Re: [asterisk-users] Asterisk SMDI for Nortel Option 11
This is off topic but I have inherited a Nortel Option 11c and am curious what features you are using that make you want to continue using it in conjunction with Asterisk as opposed to moving completely over to Asterisk. I am just learning the Nortel and find it powerful but haven't found the features that I have become accustomed to in Asterisk. -- _ -- 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] Iphone setup
I think siax -from cydia- could also be an alternative as they stated to use natively 3g. I only test WIFI. SIAX on WIFI works SIAX on WIFI works great so far. I don't have a router that i can secure my network with so I haven't tested it over 3G yet. I plan on doing that soon. Putting SIAX in the background only works for a little while. Also it does hangup the call if a call comes in on the regular number. That is of course a problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iphone setup
Thank you for the heads up. I will look into both weephone and voipover3g ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iphone setup
I would like to setup an iphone to be an extension on my pbx. I have looked at SIAX as well as Asteriskc2d. Does anyone have any experience with either of these or another app? The important thing for me is that I can run it in the background so I can always be available to receive a call. It has to work over both 3g as well as wifi so I can take advantage of my data plan and not use minutes on my phone. Thank you in advance for any info you may have. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spurious hangups on Sangoma A102d, Trixbox 2.6.1
On Fri, Jan 9, 2009 at 2:33 PM, Jeff LaCoursiere j...@jeff.net wrote: [also posted on Trixbox trunk forum] I am also working with Sangoma directly to debug this, but so far no real luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE 3.2.6 (3.2.7 is out, but nothing has changed that would affect this problem). The system gets about 200 calls inbound on the trunk, which is not very heavily used, and of those calls one or two a day is randomly terminated in the middle of a call. Other than this problem everything is working fine. All phones are Polycom IP501 with latest firmware as of a year ago... There is only one ethernet switch (Linksys 100/1000 managed) between the phones and the Trixbox, and the runs are less than 50 feet. Calls extension to extension seem to have no issue at all. The network *is* shared data/voice with no QOS and no virtual segments, but if the network was the issue I would expect to see extension to extension calls report this issue, which they have not. This is actually a hotel, and the data portion of the traffic isn't heavily used either. They don't even have a file server. I have the full logging enabled, and here is an excerpt of a call that was terminated. You can see the conversation lasted about forty seconds before it was hungup. [snipped the beginning of this process...] [Jan 9 12:34:12] VERBOSE[2778] logger.c: -- Executing [...@macro-dial:7] Dial(Zap/9-1, SIP/2607SIP/2605SIP/2510|20|trM(auto-blkvm)) in new stack [Jan 9 12:34:12] VERBOSE[2778] logger.c: -- Called 2607 [Jan 9 12:34:12] VERBOSE[2778] logger.c: -- Called 2605 [Jan 9 12:34:12] VERBOSE[2778] logger.c: -- Called 2510 [Jan 9 12:34:12] VERBOSE[2778] logger.c: -- SIP/2510-0a29a140 is ringing [Jan 9 12:34:12] VERBOSE[2778] logger.c: -- SIP/2607-0a30c8d0 is ringing [Jan 9 12:34:12] VERBOSE[2778] logger.c: -- SIP/2605-0a372cb0 is ringing [Jan 9 12:34:21] VERBOSE[2778] logger.c: -- SIP/2605-0a372cb0 answered Zap/9-1 [Jan 9 12:34:21] VERBOSE[2778] logger.c: -- Executing [...@macro-auto-blkvm:1] Set(SIP/2605-0a372cb0, __MACRO_RESULT=) in new stack [Jan 9 12:34:21] DEBUG[2778] app_macro.c: Executed application: Set [Jan 9 12:34:21] VERBOSE[2778] logger.c: -- Executing [...@macro-auto-blkvm:2] Set(SIP/2605-0a372cb0, __CWIGNORE=) in new stack [Jan 9 12:34:21] DEBUG[2778] app_macro.c: Executed application: Set [Jan 9 12:34:21] VERBOSE[2778] logger.c: -- Executing [...@macro-auto-blkvm:3] DBdel(SIP/2605-0a372cb0, BLKVM/602/Zap/9-1) in new stack [Jan 9 12:34:21] VERBOSE[2778] logger.c: -- DBdel: family=BLKVM, key=602/Zap/9-1 [Jan 9 12:34:21] DEBUG[2778] app_macro.c: Executed application: DBDel [Jan 9 12:34:21] DEBUG[2778] app_dial.c: Macro exited with status 0 [Jan 9 12:34:21] DEBUG[2778] chan_zap.c: Took Zap/9-1 off hook [Jan 9 12:35:01] VERBOSE[2778] logger.