search for: transfercap

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb

Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf? I'm changing rxgain in zapata.conf, and reloading zaptel, but sound level on ISDN(HFC) is always the same (loud).

Hi, I installed Digiums Free Fax for Asterisk and found out, that it automatically retries failed faxes, is there a way to stop that? Thanks Markus

...chan_capi: root@localhost:#make ./create_config.sh "/usr/include" Checking Asterisk version... * no 'struct ast_channel_tech', using old pvt * ast_dsp_process() without 'needlock' * no 'struct ast_callerid' * found 'struct timeval delivery' * no 'transfercapability' * no 'ast_config_load' config.h complete. gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/i nclude -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DASTERISKVERSION=\"\ " -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o...

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I am trying to set up asterisk and hylafax to send and receibe fax. The machine is connected to the PSTN is an Eicon Diva BRI (1 BRI port). My problem is that , when I send a Fax from the PSTN to this machine, the asterisk or diva or hylafax, does not detect this call as a fax and asterisk answer that call like a voice call. How sould I configure the software modem (iaxmodem) to use

Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's

Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When