similar to: Multiple SIP endpoint registrations

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, 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

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, 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, 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, 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, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm

Hi, Over the weekend tried to setup a test using the new app_calendar code but receiving the following error: [Jan 17 09:23:35] WARNING[27663]: res_calendar_icalendar.c:146 fetch_icalendar: Unable to retrieve iCalendar 'testcal' from 'https://office.test.net/home/teamshare at test.net/Calendar/': Server certificate verification failed: issuer is not trusted The target server is

Hi, Running Asterisk 1.6.2.11 and wondering what would be the best way to send an email when a missed call has occurred ? I believe you can modify [stdexten] is this still the case in V1.6 ? -- Thanks, Phil

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

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know