similar to: Actiontec's InternetPhoneWizard and Asterisk

Hi all, Anyone succeed using * with Asterisk with full functionality (dialing, ringing, etc.). Thanks, Dan

Hi, Anyone succeed using PCPhone application from iConnect Here with Asterisk? It is basically a software SIP phone. I ask this because that application is the only one fully supporting Actiontec's Internet Phone Wizard USB FXS device. Thanks, Dan

Hi, There is someone on this list who has the specification for the IPW driver? I want to provide full support for it in DIAX Phone, but Actiontec does not answer to my mails. There is any windows driver available for the Digium's S100U USB interface? Anyone knows if it can provide full functionality for the connected phone, like callerid, callwaiting callerid, message indication, ring,

Hi all, I want to get the following functionality: define one extension as a virtual fax machine. Every fax redirected to that extension to be converted in a picture file (bmp/jpg/gif or something else) and then attached to an email and send to an e-mail address. Are you aware of a linux based application who does something like this and can be installed on the same computer as Asterisk? Another

Hi, I wish to connect several ATA186 Phones to each other, to iconnecthere and to the PSTN using asterisk. Please tell the appropriate settings for firewall (ports to open etc.) sip.conf and extensions.conf(part relevant to iconnect). Also I would be glad to get a working example of your ATA186 configuration. I tried searching the mailing lists and several sites but did not find an answer.

Hi all, There is something special I must configure in order to get the voice mssage by mail? In voicemail.conf I have: serveremail=asterisk@mydomain.ro attach=yes [default] 301 => 6535,Home Mailbox,dtoma@fx.ro I have tried to let a message for 301, but this message is not forwarded by mail. I am missing something? Thanks, Dan

Hi, I have two Asterisks identically installed on two computers. One of them is directly connected to the Internet, the other one through a NAT router (Netgear MR314). On the one behind the router I have an X100P card installed for PSTN connections. In the local LAN of each PBX they are several hardware IP phones (Cisco 7960 and 7940 with SIP images, firmware image P0S3-04-4-00.bin). I have

I resend this message, as it was not posted on the list first time I send it.... Dan ----- Original Message ----- From: "Dan" <dtoma@fx.ro> To: <asterisk-users@lists.digium.com> Sent: Friday, June 27, 2003 10:13 AM Subject: Re: [Asterisk-Users] Basic Asterisk questions - personal coments > > Why is it that most users who don't understand threaded email is on

Hi, I have succeed using Asterisk on VMWare on an Athlon@1GB with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem

Hi, There is any possibility to dial a specific extension and then enter in your own mailbox (the one defined for that specific SIP phone) without asking for the exxtension number but only for the password? I want to be the same extension for all phones, not a specific one for each of them. It is possible to have a time stamp in the recorded message? I want to know when the message has been

I have an asterisk Redhat 9 box running 4 hardphone extensions. Inter-extension calls are crystal clear. However when I dial out through my iconnect account I get a lot of jitter. At first I thought it was my nat gateway but after I programmed on of the hardphones (budge tone 100) for direct dial to iconnect I have clear voice transmission. I have no way of explaining this. My asterisk sip.conf

Hi all, A new version (0.9.6) of DIAX is available for download at: http://www.laser.com/dante or http://www.geocities.com/tdanro There are no new functions, but some bugs fixed: What's new in version 0.9.6: - add Default_user locales as new language. The program language can be automatically selected based on default user locales on your system. You still can manually select the language,

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hi all, DIAX 0.9.4 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. What's new in 0.9.4: - IAX2 support (new DLL); - selectable DSP: Echo cancellation, AGC, Denoise; - plaintext and md5 authentication supported; - the phonebook is now in a separate

*This message was transferred with a trial version of CommuniGate(tm) Pro* Will asterisk actually convert between two different codecs????? ie, a SIP endpoint running GSM and another running G.711? Wouldn't that add quite some latency? I was always under the impression Asterisk did not recompress and was smart enough to negotiate the right codec at each end and just pass through the RTP

Hi all, Which is the practical (from your experience) limit of the number of X100P cards installed in a single Asterisk box? Asterisk can work reliable with 6 X100P cards in the same box? Anyone know when the 4 ports FXO Digium card will be available on the market? Many thanks, Dan P.S. Please do not aswer with RTFG ...tried before without success...:-))

Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed file, but still get that error and the macro cannot continue. Why? I have tried even with a simple

Hi, I was able to connect asterisk to iconnect's service. It took me almost two hours, but it's because I was having NAT trouble. I finally discovered that you can set the iconnect host to natrealy.deltathree.com to make it work. (for those of you who, like me, don't have the time to search the archive I'll provide a working sample in a minute) My problem was sound

hi i got iconnect to work, works pretty well now except calls sometimes (more often than not) hang up after a couple of minutes.. heres a bit of the debuging Record-Route: <sip:61892142222@213.137.73.178:5060;maddr=213.137.73.176> From: sip:61892142222@natrelay.deltathree.com;tag=3281050172-73809 To: "JUSTIN XLITE" <sip:2001@61.95.68.84>;tag=as09766a78 Call-ID:

Hi, Anyone knows a software fax device which can act as a SIP UA? I want to have a SIP based FAX machine (sofware) on a PC associated with an Asterisk extension. Thanks, Dan