similar to: why I don't do this test ?

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

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

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

May anyone help me solve the problem? I use samba 2.2.2 on Solaris 8 ngnvob02 [** NONE **]/export/home/sitlb $ cd /usr/local/samba/bin ngnvob02 [** NONE **]/usr/local/samba/bin $ ./smbpasswd Old SMB password: New SMB password: Retype new SMB password: machine 127.0.0.1 rejected the tconX on the IPC$ share. Error was : = ERRSRV - ERRbadpw. Failed to change password for sitlb But I can change the

Hi, after having read so much about Asterisk, I went on and tried out to create a little sample-setup. I'm using a Fritz Card USB with the AVM Capi Driver and two X-Lite Softphones. Dialing between the softphones makes no problem. Calling the MSN fron an external phone also works. I'm getting to the asterisk demo-voicebox which works flawlessly. Now may next step has been to enable

I have seen a number of people in this newsgroup asking for information regarding asterisk interworking with Vocal. I was able to configure Vocal and Asterisk so that calls originating from vocal can land on an extension in Asterisk. I would like to share this info with the group The scenario that I tested was as follows. A call was originated from extn. 1001 on Vocal and the call was made to

I was always puzzled by posts to the list about people having problems getting hints to work on a Snom, since I always seem to have no problem making it work. That is, until today when I tried to get a sidecar to work. All I could do was get a monitored extension light to light up continuously, regardless of state. Frustrating! Going back to my working dialplans where I got 1 or 2 lights working

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

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 am blessed with this user forum and able to set my Dev-PCI Digium card working fine with the Asterisk server of mine (i)But today i just wanted to know if someone can help me to set X-Ten Lite to call PSTN line using my FX0 Currently , I am able to use X Lite to call another X lite user locally (LAN) I also has attached my setting together Thanking you all in advance --------------

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

Hi, I need to create a simple number checking for authorizing the calls. if a person dial 91800555121212345 where 12345 is the authorization code. If the authorization code is correct the call will go through if not it will play something saying wrong authorization code or just hangup. This my dialplan to get the authorization code AUTH=12345 exten => _9.,1,Answer() exten =>

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,

We've discovered that although Winbind supports password changes when the account password is expired, this only works with *interactive* shells. This is a major problem for us. Use case 1: SSH tunnels: $ ssh user2@localhost -N -L 4711:localhost:22 user2@localhost's password: <trying to use the tunnel> channel 2: open failed: administratively prohibited: open failed As you can

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

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

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

Hi, all, I met a problem for several days, any suggestion is really appreciated!!! I'd like to do autodial using Asterisk. For example, I have a file under /var/spool/asterisk/outgoing, which include: channel: zap/g1/12345 MaxRetries: 0 RetryTime: 60 WaitTime: 20 Context: default Extension: 2222 Priority: 1 And in my "extensions.conf" file, I have [default] exten =>