Re: [asterisk-users] DTMF lose with TE-121F

Problem solved turning off echo cancellation. Any known bug? Pepe Aracil escribió: Hello. I'm using asterisk in alarm reception system. The system is DTMF intensive and works well while all concurrent channels are online. But when one channel goes hangup the other channels lose tones

[asterisk-users] DTMF lose with TE-121F

Hello. I'm using asterisk in alarm reception system. The system is DTMF intensive and works well while all concurrent channels are online. But when one channel goes hangup the other channels lose tones while one second. ___ -- Bandwidth and Colocation

Re: [asterisk-users] DTMF lose with TE-121F

Hi. I probed more tests and I detect when a channel goes on-hook or goes off-hook in the other active channels I listen a short noise or distortion. I attempt to select internal clock source from TE121 but with the same results. Thanks Pepe Aracil escribió: Hello. I'm using asterisk

[asterisk-users] softmodems bank for ast.

Hi. We need a full featured modem bank 20+ to attend data calls. IAXmodem only supports fax protocols because spandsp only support fax protocols. The idea is to do a IAX wrapper like IAXmodem but with a full featured (but propietary) softmodem library like PCTEL or linuxant. I hate

[Asterisk-Users] Console detach.

Hello. I have installed asterisk 1.0.9.dfsg-5 in debian sarge. if I run /etc/init.d/asterisk stop and then /etc/init.d/asterisk start . Asterisk don't detach from console where i started it. It beguin to write all warning,debug,AGI dialog,... to the console. If I start ast. manually without

[Asterisk-Users] RTP traffic

Hello. How can I check if the RTP traffic between two channels is bypassed? Some * console command? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] app_senddtmf.so.

Hi. I have installed asterisk 1.0.7. I need to send dtmf with tone duration. This functionality is in the cvs version. Can any body send me the app_senddtmf.so binary compiled for i386 or pentium IV to replace the 1.0.7 version.? I want to preserve the rest of 1.0.7 version of asterisk,

Re: [Asterisk-Users] sip.conf match

and 22 it works as follows: [sip-in] exten = 11,1,Noop(First number dialed) exten = 22,1,Noop(Second number dialed) --- MARK. Pepe Aracil wrote: Hello. I have two hired pstn numbers with the same voip provider. I want to distingish in the sip.conf file

[Asterisk-Users] sip.conf match

Hello. I have two hired pstn numbers with the same voip provider. I want to distingish in the sip.conf file, what of two phone numbers was dialed, but i don't know how to do the match, because the sip client and the sip host are the same for both numbers. How can i match in sip.conf by the

[Asterisk-Users] AGI kill

Hello. When the caller hangup the phone, asterisk kills my AGI python script without notification. I caught all signals, but none was trigered. How can i trap this event to resume some operations. Sorry for my poor english :) Thanks. ___

[Asterisk-Users] sip.conf entry precedence

Hello. I'm new in the list and sorry for my poor english :) I have this two entrys in the sip.conf file, one for incoming calls (vtele_in) an the other for the outgoing calls (vtele_out) -- piece of sip.conf --- ; entry for incoming calls [vtele_in] type=user context=sip-in host=voztele.com