[asterisk-users] Unix connections not always disconnecting
Hi We are using asterisk 1.8.23.1 We have a script that runs on a minute cron which polls the asterisk server for 3 bits of information by using asterisk -rx 'command' which then gets pushed to a graphite server we have 99% of this runs smoothly. Every now and again, the asterisk service will become completely unresponsive and if we look at the logs we will see the following: [2013-11-07 00:17:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:17:01] VERBOSE[26566] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:17:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:17:01] VERBOSE[26586] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:17:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:18:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:18:01] VERBOSE[26650] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:18:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:18:01] VERBOSE[26670] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:18:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:19:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:19:01] VERBOSE[26736] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:19:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:19:01] VERBOSE[26756] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:19:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:20:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:20:01] VERBOSE[26820] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:20:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:20:01] VERBOSE[26841] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:20:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:21:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:21:01] VERBOSE[26914] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:21:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:21:01] VERBOSE[26934] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:21:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:22:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:22:01] VERBOSE[27000] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:22:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:22:01] VERBOSE[27020] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:22:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:23:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:23:01] VERBOSE[27084] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:23:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:23:01] VERBOSE[27104] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:23:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:24:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:24:01] VERBOSE[27170] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:24:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:24:01] VERBOSE[27190] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:24:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:25:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:25:01] VERBOSE[27255] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:25:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:25:01] VERBOSE[27276] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:25:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:26:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:26:01] VERBOSE[27624] asterisk.c: -- Remote UNIX connection disconnected [2013-11-07 00:26:01] VERBOSE[25963] asterisk.c: -- Remote UNIX connection [2013-11-07 00:26:01] VERBOSE[27644] asterisk.c: -- Remote UNIX connection disconnected As you can see, at these times there isn't a disconnect for every connect. I think this ends up maxing out the amount of connections the service allows and that's what makes it unresponsive. A service restart fixes the issue at this time. This happens completely randomly, I've not been able to correlate this happening with any other events that are going on at the time. Can anyone think of any reason why doing the asterisk -rx command might not disconnect cleanly? Thanks
Re: [asterisk-users] Unix connections not always disconnecting
On 07/11/13 11:20, Ishfaq Malik wrote: Hi We are using asterisk 1.8.23.1 We have a script that runs on a minute cron which polls the asterisk server for 3 bits of information by using asterisk -rx 'command' which then gets pushed to a graphite server we have 99% of this runs smoothly. Out of interest what are you trying to monitor? We tend to use cacti for graphing and snmp provides all the information we require. -- _ -- 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] Unix connections not always disconnecting
On 7 November 2013 15:26, Gareth Blades mailinglist+aster...@dns99.co.ukwrote: On 07/11/13 11:20, Ishfaq Malik wrote: Hi We are using asterisk 1.8.23.1 We have a script that runs on a minute cron which polls the asterisk server for 3 bits of information by using asterisk -rx 'command' which then gets pushed to a graphite server we have 99% of this runs smoothly. Out of interest what are you trying to monitor? We tend to use cacti for graphing and snmp provides all the information we require. Active calls, sip peers connected, sip peers disconnected and then breaking all of those down by customer as we run a multi tenanted set up. SNMP would give us totals but I don't think it would do the breakdown by customer. -- 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] Unix connections not always disconnecting
On 13-11-07 10:31 AM, Ishfaq Malik wrote: On 7 November 2013 15:26, Gareth Blades mailinglist+aster...@dns99.co.ukwrote: On 07/11/13 11:20, Ishfaq Malik wrote: Hi We are using asterisk 1.8.23.1 We have a script that runs on a minute cron which polls the asterisk server for 3 bits of information by using asterisk -rx 'command' which then gets pushed to a graphite server we have 99% of this runs smoothly. Out of interest what are you trying to monitor? We tend to use cacti for graphing and snmp provides all the information we require. Active calls, sip peers connected, sip peers disconnected and then breaking all of those down by customer as we run a multi tenanted set up. SNMP would give us totals but I don't think it would do the breakdown by customer. You should avoid using the CLI to access that information. You'd likely getter better results using AMI or CEL. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?
