[asterisk-users] UK Colocation services
Can anyone direct me to a colo provider in the UKwhere I can park an asterisk server and buy UK toll free inbound services over SIP? Thanks Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --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] Chanspy crashing the server, again
I upgraded to 1.2.12.1 - the problem is much, MUCH worse. Before, it was once or twice per week, now EVERY TIME someone uses chanspy it crashes the machine. Anyone with any ideas? This is killing me. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --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] Chanspy crashing the server, again
Yes, I deleted the old modules. All the modules come up listed as the date of the upgrade. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --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] Chanspy crashing server, again
Here is the log excerpt from rright before the crash: Sep 18 09:58:58 NOTICE[23492] app_chanspy.c: Attaching SIP/403811-0a271260 to SIP/403415-0a2d8020 Sep 18 09:59:17 NOTICE[23492] app_chanspy.c: Attaching SIP/403811-0a271260 to SIP/403425-0a2f0cf8 Sep 18 09:59:53 WARNING[23698] pbx.c: Timeout, but no rule 't' in context 'qa' Sep 18 09:59:55 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403429-0a18 Sep 18 10:00:03 WARNING[23689] pbx.c: Timeout, but no rule 't' in context 'custcare' Sep 18 10:01:57 WARNING[23977] pbx.c: Timeout, but no rule 't' in context 'custcare' Sep 18 10:02:19 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403423-0a37c0e0 Sep 18 10:02:48 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403416-0a384540 Sep 18 10:05:40 NOTICE[23492] app_chanspy.c: Attaching SIP/403811-0a271260 to SIP/403404-0a1e6208 Sep 18 10:06:36 WARNING[23492] file.c: Failed to write frame Sep 18 10:10:09 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403403-0a1e6208 Sep 18 10:11:12 WARNING[23339] channel.c: Avoided initial deadlock for '0xb7a1c4f8', 10 retries! Sep 18 10:11:57 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403423-b7ad3d58 Sep 18 10:12:33 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403414-b7a9d748 Sep 18 10:13:39 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403426-0a49d028 Sep 18 10:14:49 WARNING[26819] pbx.c: Timeout, but no rule 't' in context 'CancelSave' Sep 18 10:15:05 ERROR[23351] chan_sip.c: We could NOT get the channel lock for SIP/403101-b7844c38! Sep 18 10:15:05 ERROR[23351] chan_sip.c: SIP MESSAGE JUST IGNORED: ACK Sep 18 10:15:05 ERROR[23351] chan_sip.c: BAD! BAD! BAD! The problem I have withthis though, is that 403101 was not being monitored or monitoring a call. There must be something else. I downgraded back to 1.2.7.1, everything seems to be stable again. I am going to test chanspy later and see if it works at all. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --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 crashing when monitoring SIP device with Chanspy
Forgive me ifthis thread already exists - Am running asterisk 1.2.6.1 - Every so often, asterisk crashes when doing a Chanspy on a SIP phone monitoring a sip phone. Dialplan is set up as follows: exten = 6274,1,ChanSpy(SIP/403274) Intra-building we dial 3 digit extensions, monitor is obviously 6+3 digit extension I don't think it's a utilization issue, itseems to happened when there were 80 phones on the network, and still happens with 160 phones. I can tell when it happens, a manager will call me and say they have 15 calls in queue but no calls are getting to the agents, then a minutelater I get a call complaining about 1 way audio, then the asterisk daemon stops - I have tostart asterisk. There is nothing in the log file either - except the last entry being NOTICE -ChanSpy attaching toSIP/blahblahthen no entries after. I saw in bug tracker 3851 about people having crashing issues but that was over a year ago. Anyone else having problems or enlighten me on a fix? Thanks Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --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] TNT Max Config
I am looking for someone who know what they are doing with a TnT MAX to help me get started with configuring the thing. The unit will have 6 PRI's and 18 EM T1's going into it and sending the calls out VoIP to asterisk boxes and to upstream voip providers. Has 3 x 8T1 cards and 8 x 96 VoIP DSP cards. Willing to pay for your time. Email me at mezzmor at aim dot com. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --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] starvox communications
Anyone have any experience with them...specifically, how to actually use their service from an asterisk server? They only do SIP, have no proxy, don't accept registrations, and only allow or disallow calls based on client IP address. I am at a loss as to how to make asterisk work this way. Thanks Jason Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --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 with an external predictive dialer
Does anyone have any experience using asterisk with an external predicitve dialer, like MediaTel? Specifically: The predictive dialer dials out over T1 circuits. It connects to asterisk via amphenol cable from an fxs card in the dialer to asterisk with a tdm2406 fxo card. In the analog world, the dialer dials out through the t1 circuit, and the fxs card is plugged into a 66 block so the extensions are basically hard wired. The phone is off hook, and when the dialer completes a call, it beeps over the phone and the agent knows a call has been connected. After the beep the agent starts talking. So, now I need put an asterisk box between the predictive dialer with a sip softphone at the agent. Since the channels are always off hook, how am I going to get the sip client to work like the analog phone version? And no, we aren't a cold call spit shop. These calls are for existing customers that own the product. Reasoning behind this - equipment consolidation and the ability to switch campaigns to different call centers on the fly. Any ideas would be greatly appreciated. Thanks! Jason Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --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
