Re: [asterisk-users] Re: Loosing IAX connection between offices
Louis-David Mitterrand wrote: On Thu, Nov 30, 2006 at 08:52:50AM -0600, DM wrote: Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's unreliable and perfectly good hosts will become UNREACHABLE for no apparent reason, while SIP connections keep going through. For trunking, avoid IAX and use SIP. For the record I have had 4 asterisk (1.2.6) boxes connected with IAX trunks (every box registers with every other box) for over 6 months now with no issues. I did have some issues with a Linksys firewall, but that was with SIP registrations, not IAX. A firmware update fixed that. I'm no * dev, but what exactly is so crap about IAX? -Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk billing software
My company has recently purchased AgileBill/AgileVoice and have had numerous problems getting it up and running. We submitted support tickets within our 30 day trial period, of which only a few were resolved. As of right now, their software is crashing when it attempts to provision new accounts, and nobody has gotten back to us about this... FOR THREE WEEKS! Is there anyone else out there that has dealt with them before? If so, I'd like to hear your opinion. Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk billing software
Andrew Joakimsen wrote: Chris: We were evaluating AgileVoice currently, could you please elaborte on your problems? Did they not do the instattion for you? They did the installation. I'm going to be very careful with my wording here, but if you are currently evaluating their software, I recommend you ask to see *all* of the documentation for AgileVoice (not just AgileBill). -Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 900 rules
There are several Caribbean countries within the 8XX range, as well as more toll free, and regular area codes. Here's the full list at NANPA: http://www.nationalnanpa.com/nas/public/npasInServiceByNumberReport.do?method=displayNpasInServiceByNumberReport Doug Crompton wrote: Ok so ONLY 900 numbers are pay. Next question 18XX numbers. are they all toll free? Is there any space in 8xx that is used otherwise? Doug On Tue, 14 Nov 2006, Eric ManxPower Wieling wrote: Doug Crompton wrote: I had a 19xx rule in asterisk and realized when I was trying to dial an area code 978 in MA that that was not a good idea. Is there a more defined rule for 900 space of non pay vs. pay codes? _1900NXX _NXX976 ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chris Mazuc Systems Administrator DataGroup Technologies (252)329-1382 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 flickering screen
My company has had a few screens go out on us, but all of those were completely blank. I'm not sure if we just got a bad batch or what, but the Snom phones are usually a solid piece of hardware. I'd try to RMA it. Nick Hoffman wrote: Hi guys. I just bought and configured a Snom 360 and have noticed that the LCD is constantly flickering at a rate of ~10-15Hz (that's a guess). Either way, it's very distracting. Has anyone else encountered this before? Any solutions? Cheers, -- Nick E: [EMAIL PROTECTED] P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with setting outbound caller id when calling another asterisk
Asterisk seems to have a bug which is not letting me set the caller id to another peer's caller id. http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg23230.html I've sent this to the asterisk-users mailing list, hopefully I get a response soon if there is a workaround. I'm going to see if there is a way to blindly accept calls from a known IP address, but I don't think there is a way that would retain CDR information. Chris Mazuc wrote: I have an asterisk box at a remote location (which I will call remote), which registers to my local asterisk box (I'll call that one local), and uses that to route calls to the outside world. The problem I am having is that the remote location needs to lie about it's callerid sometimes, however if I set a callerid that matches the extension of another peer that exists, local returns a 403 to remote. I can set the callerid to the did and it will work fine, or I can set the callerid to something random and it will work fine. What does * do with the proxy-authorization header, because it seems to be ignoring the username part. Any help is greatly appreciated. Thanks, Chris Mazuc -- SIP read from REMOTE:1025: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;rport From: My Name sip:[EMAIL PROTECTED];tag=as4f42dab4 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username=1XX1205, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=45a347bc, response=934b409f19a0ebf28d1cf266db29f497, opaque= Date: Tue, 24 Oct 2006 20:26:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 2238 2239 IN IP4 REMOTE s=session c=IN IP4 REMOTE t=0 0 m=audio 15384 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (14 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to REMOTE : 5060 (NAT) Found user '1XX1200' Reliably Transmitting (NAT) to REMOTE:1025: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;received=REMOTE;rport=1025 From: My Name sip:[EMAIL PROTECTED];tag=as4f42dab4 To: sip:[EMAIL PROTECTED];tag=as1f40e0ec Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with setting outbound caller id when calling another asterisk
I have an asterisk box at a remote location (which I will call remote), which registers to my local asterisk box (I'll call that one local), and uses that to route calls to the outside world. The problem I am having is that the remote location needs to lie about it's callerid sometimes, however if I set a callerid that matches the extension of another peer that exists, local returns a 403 to remote. I can set the callerid to the did and it will work fine, or I can set the callerid to something random and it will work fine. What does * do with the proxy-authorization header, because it seems to be ignoring the username part... or maybe I need to go read some RFCs. Any help is greatly appreciated. Thanks, Chris Mazuc -- SIP read from REMOTE:1025: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;rport From: My Name sip:[EMAIL PROTECTED];tag=as4f42dab4 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username=1XX1205, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=45a347bc, response=934b409f19a0ebf28d1cf266db29f497, opaque= Date: Tue, 24 Oct 2006 20:26:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 2238 2239 IN IP4 REMOTE s=session c=IN IP4 REMOTE t=0 0 m=audio 15384 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (14 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to REMOTE : 5060 (NAT) Found user '1XX1200' Reliably Transmitting (NAT) to REMOTE:1025: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;received=REMOTE;rport=1025 From: My Name sip:[EMAIL PROTECTED];tag=as4f42dab4 To: sip:[EMAIL PROTECTED];tag=as1f40e0ec Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
