similar to: Difference between WaitExten and TIMEOUT (response)

Hi List; In the outbound, I read in the documents the Wildcard match "by using the . (period)", but I did not understand how Wildcard will work (like what)? As I know that Wildcard is a term used with the Diguim TDM card (FXO and FXS), so what is the relation between such cards and the matching in the dial plan? Any help? Regards Bilal

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone

Hi List; My Asterisk version is 1.4 and I am trying to use the ResponseTimeOut() application to control the response time of the Background function, but when the running arrive for the ResponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension (Test_Bilal,s,3) Spawn extension (Test_Bilal,s,3) exited non-zero on 'Zap/1-1' Hangup

Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why?

Hi All; How to increase the waiting time between entering the digits for the analoge phone device that is connected to fxs? Is it by DigitTimeout? But how it will be apply for analoge station if the user just pickup the handset and dialed the number? Any help? Regards Bilal

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi Danny; Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager? Regards Bilal ------------------------- It depends on how you are configured. The gui interfaces using Asterisk Manager, so you get the Same IO from the gui that you would get from a native manager session. -----Original Message----- From: asterisk-users-bounces at

Hi All; Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Any advise? Regards Bilal

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.

Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards

Hi List; I read the following sentence: "The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable" In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad

Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? Regards Bilal

Hi All; I tried Firefly softphone with IAX and it gave very poor quality. Any one advise a good strong softphone that can work with IAX fine? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

Hi All; Any one can suggest a TFTP server to be installed in Fedora (same machine that Asterisk is installed) to be used for Cisco IP Phones to download the required firmware and configuration files. Thanks for the help in advance. Regards Bilal

Hi All; My ISDN was working fine, and suddenly I start getting the below: sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway! There is a Yellow Alarm, so what it could be the problem? My configuration as following: system.conf span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 chan_dahi.conf context=IncomingPSTN group=0 signalling=pri_cpe switchtype=euroisdn

Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal

Hi All; Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? Regards Bilal

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,