[asterisk-users] chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9

happens when I sent call to my quintum gateway server, the status appears as soon as the call get connected. cheers Aby Azid ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or

Re: [asterisk-users] chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9

Thank you for replying, How would i know, whether i have the valid indicitions.conf ? On Sun, Apr 20, 2008 at 8:47 PM, Eric Wieling <[EMAIL PROTECTED]> wrote: > Make sure you have a valid /etc/asterisk/indications.conf > > aby azid wrote: > > Hi, > > > >

Re: [asterisk-users] chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9

Hi Eric, i copy the indications.conf.sample from the asterisk source and paste it in the /etc/asterisk directory. I reloaded asterisk and still the message appear when i sent call to Quintum. Am I doing it right? cheers, Aby Azid On Sun, Apr 20, 2008 at 11:54 PM, Eric Wieling <[EMAIL PROTEC

Re: [asterisk-users] chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9

nnected to SIP Proxy and to quintum gateway. In this case, I'm using Wellsip from Welltech. cheers, Aby Azid On Sun, Apr 20, 2008 at 11:54 PM, Eric Wieling <[EMAIL PROTECTED]> wrote: > Use the indications.conf.sample that comes with the Asterisk source. > > aby azid wrote: > >

[asterisk-users] G723 pass thru

Hi, I have softphone with a g723 codec, my question is how do i set it as Pass thru in Asterisk? cheers, Aby Azid ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] G723 pass thru

told me to use pass-through. cheers, Aby Azid On Thu, Apr 24, 2008 at 10:42 PM, Anthony Francis <[EMAIL PROTECTED]> wrote: > More importantly, for it to "pass-through" you need something that > processes g723 on the other end. If Asterisk is terminating the call by > hand

Re: [asterisk-users] chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9

hi, yes, apparently the status message appeared due to the codec setting from the gateway. cheers, Aby Azid On Thu, May 1, 2008 at 5:34 PM, Cyril SCETBON <[EMAIL PROTECTED]> wrote: > Hi, > > Did you resolve the issue you were hitting ? > > aby azid wrote: > > hi, >

[asterisk-users] -zapg729toulaw did not update samples 160

samples 160 [May 9 12:53:39] WARNING[3626]: indications.c:149 playtones_generator: Can't generate that much data! [May 9 12:53:39] WARNING[3626]: translate.c:211 framein: zapg729toulaw did not update samples 160 *Thank you in advance, Regards, Aby

[asterisk-users] - Failed to authenticate user

Hie, I managed to connect two Asterisk box via SIP. My problem is when I login using Realtime SIP, I will get chan_sip.c:8373 check_auth: username mismatch, have <8003000777>, digest has Failed to authenticate user "8003000777" >;tag=as4d11916d when trying to send call to the 2nd server. bu

[asterisk-users] -

Hi, I have question regarding Asterisk Local channel. Is it possible to define codec used in Local channel as like in SIP channel?. If it's possible, how do i do it? Thank you Regards, Aby Azid ___ -- Bandwidth and Colocation Provided by

[asterisk-users] Fwd: - Asterisk Local channel

Hi, I have question regarding Asterisk Local channel. Is it possible to define codec used in Local channel as like in SIP channel?. If it's possible, how do i do it? Thank you Regards, Aby Azid ___ -- Bandwidth and Colocation Provided by

[asterisk-users] Asterisk stress call test

th ideas or tools for me to achieve this. Cheers, Aby Azid Vyke Asia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mail

Re: [asterisk-users] Asterisk stress call test

ls on the method you used. Cheers, Aby Azid Vyke Asia On Fri, Aug 15, 2008 at 1:45 PM, Saul Bejarano <[EMAIL PROTECTED]> wrote: > Remember the rule of 30Mhz per call when you kill the machine and also > the bandwidth usage on connected calls. > > Kind regards, > > Saul Bejara

Re: [asterisk-users] Asterisk stress call test

there such tools and ways for me to create simultaneous AIX2 calls?. Again, really appreciate if anyone can come up with ideas or tools for me to achieve this. Thank you in advance, Regards, Aby Azid ___ -- Bandwidth and Colocation Provided by http