[Asterisk-Users] unsubscribing

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[Asterisk-Users] SIP client auth

Dear list, does any one know how to do a SIP client auth via central database instead of specifying in the sip.conf ? if we could do with central database, should we use RADIUS or other better way to do it. Thanks, George ___ Asterisk-Users mailing

[Asterisk-Users] manager.conf

Hi, Do you know if we can use AGI or other script to handle the asterisk events by using the existing asterisk manager process ? Please advise. Thanks George ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/list

RE: [Asterisk-Users] GSM compression tool

Thanks Steve. In fact, I am looking for a ZIP tool to zip a GSM file. currently I found that winzip ONLY compress 10% of a WAV file. I am wondering is there any good ZIP tool for a GSM file and or WAV file. Thanks, George Lin. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL

[Asterisk-Users] GSM compression tool

Hi list, Can anyone suggest us what kind compression tool is best to compress a GSM file. And what kind compression ratio can be? Thanks, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

[Asterisk-Users] FW: Identify the channel ID within a SPAN

the dial,zap/g1/BYEXTENSION to let asterisk choose the available channel, and once the call gets answer, can I know which channel is used for the call, is there any application I can use to figure out when the call is answered ? Please advise. Thanks, George Lin

[Asterisk-Users] SIP problem with asterisk

/Transaction Does Not Exist and I specified in sip.conf [6002] type=friend host=dynamic nat=1 qualify=yes [5009] type=friend host=dynamic nat=1 qualify=yes Can anyone help me what can be wrong ??? Thanks, George Lin ___ Asterisk-Users mailing list

[Asterisk-Users] Asterisk and ACD system

obtained via PBX CDR or signalling between pbx and asterisk or asterisk can know from the call set up ?? Regards, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] MGCP question

message ??? Thanks. George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] H323/SIP gatekeeper

Hello List, Does asterisk H323/SIP allowes me to conditionally use diff gatekeeper to route the call ? e.g. for the call to germany, I want to use gatekeeper1, and for the call to UK, I want to use gatekeeper2. if yes, where and how to specify in these .confs file ? Thanks, George Lin

[Asterisk-Users] DTMF SIP

the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind configure for the SIP2 as well or I can configfure SIP2 with inband DTMF in sip.conf ? Thanks, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED

[Asterisk-Users] CODEC & DTMF

, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP NAT question

Forgot to mention that we have specified the nat=yes for all sip entries in sip.conf. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of George Lin Sent: Wednesday, August 13, 2003 10:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP NAT

[Asterisk-Users] SIP NAT question

ds the 211.x.x.x:port-number to the office router ?? 3. If it is the office router's responsiblity, what should we configure the office router even there is no firewall??? Please advise , and thanks alot. George Lin ___ Asterisk-Users mailing list [

[Asterisk-Users] cisco5300 with asterisk through H323

Dear all, Can some of you give us some suggestion how to configure the asterisk in order to make a call to cisco5300 in g729a codec. And how to confiure the cisco5300 part in order to receive a call from cisco5300 via h323 g729a. Your advice /help will be highly appreciated. Thanks, George Lin

[Asterisk-Users] SIP and H323 warning message

[131091]: File res_parking.c, Line 209 (ast_bridge_call): Bridge failed on channels H323/ip$67.64.223.74:33136/20735 and H323/67.64.223.76 Can someone help me or explain me what is about these warning messages. Thanks, George Lin ___ Asterisk-Users

RE: [Asterisk-Users] Channel Banks

Hello all, Can someone point me where I can buy a E1 channel bank ( incluidng model and vendors ) which is compatible with digium E400P card. Thanks, George Lin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, April 10

[Asterisk-Users] Fax support over MGCP, H323 and SIP in asterisk

Hello, Can someone tell us if current asterisk supports FAX over MGCP, SIP and H323 ?? Thanks, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] NAT=1 questions

to understand what NAT=1 mean to sip-phone ip address or to asterisk ip address or to both ??? Please advise. Thanks, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] VoIP Gateway Performance

Lets say G.711. So what is maximum number calls between PRI-H323 ??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, March 27, 2003 9:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP Gateway Performance On

[Asterisk-Users] SQL

Hello everyone, I would like to have soneone who knows how to use SQL method to update asterisk's conf files, without disrupting the ongoing calls. Can someone give me some sample about using SQL to update conf files. Thanks George Lin ___ Ast