[asterisk-users] odbc connection timeout varable
Hi all Does anyone know of a variable that i could check to see if the reason func odbc didnt return results was because of a timeout error so i could play a audio file about that thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
Why are you concerned? ODBC reconnects automatically if necessary. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
On Tuesday, November 11, 2014, jg webaccounts...@jgoettgens.de wrote: Why are you concerned? ODBC reconnects automatically if necessary. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I would like to tell callers the system is down as apposed to saying no info in system when the system is up or saying the info in the system -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
It takes a small fraction of second to reconnect. You should not experience any missing info. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
Unless of course the database server is not running at all for some reason. Regards; JVC -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg Sent: Tuesday, November 11, 2014 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] odbc connection timeout varable It takes a small fraction of second to reconnect. You should not experience any missing info. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
On 11 November 2014 15:27, Tech Support aster...@voipbusiness.us wrote: Unless of course the database server is not running at all for some reason. Regards; JVC Surely that should be monitored by some system designed for that purpose such as Nagios? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
Unless of course the database server is not running at all for some reason. But that's not exactly an Asterisk problem. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
right but that is the problem and i was wondering if there is way for asterisk to set a variable when that happens just like curl On Tue, Nov 11, 2014 at 5:37 PM, jg webaccounts...@jgoettgens.de wrote: Unless of course the database server is not running at all for some reason. But that's not exactly an Asterisk problem. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
It should not happen. I have a couple of Asterisk servers using the ODBC connection. I never ever had any problem with ODBC or the database. What database are you using? jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
well it should but this morning my database hosted at a remote location was down due to conditions at the remote site the question isnt if it should happen or not the questions is there a way for me to know that the odbc query retruned empty because of a connection timeout? in curl i could get that info is there a way also for odbc? On Tue, Nov 11, 2014 at 9:33 PM, jg webaccounts...@jgoettgens.de wrote: It should not happen. I have a couple of Asterisk servers using the ODBC connection. I never ever had any problem with ODBC or the database. What database are you using? jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc connection timeout varable
So, your DB is not on the same machine? WAN or LAN? jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ITSP Gateway Solution?
Right now we I am using Asterisk boxes as a gateway between our Level 3 SIP trunks and our customer PBXs. I love and understand Asterisk but the company I am working for is looking for a more Commercial type solution where we can go to a vendor for support etc. I know, we can get Asterisk support etc.. It's not my decision and I sort of get why they are leaning away from Asterisk, I just don't agree. I need to at least explore other options for more appliance products that will do the job the Asterisk boxes are doing now but, with a simple interface to add/remove trunks, DIDs etc. Integrated security and billing options/add-ons would be great. I know Digium offers appliance solutions but they don't seem to be anywhere near the power of what we are currently using. One big advantage I could see is going diskless but, I am really not sure whats out there, I am just kicking tires at the moment. The best of all worlds would be something with commercial support, a good GUI, billing and security built in but all based on the Asterisk core which I can understand :-) Again, just kicking tires as I can't just scream Asterisk and not be willing to look around to see what's out there. Everything I see out there seems to want to Transcode and such.. All we need is something to do SIP to SIP, no TDM here at all. Some codec support beyond G711 of course but that's it. I know there is every reason to do all this with Asterisk and that is my preference but in this case, I have lots of folks that lean more towards commercial products and I have not been able to completely sell them on the joy and flexibility of Asterisk. I don't want a Virtual PBX GUI solution, I want something that is built to be a work-horse, as a gateway only. No extensions, voicemail, ring groups or any of that. Just calls in/out to/from trunks, security and billing. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls to a GSM gateway SIP/2.0 401 Unauthorized response when dial 7777 to Asterisk
Hello: I'm newbie in asterisk, please help me. My context is as follows: 192.168.4.2 -- Asterisk 11.13.1 complied from source 192.168.4.4 -- Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension (configured as a hotline on TG100) to asterisk server, but asterisk server sends me SIP/2.0 401 Unauthorized response, I think it's a matter of contexts but I don't find the problem. Attached are sip.conf, extensions.conf and debug from 192.168.4.4 (TG100 GSM gateway). Thanks in advance. --- SIP read from UDP:192.168.4.4:5060 --- INVITE sip:@192.168.4.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport Max-Forwards: 70 From: 9 sip:9@192.168.4.4;tag=as67354416 To: sip:@192.168.4.2:5060 Contact: sip:9@192.168.4.4 Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 CSeq: 102 INVITE User-Agent: TG100 Date: Wed, 12 Nov 2014 10:13:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 281 v=0 o=root 1426707418 1426707418 IN IP4 192.168.4.4 s=Asterisk PBX 1.6.2.6 c=IN IP4 192.168.4.4 t=0 0 m=audio 10048 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv - --- (14 headers 13 lines) --- Sending to 192.168.4.4:5060 (no NAT) Sending to 192.168.4.4:5060 (no NAT) Using INVITE request as basis request - 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 Found peer '5' for '9' from 192.168.4.4:5060 --- Reliably Transmitting (no NAT) to 192.168.4.4:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;received=192.168.4.4;rport=5060 From: 9 sip:9@192.168.4.4;tag=as67354416 To: sip:@192.168.4.2:5060;tag=as16de6e5c Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 CSeq: 102 INVITE Server: Asterisk PBX 11.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=72011a6b Content-Length: 0 Scheduling destruction of SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' in 6400 ms (Method: INVITE) --- SIP read from UDP:192.168.4.4:5060 --- ACK sip:@192.168.4.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport Max-Forwards: 70 From: 9 sip:9@192.168.4.4;tag=as67354416 To: sip:@192.168.4.2:5060;tag=as16de6e5c Contact: sip:9@192.168.4.4 Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 CSeq: 102 ACK User-Agent: TG100 Content-Length: 0 - --- (10 headers 0 lines) --- Really destroying SIP dialog '6e9eab843b74d0860b108ed13a5d22c9@192.168.4.4' Method: OPTIONS Really destroying SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' Method: ACK uc*CLI extensions.conf Description: Binary data sip.conf Description: Binary data -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