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'Zap/9-1' in macro 'dial' [Jan 9 12:35:01] VERBOSE[2778] logger.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'Zap/9-1' [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Executing [...@macro-dial:1] Macro(Zap/9-1, hangupcall) in new stack [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Executing [...@macro-hangupcall:1] ResetCDR(Zap/9-1, w) in new stack [Jan 9 12:35:01] DEBUG[2778] app_macro.c: Executed application: ResetCDR [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Executing [...@macro-hangupcall:2] NoCDR(Zap/9-1, ) in new stack [Jan 9 12:35:01] DEBUG[2778] app_macro.c: Executed application: NoCDR [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Executing [...@macro-hangupcall:3] GotoIf(Zap/9-1, 1?skiprg) in new stack [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Goto (macro-hangupcall,s,6) [Jan 9 12:35:01] DEBUG[2778] app_macro.c: Executed application: GotoIf [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Executing [...@macro-hangupcall:6] GotoIf(Zap/9-1, 0?skipblkvm) in new stack [Jan 9 12:35:01] DEBUG[2778] app_macro.c: Executed application: GotoIf [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Executing [...@macro-hangupcall:7] NoOp(Zap/9-1, Cleaning Up Block VM Flag: BLKVM/602/Zap/9-1) in new stack [Jan 9 12:35:01] DEBUG[2778] app_macro.c: Executed application: Noop [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Executing [...@macro-hangupcall:8] DBdel(Zap/9-1, BLKVM/602/Zap/9-1) in new stack [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- DBdel: family=BLKVM, key=602/Zap/9-1 [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- DBdel: Error deleting key from database. [Jan 9 12:35:01] DEBUG[2778] app_macro.c: Executed application: DBDel [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Executing [...@macro-hangupcall:9] GotoIf(Zap/9-1, 1?theend) in new stack [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Goto (macro-hangupcall,s,11) [Jan 9 12:35:01] DEBUG[2778] app_macro.c: Executed application: GotoIf [Jan 9 12:35:01] VERBOSE[2778] logger.c: -- Executing [...@macro-hangupcall:11] Hangup(Zap/9-1, ) in new stack [Jan 9 12:35:01]
Re: [asterisk-users] Spurious hangups on Sangoma A102d, Trixbox 2.6.1
On Fri, Jan 9, 2009 at 4:46 PM, Jeff LaCoursiere j...@jeff.net wrote: On Fri, 9 Jan 2009, James Noble wrote: I had the same problem with a sangoma card and a clean install of asterisk as well as a trixbox set up. I finally started using a vegastream to handle the T1 connections and was able to get rid of the problem. James $5K for a sinlge T1? Thats an expensive solution! PRI or RBS T1? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yeah it wasn't cheap but we dropped enough calls that it was worth it. We had both PRI and RBS. I am not sure if the problem presented itself on the PRI but it definitely was a problem on the RBS T1 James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk v1.6 on Ubuntu Intrepid?
Scott, I had the same problem when I downloaded http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz This downloaded asterisk-1.6.0.2.tar.gz To fix the problem I downloaded http://downloads.digium.com/pub/asterisk/asterisk-1.6.0.3-rc1.tar.gz and I was able to compile without any problems. James On Tue, Dec 16, 2008 at 5:55 PM, Scott Berry n7...@northlc.com wrote: Hi Tillman, I am havingthe same problem can you expand on your answer here? I am not sure I understand what your saying. Are you saying that this is really not an Asterisk problem? And just another thought. Where is sentinel coming from? Interesting I wounder if it's something left over from another version of Asterisk from an early version? Scott On Tue, 2008-12-16 at 13:38 -0600, Tilghman Lesher wrote: On Tuesday 16 December 2008 13:14:06 Christian wrote: Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c - manager.o manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 In neither the 1.6.0 branch nor the 1.6.1 branch is SENTINEL used within main/manager. So you're clearly using a third party patch. You need to contact the person from whom you obtained that patch and ask this question. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail.conf: where to fin strftime manual entry?
Here is another resource http://us2.php.net/strftime ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
