#asterisk - Tue 17 Apr 2007 between 13:35 and 13:47

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_VoiceMeUp_COMtrue
Qwell[laptop]_VoiceMeUp_Com: f that didn't work, you'd never be able to upgrade things like glibc, or your kernel, or rpm
unlike...*cough*windows*cough*
_mike3_I'm looking for a good web source with third party asterisk addons. Eg: Alarm. Anyone have a site?
alexnstried lsattr ------dA----- ztcfg
need some ideas
Dovidhi guys. i have an off topic question. my provider is doing authentication based on IP and i am trying new equpiment on a DSL connection that gets changed. I went to send the calls thru a linux box. anyone know of sofware that will accept data on a certain port and then just pass it along on the same por to a diffrent iP ?
[TK]D-FenderVec: Executing [s@macro-zapDial:1] Dial("SIP/fax2-9bf51600", "Zap/g1/0118845293") in new stack
DovidIP*
[TK]D-FenderVec: See anything missing in there concerning the ABILITY to TRANSFER?
Vec[TK]D-Fender : yeh, I am trying to transfer the call I initiated, from the SIP/fax2 ?
its a phone not a fax, just called it fax
[TK]D-FenderVec: think about what OPTIONS you have to pass DIAL to allow the caller to TRANSFER calls...
Vec[TK]D-Fender : oh yeh, dumb, sorry tT, errr
thanks
I should have seen that
maviorhello everybody,i have some problems with my phones and my "r" buttons that seems to be related to that problem http://www.asteriskguru.com/archives/asterisk-users-flash-hook-hangup-problem-vt30039.html?highlight=flash+button , now i want to use the callwaiting feature, anybody can say how can i set a simple extension to flash my channel to achieve the same behaviour as I pressed my...
...hook/flashbutton ? tnx
_VoiceMeUp_COMso what does this error mean ? tehcnically
DeeJayTwoI have canreinvite=yes in sip.conf.. when two sip phones get on a conversaion
a rtp debug IP show the rtp packets...
mavior?
etfonhomeyCan anyone recommend a good ATA with at least 2 FXO ports?
[TK]D-FenderDeeJayTwo: Pastbin the CLI output of a call at verbose 10, and then do "show channels concise" followed by show channel [channel]" for each leg of the call.
DeeJayTwo: ^^^^^ I asked you this a LONG time ago....
_VoiceMeUp_COMok i found out
Apr 16 14:52:48 WARNING[15726]: chan_sip.c:1084 __sip_xmit: sip_xmit of 0x8713fd8 (len 893) to 12.3.12.1.123:0 returned -1: Invalid argument
mavioranybody flashing here ? :P
Mercestes_VoiceMeUp_Com, Ok, what does it mean? The suspense is killing me.
_VoiceMeUp_COMthis means.. the PORT is bad.. the user has a default ip addrss in case of not online..
but the asterisk is maping in memory has default :0
Mercestes_VoiceMeUp_Com, So 5060 does not exist on the specified peer?
_VoiceMeUp_COMwhy choose port 0 if theres an ip in htere.. the default should be 5060 at least no ?
Mercestes_VoiceMeUp_Com, or whatever port it's trying to send to
_VoiceMeUp_COMit exists
when he regusters its addr :1.2.3.4:5060
but default 1.2.3.4:0
sip show peer blah
Mercestes_VoiceMeUp_Com, So your sending a call to an invalid peer?
_VoiceMeUp_COMyes

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