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I install ser and found my ser don't support mysql. my ser version : ser-0.8.14_src.tar.gz and ser-0.8.14_linux_i386.tar.gz who can help me? thanks. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:Zigang.Zhao@alcatel-sbell.com.cn

I use xlite , but it isn't support video when it is free. who used better softphone ? Thank u. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:Zigang.Zhao@alcatel-sbell.com.cn

I want to build a SER in China, but I don't build a PSTN gateway in China. because goverment don't give permission in China. I want to look for a cooperater, let's my user can dial PSTN to world. if you are interested in my idea,please mail to me. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:Zigang.Zhao@alcatel-sbell.com.cn

my means , how could use asterisk and ser in same box. my ser support mysql database , so whether asterisk don't config user in sip.conf ? and how to do I should ? thanks a lot. and I want to agent asterisk product in China Mainland, who can contect me. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:Zigang.Zhao@alcatel-sbell.com.cn

I am in China , I cann't buy digium card. I want to resales asterisk in China for chinese enterprise. who can give a card for test ? I only hope COD. I hope buy a TDM400 and a FXO . Thank u. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:Zigang.Zhao@alcatel-sbell.com.cn

I have a system include asterisk + ser. when I want to limit a dial out to pstn , I will do that : extensions.conf exten => _9NXXNXXXXX/myaccount@sip.com,Congestion exten => _9NXXNXXXXX, 1,Dial(ZAP/g2/{EXTEN:1},30,t) exten => _9NXXNXXXXX, 2,Hungup but I don't confirm is it right. I have no env to test it. who can help me?

I don't know what's means about register in sip.conf such as: register => user:secret:authuser@host:port/extension even if I registe a sip proxy , but how use it ? I think : when incoming from sip proxy to asterisk : user a --> sip proxy --> asterisk --> pstn sip proxy : SER in ser.cfg farword ( "192.168.0.10" , 5060 ) this will forward a call to

Greetings. I did some digging with Google, the wiki, and on the archives, but didn't find a recent conclusive answer. If this is answered in the wiki or archives somewhere, please point me to it. I'm in the process of setting up an Asterisk box for home use. I've got a X100P card on the way. I've not decided what analog adapter(s) to get yet. The only phone service to hook up

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

Hello all, I want to use asterisk pbx to give a ring for sip user.when A call B , user B 's mobile will ring.(B always register his sip number and his mobile number first.) ignorepat => 9 exten => _9NXXXXXX,1,Dial(Zap/g2/${EXTEN:1}) exten => _9NXXXXXX,2,Congestion but I want only let B's mobile ring,B can't access. or when B access,the phone auto hang up. what I should

I want to use asterisk dial to PSTN,but only dial,don't connect. when you hear ring,you only can hungup,don't connect. when you connect , asterisk will disconnect . who can tell me what write extension.conf? Thanks.

quesion 1: If I write extension.conf exten => 100,1,dial(sip/100,20,L(0)) I will listen ring but I don't access when other one dial 100. who have this experience ? quesion 2: and I look at this extension.conf in voip-info.org exten => _908.,1,Dial(Modem/ttyI0:${EXTEN:1}) Could this modem dial to PSTN? and will I install dial program for the modem? and when I write : exten

I do this test : exten => _[123456789],1,Dial(SIP/${EXTEN},20,L(0)) exten => _[123456789],2,Hangup when I use 12345 dial 12346 , it should be hangup. but it don't link I think. why?

I will build a voip network by use SER+asterisk in China. but I don't connect my sip network to pstn , this is don't be permission in China. if you have a asterisk and have pstn access, then you can contact me about copartner. I will let my user that he want to dial to pstn will use your pstn . and I need your rate to pstn and I only hope can get a little part. I hope negotiate about it.