search for: voljatel

Displaying 6 results from an estimated 6 matches for "voljatel".

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2003 Jun 17
3
sip.conf
HI, can somebody tell me how and where must I put the SIP register line? I think is in [general] section of the sip.conf and that I have to put: register => user:password@host:port/localextension but, user and password of the SIP gateway? Because I'm trying this and doesn't work... thanks a lot in advanced michelle ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam
2005 May 12
4
gnugk
Hi I've a problem with a gnugkv2.0.7 I've compiled gnugk successfully I've installed PWlib-1.6.6 and openh323-1.13.5 libraries successfully When i run gnugk i have this error: error while loading shared libraries liboh323_linux_x86_r.so.1.13.5 cannot open shared object file No such file or directory I try to use command export: export
2003 Apr 29
28
H323
Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm
2005 May 16
1
SIP-->h323 conversion
Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sip----ASTERISK----h323-----GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could
2009 Oct 16
2
Invite after bye?
Hi there noticed a strange thing in asterisk 1.6.2x 1.6.1x after one of the clients sends bye asterisk first sends invite to other side then after 200 ok it sends bye I am not sure but that could be some missconfiguration issue or a bug? so it's like this: side A sends bye to asterisk, asterisk responds with 200 OK to side A, then it sends INVITE to side B, expects 200 OK
2009 Oct 22
2
carefulwrite: write() returned error: Broken pipe
Dear, I am getting this in CLI on release candidate version of Asterisk. Any ideas, or points where to look? -- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi [Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe -- <SIP/916-fc001968>AGI Script rad-auth.agi completed, returning 0 Best regards, Josip