similar to: Attempted break in ?

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

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, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my extensions.conf I have a simple plan for testing :- [inbound-sip] exten => uxbod,1,Dial(sip/1001,20,t) exten => uxbod,n,PlayBack(uxbod) exten => uxbod,n,VoiceMail(1001@voicemail,s) exten => uxbod,n,Hangup() exten

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, 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, 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, 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, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. -- Thanks, Phil

Hello all, After numerous issues with Snom phones (360/370/870) potentially looking to migrate too Yealink as their product range looks very promising indeed. Are any of you using them with Asterisk ? How do they perform ? Do you use mass deployment at all ? Would be very interested to hear from you. -- Thanks, Phil -------------- next part -------------- An HTML attachment was

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 sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun

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)

I see this has now been released :) What is the best way to upgrade both the Dom-0 and Dom-U to Xen 3.2 within CentOS 5.2 ? Regards, -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone:

Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be loaded. Yet I do have iksemel installed: ls -l /usr/local/lib/libik* -rw-r--r-- 1

Anybody tried one with Asterisk yet ? Views ? Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration

Experiencing a problem when users attempt to forward a voicemail from within VoiceMailMain(Option 8) I see the console message: Couldn't not find mailbox XXX in context default As why are running in a multi-tenant environment voicemail.conf has been separated into individual contexts. The users retrieve their email by dialing an extension which calls VoiceMailMail(XXX at VMContext) so how

Hi, I have been using Zapateller with a TDM400 no problems at all, but recently I have ported our BT number to a VoIP provider, and have a strange problem. When I phone our number I first get the BT unavailable three tone sound, and then it actually connects the call via IAX2. So, I disabled zapateller in the dialplan and tried again. Would you believe it worked fine. Has anybody else come

Looking for some help from the UK please. I backed up all my Asterisk configuration before re-installing the server from 32 -> 64 bit. Unfortunately I did not transfer the backup to another machine!!!!! I now have a TDM400P that is not picking up the line. Can you see what I have done wrong when I have rebuilt the config please: dahdi_scan ---------- [1] active=yes alarms=OK