator (SIN). The SIN is used to determine if the call is voice, fax or data. It's essential to set

...and clearing call disabled echo cancellation on channel 35 Set option TDD MODE, value: OFF(0) on Zap/35-1 Updated conferencing on 35, with 0 conference users Set option AUDIO MODE, value: OFF(0) on Zap/35-1 disabled echo cancellation on channel 35 -- Hungup 'Zap/35-1' regards frank sautter

I recently changed to all IAX providers for my DIDs, and none of them offer incoming caller-names. Back in the days when I did have incoming caller names, I found the names provided by the various phone-companies fairly useless -- "WIRELESS CALLER", "YOURCITYHERE" etc. Last Friday I finally set off to roll my own, in order to meet the following requirements: - uniformely

...rcecode of chan_zap and a little bit through libpri but did not find any variable containing the number. our ericsson BP250 PBX formerly connected to our E1 showed the callerid set by the caller correct. in the CDR of the ericsson both numbers appeared. any help is appreciated. regards frank sautter

...there anybody with experience on these pager devices? do they have a common standard? - does anybody know of a pager base station with an SIP interface? - does anybody know of a pager base station with an EuroISDN interface? what's your general advice on those paging systems? regards frank sautter

hi, after stumbling over the compile time flag in zaptel and after reading the new features of the 2nd generation firmware of the TE405P/TE410P, i was wondering if the cards are capable of upgrading the firmware in field? regards frank

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

One of the 3 ups's that I am trying to configure is a Tripp Lite SU5000RT4UHV. The ups is recognized by the IETF 1.4 MIB, but I am getting Unhandled ASN 0x80 and 0x81 when starting upsdrvctl with the -D option. When I do a upsc {name of ups} the message returned is Error: Connection failure: Connection refused. Thank you for your support V/R Bill -------------- next part

...nom220-boot2.1.bin Firmware: http://www.snom.com/download/snom220-3.52-beta-SIP.bin he functions keys are configured: fkey5!: dest <sip:180@snom.info;user=phone> how can the blinking state of the leds be achieved? is this a firmware version issue of the 3.52 i'm using? regards frank sautter

OS: Redhat 9.0 Kernel: 2.4.20-8 Asterisk: latest CVS head Hi all, I try to compile the latest chan_capi, but it won't work. I also tried the patch that is arround there from frank sautter and I have installed the asterisk header files. Asterisk is working very good :) and the Fritz AVM pc Card is running. capiinit: 1 fcpci running fritz-pci A1 3.11-02 0xbc00 5 lsmod: fcpci 587744 1 capi 20896 0 kernelcapi 33792 2...

hi, since my latest libpri update i get these messages: !! Unable to handle ROSE operation 36 !! Unable to handle ROSE operation 30 i searched through ITU X.219 and X.229 but can't find any values for the Remote Operations Service Elements. are these AOC-E messages? regards frank

...hed the chan_capi to let it compile under the new CVS head Give it a try please You have to start from the original chan_capi http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz and then apply the patch http://www.c-net.it/chan_capi.diff.bz2 it also includes the fax patch from Frank Sautter Let me know please Sergio

...xes work from europe to north america - the telcos have the same problem)? - which signalling protocoll will be used on the T1 side? is asterisk translating this correctly? - btw. where is the different bitrate coming from? is it 7bit T1 and 8bit E1 or 7kHz and 8kHz sample rate? regards frank sautter

Hello, we are a firm who wants to develop some VOIP solutions. The first infrastucture we choose for development is: - an Asterisk machine connected to a traditional PBX (s0). In this way people is not (yet) obligated to migrate its extisting PBX (and analog phones) to VoIP. - The PBX will be then configured to redirect specific outgoing calls (i.e. a remote branch office) to Asterisk, that

hello everybody, one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco <---pri---> asterisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with

Hi! First I have to say, that I'm not very familiar with CVS and patching. I tried to patch & compile CVS-HEAD. First I checked out zaptel, libpri and asterisk with this command: "cvs co zaptel libpri asterisk" But the latest patch sipsubscribe-20050812.rev806v2.txt from http://bugs.digium.com/view.php?id=3644 didn't worked, so I tried to check out an older CVS-Versions