similar to: help me : about dial to PSTN

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 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

Hi all, I'm running Asterisk since two days, and it's really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me?

Hi, I have just put in a tdm400p with 4 fxo modules and am trying to dial out from x-lite to dial my mobile phone just to test. The output in the asterisk console is like this Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack -- Goto (mobile,61400039953,1) -- Executing Goto("SIP/2002-239b", "localcall|61400039953|1") in new

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

Calling from PSTN let extension 601 ring twice, hang up and starts over again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SIP/102,20) exten => s,3,Voicemail(u102) exten =>

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

Hi, I have this setup: E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones Can someone tell me what's wrong with this call initiating from an analog phone connected to Alcatel PBX? It dies with NOANSWER but all works if I call other destination numbers. Dialplan is a simple Dial(zap/g1/0984465691) statement. At the end you'll find also zapata.conf.

Hi, I start to be a little confused so I am asking to the list. I want to make with * a gateway from PSTN to H323, and to send all incomings call to a predefined IP, which will treat the h323 calls. let's assume that all my incoming numbers starts with 00 here is my extensions [incoming] exten => s,1,Answer exten => _00.,1,Answer exten =>

I installed this card, everything work, i can make call and receive call with no echo and great sound quality, but after between 5 to 50 secs the call disconnect by itself, in the log i don't see nothing revelant. I don't share any IRQ, zttest show me values between 99.98 and 100. The only thing i see, the pstn line is not realy a true pstn line, it's plugged in an Arris Telephony

Hi all, Im having problems using a fax machine conected trough a grandstream 286 sip ATA, it must be able to send and recive fax from pstn, but fax always ends with communication errors 252/244/232 and others. Im using alaw/ulaw codes on pass trough mode, also have tried asterisk faxdetection, nvfaxdetect, disable echo cancellation by hand always with same results. Grandstream ATA is using

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

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 all, I'm playing with Asterisk and I've already configured all needed .conf files. It works quite well, but now I need your help to tune the system: when I place a call from a softphone to the PSTN, I can't hear directly Telco's tones and I can't use its services, e.g. a mobile's answering machine. I don't know if I have to modify the dialplan or if it depends on my

Using 1.6.2.1 with a TDM400, attached to internal analog phones and PSTN. When I dial out to PSTN, I cannot send tones, like push "1" for something stupid. The call itself works, but the DTMF tones fail. -- Starting simple switch on 'DAHDI/1-1' -- Executing [6258013 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [6258013 at

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

Hello, I have asterisk@home with zaphfc patch applied (http://dondisperato.blogspot.com/), but I can not make call to legacy PBX (Alcatel 4400). I can only accept incoming calls. I am dialing with this: exten => 202,1,Dial(Zap/g1/242) --- asterisk1*CLI> bri debug span 1 Enabled debugging on span 1 -- Executing Dial("SIP/201-4678", "Zap/g1/242") in new stack