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AXvoice server hacked. Here are few working accounts USE XLITE to make calls.... Registrar/Proxy magnum.axvoice.com:9060 Free Sample account.... username=xMaxwellSmartx secret=thanksapache username=woodsy type=friend secret=haramikuttasala username=wumingzi type=friend secret=kickyourass Enjoy! B.R...

...one without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce received from '<sip:diet at magnum.axvoice.com>' I dont know what is the problem. Can somebody explain me this? Below is my client configuration. [general] bindport=9060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm context=incoming allowexternalinvites=yes register=> diet:pepsi at magnum.axvoice.com...

...omatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070817/9706efcf/attachment.htm

...d by asterisk server in a previous sip response. As you can see in the sip debug (labled in red). <--- Transmitting (NAT) to 208.120.167.146:80 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 208.120.167.146:80;branch=z9hG4bK03891485;received= 208.120.167.146 From: <sip:bernart48 at magnum.axvoice.com>;tag=as65460c44 To: <sip:bernart48 at magnum.axvoice.com>;tag=as3a5cc850 Call-ID: 48f3a8b426c375a161dc1f4479bba956 at 127.0.0.1 CSeq: 19680 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest...

...te provisioning system. I cant find anything reliable on the internet. Have already checked ataconfig.com and voxilla-ays.com. have tried to contact them but got no response. So if anybody knows a good provisioning system then plz tell me about it. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071024/0a228d78/attachment.htm

...er also, so if your idea needs changes in the code please dont hesitate to share, otherwise you WILL get a call from me with a special background noise crafted just for you :) Meanwhile i'll try my best to come up with a solution. Cheers -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: rizwanhasham at gmail.com W: www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110506/460a82e8/attachment.htm>

...usy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R and r options in Dial application but they dont work. -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/ac78db36/attachment.htm

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080125/4f833de8/attachment.htm

...nt think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from different IPs for the same username. Can anybody help? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070911/62694010/attachment.htm

...ike configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070801/bf3c906b/attachment.htm

...need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071029/9d528ea8/attachment.htm

Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XXXXXX the call will be forwarding automatically to anther number 0520xxxxxx Does anybody have a solution to this problem. Thanks and Regards. -------------- next part -------------- An HTML attachment was

...l/asterisk/voicemail/default/1757XXXXXXX/INBOX'] I know that the error is caused due to stale odbc connection with mysql. But i want to find out if there is a cure for it. Why the connection went stale in the first place also. Any ideas? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: rizwanhasham at gmail.com W: www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110428/b584d5b3/attachment.htm>

...s why its call limit is not reset to zero. I dont have sip debug for this problem yet, i'll post it later when i have it. meanwhile if somebody has experienced a similar problem and has successfully fixed it, then plz share my burden and help me. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070823/2d2c574c/attachment.htm

.... My guess is that someone has been sniffing my server's sip traffic. In that case what should i do to get rid of the sniffers? If you think there is another reason for that then please tell me even if you dont have the solution. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: rizwanhasham at gmail.com W: www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110228/5075d095/attachment.htm>

Hi, I was trying out call file just to see how they worked and my system does not seem to do anything with them, although asterisk *is* deleting the files that I put into /var/spool/asterisk/outgoing. 1. I nano'd a quick call file like so: Channel: SIP/axVoice/9105555555 CallerID : Leebo <5555555555> MaxRetries: 2 RetryTime: 30 WaitTime: 10 Context: main_menu Extension: s Priority: 1 2. And then mv'd to /var/spool/asterisk/outgoing As I mentioned, Asterisk appears to be grabbing the file, but there is no call made. Q. Do calls originated...

I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

...><IMG height=2 src="http://graphics.hotmail.com/greypixel.gif" width="100%" vspace=9></PRE><PRE class=quote><FONT face="Courier New, Courier, Monospace" color=#000033 size=2>Syed Jamshed Zaidi <BR>Asterisk Admin/Developer <BR>@ Axvoice +92-0321-4087492 <BR>(JAMY-VIRUS)</FONT></PRE><PRE class=quote><PRE class=quote><FONT face="Courier New, Courier, Monospace" color=#000033 size=2>"Shoot for the moon. Even if you miss, you'll land among the stars"</FONT></PRE&...

Hello there! We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -,

...this: [incoming] exten=>5555555555,Goto(incoming,s,1) Thus transferring the call to the context that I want it to come in on. The problem that I have is the caller ID ${CALLERID(num)} always shows the actual number provided by Telasip and not the actual caller id information. I also have axVoice and they do not do it this way. They simply send it to the context without specifying an extension. Below is a sip packet. The Caller ID comes through correctly on the sip packet by for some reason as I mentioned, Asterisk is reporting it as the number I have with the sip provider. Below is...