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I moved the Asterisk box down to the master POTS sockets, and plugged the TDM410P directly in. No difference. The first outgoing lineworks, but the second fails, with three of this error: WARNING[1737]: translate.c:175 framein: no samples for gsmtolin Incoming calls work on the first line. On the second line, the external calling party hears a ringing tone, but there is no activity ...

Hi! I was looking for information about Sysmaster and I this is what I found!!!

Hey - just getting started with my new SwitchVox SOHO. I use Polycom 330 phones, internal all sounds good but when calling out via an analog line I can hear them with no echo but the start of each sentence is clipped. They can hear me fine. It's as if echo cancellation is too extreme. I have turned it down to 64 for tail length, agressive cancellation is not enabled and I'm using analog ...

Hi, I'm trying to implement a SIP attented transfer scenario as described in http://tools.ietf.org/html/draft-ietf-sipping-cc-t ransfer-09#section-7.3 . I'm using 3 SIP phones, A, B and C. B is registered with Asterisk and acts as transferor. A and C are not registered with Asterisk and act as transferee and target respectively. A calls B, B puts A on hold, then calls C. If C ...

For some days I have been testing different codes in asterisk.conf, but I cannot get it right for my purposes The goal is as following: Somebody calls me at my sipgate number in austria Then he shall hear a short beep. When nothing happens within one second, the call shall be forwarded to my sip hardware phone(it shall ring, so i can take the call) BUT, when the ...

Hi everybody, I want to picup this topic, because I have the same problem. I can click in the GUI on 'System Configuration', but I will ask for a login. And here is the problem: The admin-Account, which was created in the installation will not accepted here. It works on the Asterisk-GUI an on the Linux-Console-Session, but not here. In the setup it was only possible to give a static ...

Good afternoon, I am trying to setup an asterisk 1.6 since some time and still stuck with the logging of cdrs into mysql database like I do in 1.4. System is Debian Etch, Mysql is installed and running. I re-installed libmysqlclient15-dev twice, recompiled asterisk and addons but somehow the .so are not generated and therefor not loaded into Asterisk. During addons make/make ...

I need to add to user ability to cancel Orginate even before is start ring on his extension?

Hi Friends, I have a DEADAGI Script being called from asterisk. Everything is working fine, but for the music on hold. When I ask asterisk to play music on hold while I do some other things (DB connectivity etc), it just plays silence. I have observed the MOH classes etc and everything is fine. In face, the music is played if I replave AGI for DEADAGI in the script call from Asterisk. So...

Greetings I live in Ukraine and want to create a call centre with virtual number from USA and UK.Currently i'm looking for the software which will suit my requirements: - ability to quickly exchange incoming calls among call centre operators. - instant messaging. Can i create it using asteriks?

Hi everyone, i am new to asterisk - but im very close to getting this configured. My outgoing calls are working, but with no incoming audio, i also cannot get incoming calls working at all - im just getting the operator message at the provider. my clients are inside a firewalled connection behind a draytek vigor, i have nat rules to point udp and tcp ports 5060 to 192.168.1.13 (my ...

I have installed asterisk 1.6.0 release, asterisk-addon-1.6.0, Register user from sip.conf is fine. Regiser user from sip_buddies table, 404 not found. module reload res_config_mysql, get the following error res_config_mysql.c:855 mysql_reconnect: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. res_config_mysql.c:855 mysql_reconnect: MySQL ...

Hello there, I am a newbie to Asterisk, but have spent quite a bit of time reading reference materials and scraping the web for answers, and so far I can't get past this error, so I was hoping I could find some help here. Basically, I am running the AsteriskNOW appliance: OS Version: Linux srp-techserv.nrmdomain.com 2.6.22.13-0.1.gcc3.4.x86.i686 #1 Mon Nov 19 19:39:41 EST ...

I moved the Asterisk box down to the master POTS sockets, and plugged the TDM410P directly in. No difference. The first outgoing lineworks, but the second fails, with three of this error: WARNING[1737]: translate.c:175 framein: no samples for gsmtolin Incoming calls work on the first line. On the second line, the external calling party hears a ringing tone, but there is no activity ...