2013/11/5 Larry Moore lmo...@omninet.net.au On 5/11/2013 6:09 PM, Olivier wrote: Hello, I've got an analog phone which is currently receiving unsollicited FAX calls from PSTN. For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would let voice calls come in and out and translate incoming FAX calls to TIF files (forwarded through email)). My target setup is : PSTN -- analog-- SPA3102 Line Port -- SIP -- Asterisk -- SIP -- SPA3102 Phone Port -- analog -- Analog phone When a call comes in, analog phone rings. If callee answers and a fax tone is detected, then the incoming call is sent by Asterisk to ReceiveFAX application which translates incoming audio to TIF file. My setup is working ok when I'm using ReceiveFAX in fallback mode (with f option). Then I would like to improve my setup letting ReceiveFAX negociate T.38 with SPA3102. The trouble is SPA3102, as I configured it, seems to refuse T.38 negociation as I'm reading lines like this in Asterisk logs: == Using UDPTL CoS mark 5 [2013-11-05 10:36:50] WARNING[3061][C-0007]: res_fax.c:1698 receivefax_t38_init: channel 'SIP/myline-000e' refused to negotiate T.38 My question is: Any hint on how to configure SPA3102 PSTN Line port so that it would accept to upgrade to T.38 ? The SPA3102 only supports T.38 on the FXS port, the FXO port uses G711 for a fax session. That explains why I couldn't find the option ;-) The Grandstream HT503 supports T.38 on both the FXO and FXS ports. I never tried this one. How would you rate this product ? Is it easy to auto-provision an HT503 over TFTP or HTTP ? Is it easy to localize FXO/FXS setttings for non-US countries (those having played with a SPA3102 sure know what I'm thinking about) ? Do it T.38 implementation works ok with Asterisk ? What problem do you have receiving a fax over G711? I tried few calls in G711 fax mode and it worked OK but I rated T.38 as more reliable solution in the long run and in general. Larry. -- _ -- 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
[asterisk-users] Capture dead phone?
Asterisk 11.1 Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Background: we run a small call center. Our agents sit in two groups, with their IP phones running from 2 different switches. Every once in a while the power on one side of the room will go out, or one of the switches will die, or one of the agents will knock something loose with their foot. If/when that happens while the agent is on a call with a customer, I'd like to be able to save that caller and put them back in the queue (at the head of the queue). -- Mitch -- _ -- 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] Capture dead phone?
I don't have an answer for you, but I can suggest some areas to investigate. stuck calls could be detected using SIP session timers. I've had bad luck using session times in Asterisk, but the problem might somehow be unique to our setup. Dead peers can be detected using qualify Asterisk supports some RTP keepalive features. Asterisk has the ability to continue in the dialplan after one leg of the call hangs up. See hangup handlers (new in 11.x or 10.x) and the options to Dial See sip.conf.sample included in Asterisk, core show application dial and the UPGRADE*.txt files in Asterisk. The most effective thing to do is avoid phones going away unexpectedly. Get reliable switches, get your switches on battery backed up power, move important stuff away from people's feet. The BOFH in me thinks a shock sensor attached things which might get kicked, connected to a locomotive horn nearby is a delightful idea, but is neither nice nor practical. I know from personal experience a locomotive horn can cause a sleeping person to levitate several feet in the air. I imagine it would have a similar effect on a call center agent. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Thursday, November 07, 2013 7:51 PM To: Asterisk Users Mailing List Subject: [asterisk-users] Capture dead phone? Asterisk 11.1 Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Background: we run a small call center. Our agents sit in two groups, with their IP phones running from 2 different switches. Every once in a while the power on one side of the room will go out, or one of the switches will die, or one of the agents will knock something loose with their foot. If/when that happens while the agent is on a call with a customer, I'd like to be able to save that caller and put them back in the queue (at the head of the queue). -- Mitch -- _ -- 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] Capture dead phone?
On 13-11-07 07:51 PM, Mitch Claborn wrote: Asterisk 11.1 Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Background: we run a small call center. Our agents sit in two groups, with their IP phones running from 2 different switches. Every once in a while the power on one side of the room will go out, or one of the switches will die, or one of the agents will knock something loose with their foot. If/when that happens while the agent is on a call with a customer, I'd like to be able to save that caller and put them back in the queue (at the head of the queue). No, you won't be able to save the call if the far end goes down. Best you could do would be to enable qualify, track then the agent phone goes offline, if a call also drop around that time frame, initial some sort of callback. However, solve the issue at the source. Spend the money for a UPS at each desktop, convert your phones to PoE and install a UPS in your server room. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] two steps when calling from web!
Dear All. When I calling a number from web, my softphone show me Answer and Decline bottoms, and then I have to click Answer to call the number. it seems it is two step to calling the number. If I type the number direct to my client softphone, it calls directly the number without show me to choose Answer to calling. First call connect with client and then come into my screen and showing me to choose Answer and Decline.I'm not able to listen ringing sound because call is connecting first with client and then connect with my softphone. My source code is in AMI socket open to make call from web. how can I call direct to the number? -- _ -- 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] Capture dead phone?
Mitch Claborn писал 08.11.2013 02:51: Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Maybe you can connect agents and callers via MeetMe, and when AMI gets the MeetMe Leave event, put the caller on hold and return him to the queue (maybe in the first position). Just a guess, for I've never used such setup. But I strongly agree with people who say you'd better change your hardware. -- With Best Regards Mikhail Lischuk -- _ -- 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
