similar to: strange warning

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 BaBa Jigger -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, There is a parameter called "nonce" included in every register request that a UA sends to asterisk. I have read sip debug a lot and only found out that the "nonce" parameter value which is used in register request was generated by asterisk server in a previous sip response. As you can see in the sip debug (labled in red). <--- Transmitting (NAT) to

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not

Hi, how are mails forwarded, if I do have the same alias pointing to two different users like this (two entries, two lines): bon.aqua: coke bon.aqua: pepsi Will coke and pepsi get the mail adressed to bon.aqua or will only the first entry get the mail? I know, that "bon.aqua: coke, pepsi" will forward the mails to coke and pepsi, Cheers, G?tz -- G?tz Reinicke IT-Koordinator

Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy 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

i want to tag categories to its menuname. i have a csv containing menu item name and in other csv i have a column containing some strings, i want to pick that strings from categories and look into menu items if any menu item containing that string i want to create a new column next to menu item name flagged as 1 otherwise 0 and the only condition is once a menu item flagged as 1 i don't need

Hi all, I need a fully developed web based remote 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

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:

Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically 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

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont 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

Hi all, i 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

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail> and here <http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>. I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for

Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with "Asterisk <Unknown>" caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server

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:

Hello, I am trying to migrate one of our fedora-based servers to CentOS. Our PHP developers doesn't allow us to switch because there are no crucial (for them) php packages: php-smarty php-adodb php-accelerator It is not acceptable for us to download sources and compile them and repeat the process each time there is a security bug. Can you recommend any repository which will deliver all the

Hi guys This is my Ist post on this group. Is there any variable like ($VM_CALLERID for voicemail mailbox) for accessing Asterisk Voicemail password which is set through comedian mail.?????????????? plz reply me as soon as possible.... <html><div><PRE class=quote><IMG height=2 src="http://graphics.hotmail.com/greypixel.gif" width="100%"

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the