similar to: Question on Priorities

Hi, I have recently upgraded from CentOS 5.1 too 5.2 and now run Xen 3.0.3. What would be the best way to upgrade too Xen 3.2.1 ? I presume I would also need to change my network settings for xenbr0 aswell ? Any help would be greatfully appreciated. Regards, -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint: F57A 0CBD DD19 79E9

I have this running on my Asterisk server, and have opened up ports UDP/5060 and UDP/10000-20000 but for some reason when I try and connect too my SIP extension it does not work. Are these the correct ports ? -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver:

Hi, Now that I have Asterisk up and running I would like to find a good SIP gateway in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I am looking for peoples recommendations. Apologies if this is the incorrect forum for this type of request. Regards, -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint:

Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110309/fe9d7bc7/attachment.htm>

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

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, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with

Hi guys, i just came to know that CDR(dst) field is set to current extension instead of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan)

Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial

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, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

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 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, There is a problem with my dialplan. here is the dialplan: exten=> 123,1,Dial(SIP/U1,,Ttg) exten=> 123,2,Hangup exten=> h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling

Hi, I have three Snom M3s at the moment but getting pretty fed up with the issues :( I am UK based and would be interested to hear of other peoples recommendations. Key features :- * VM Notification * Good Range * G729 codec support * Common/Private Address Books per Handset(s) TIA, Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be

Hi, Odd problem have just noticed in that when I call into the PBX DAHDI detects the call and hands it off to the extension, if I then hang up it still continues to process through the dialplan. This is what I have in chan_dahdi.conf: [channels] language=en echocancel=yes usecallerid=yes cidsignalling=v23 sendcalleridafter = 2 hanguponpolarityswitch=yes rxgain=2.0 txgain=3.0 progzone=uk

Hi, This is a real strange one and trying to phantom it out. One of our clients is connected to our Asterisk installation, from two sites, via VPN which works great. Every so often one of the sites VPN tunnel goes does for say a couple of seconds. When that happens all the extensions, including both sites, ring which is bizarre. Has anybody seen this before ? I only see two places in the dial

Hi all, We are planning to shift our sip users from one platform to another. (basically from one asterisk server to another). the problem we are facing is that both asterisk servers are using different ports to listen for sip. and both have live customers on them. provisioning their ata's is not a good option for us coz of our settup. we cant just ask the customers to change their ports for

Hi, Built a new server at the weekend and install Asterisk 1.6.0.9 and IAX and SIP work great :) The one problem I am having is getting the OpenVox (TDM400 type card) to work. It is successfully identified using WCTDM kernel module and dahdi_scan picks it up just fine. The issue is when I try and setup dahdi_channel.conf as it fails everytime. When running asterisk -rvvvv I see the port pick

Hi im trying access the www.voip-info.org website since yesterday but i cant open it. google search diaplay correct search results but it doesnt open when i click the link. it displays a message about tcp error which says -->"There was a problem communicating with the server". I dont know what the problem is. I just want to ask whether their server is down or not and is everybody