Re: [Asterisk-Users] Quad BRI card
On Thursday 18 May 2006 03:35, Mark Coccimiglio wrote: Otherwise the Diva server cards are a good option (extensive, but come highly recomended from most that I hear). Good luck and happy hunting. Ouch, you weren't joking. 1453 Euro! -- Cheers Wayne ___ --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] Pulling the mISDN number from an incoming call
Hi all Which command do I use to pull the mISDN number from an incoming call. -- Cheers Wayne ___ --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] Quad BRI card
Hi all Does Digium make a quad BRI card? I can't see anything of the sort on their page but I thought they might call it something else in the States. Failing that, can anyone recommend a make/model that would handle 4 BRI ports? -- Cheers Wayne ___ --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] Quad BRI card
On Wednesday 17 May 2006 23:14, Hadley Rich wrote: Does Digium make a quad BRI card? I can't see anything of the sort on their page but I thought they might call it something else in the States. They do, but it isn't released yet. Put B410P into google and you will get a couple of hits. Digium's marketing page says it is available and the distributor I use had one on show the other day so they can't be too far away. Failing that, can anyone recommend a make/model that would handle 4 BRI ports? Many people seem to like the Eicon Diva cards. Thanks, I'll give it a shot... -- Cheers Wayne ___ --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] Digium cards, so disappointing !
On Sunday 30 April 2006 10:27, Boris Bakchiev wrote: Opened pseudo zap interface, measuring accuracy... This may be a stupid question but how did you do this? -- Cheers Wayne ___ --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] A worrying article
Forgive me if this is old news... http://www.spectrum.ieee.org.nyud.net:8090/oct05/1846 -- Cheers Wayne ___ --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] Re: Not receiving fax
On Thursday 24 November 2005 11:37, Kristof Hardy wrote: have you tried using direct indialing, to see if rxfax works? (I assume you are now using fax-detection) That way we know if the detection is failing or the receiving itself.. (or both :)) I'm not sure what you mean, are you saying that I should some how circumvent the menu system to make calls go directly to the fax? Then I should listen for noises? Btw, I've changed my zone to za and my log output has changed slightly. It seems that the problem is in dsps busy tone detection. Nov 24 11:12:04 VERBOSE[18523] logger.c: -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/fax/1132823516.1.tif) in new stack Nov 24 11:12:43 DEBUG[18523] dsp.c: ast_dsp_busydetect detected busy, avgtone: 469, avgsilence 494 Nov 24 11:12:43 DEBUG[18523] dsp.c: Requesting Hangup because the busy tone was detected on channel Zap/1-1 Nov 24 11:12:43 DEBUG[18523] app_rxfax.c: Got hangup Nov 24 11:12:43 DEBUG[18523] app_macro.c: Extension s, priority 3 returned normally even though call was hung up Nov 24 11:12:43 DEBUG[18523] pbx.c: Extension in_fax, priority 2 returned normally even though call was hung up -- Regards Wayne Gemmell Work: +27 11 894 2530 Fax : +27 11 894 4081 Cell: +27 83 666 3325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Not receiving fax
On Thursday 24 November 2005 12:49, Kristof Hardy wrote: Yes, make a 'default' to go directly to your fax-receive macro. (rxfax witht the parameters) At least you should hear a 'fax' answering. Thanks, I'll try that. Hm, you could try enabling the busy detection in your zapata file.. As of my last posted log it was enabled and set to 15. -- Regards Wayne Gemmell Work: +27 11 894 2530 Fax : +27 11 894 4081 Cell: +27 83 666 3325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Not receiving fax
On Thursday 24 November 2005 12:49, Kristof Hardy wrote: Yes, make a 'default' to go directly to your fax-receive macro. (rxfax witht the parameters) At least you should hear a 'fax' answering. Yes, I hear a fax answering, so at least I know its working. -- Regards Wayne Gemmell Work: +27 11 894 2530 Fax : +27 11 894 4081 Cell: +27 83 666 3325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not receiving fax
VERBOSE[6125] logger.c: -- Set Digit Timeout to 3 Nov 23 22:35:51 VERBOSE[6125] logger.c: -- Executing ResponseTimeout(Zap/1-1, 7) in new stack Nov 23 22:35:51 VERBOSE[6125] logger.c: -- Set Response Timeout to 7 Nov 23 22:35:51 VERBOSE[6125] logger.c: -- Executing BackGround(Zap/1-1, custom/aa_1) in new stack Nov 23 22:35:51 DEBUG[6125] channel.c: Scheduling timer at 160 sample intervals Nov 23 22:35:51 VERBOSE[6125] logger.c: -- Playing 'custom/aa_1' (language 'en') Nov 23 22:35:52 DEBUG[6125] chan_zap.c: DTMF digit: f on Zap/1-1 Nov 23 22:35:52 VERBOSE[6125] logger.c: -- Redirecting Zap/1-1 to fax extension Nov 23 22:35:52 DEBUG[6125] channel.c: Scheduling timer at 0 sample intervals Nov 23 22:35:52 VERBOSE[6125] logger.c: == Spawn extension (aa_1, fax, 0) exited non-zero on 'Zap/1-1' Nov 23 22:35:52 VERBOSE[6125] logger.c: -- Executing Goto(Zap/1-1, ext-fax|in_fax|1) in new stack Nov 23 22:35:52 VERBOSE[6125] logger.c: -- Goto (ext-fax,in_fax,1) Nov 23 22:35:52 DEBUG[6125] pbx.c: Expression result is '1' Nov 23 22:35:52 VERBOSE[6125] logger.c: -- Executing GotoIf(Zap/1-1, 1?2:analog_fax|1) in new stack Nov 23 22:35:52 VERBOSE[6125] logger.c: -- Goto (ext-fax,in_fax,2) Nov 23 22:35:52 VERBOSE[6125] logger.c: -- Executing Macro(Zap/1-1, faxreceive) in new stack Nov 23 22:35:52 VERBOSE[6125] logger.c: -- Executing SetVar(Zap/1-1, FAXFILE=/var/spool/asterisk/fax/1132778143.93.tif) in new stack Nov 23 22:35:52 VERBOSE[6125] logger.c: -- Executing SetVar(Zap/1-1, [EMAIL PROTECTED]) in new stack Nov 23 22:35:52 VERBOSE[6125] logger.c: -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/fax/1132778143.93.tif) in new stack Nov 23 22:35:52 DEBUG[6125] chan_zap.c: DTMF digit: f on Zap/1-1 Nov 23 22:35:52 DEBUG[6125] chan_zap.c: Fax already handled Nov 23 22:36:12 DEBUG[6039] manager.c: Manager received command 'Command' Nov 23 22:36:12 DEBUG[6039] manager.c: Manager received command 'Command' Nov 23 22:36:30 DEBUG[3374] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Nov 23 22:36:31 DEBUG[6125] dsp.c: ast_dsp_busydetect detected busy, avgtone: 468, avgsilence 518 Nov 23 22:36:31 DEBUG[6125] dsp.c: Requesting Hangup because the busy tone was detected on channel Zap/1-1 Nov 23 22:36:31 DEBUG[6125] app_rxfax.c: Got hangup Nov 23 22:36:31 DEBUG[6125] app_macro.c: Extension s, priority 3 returned normally even though call was hung up Nov 23 22:36:31 DEBUG[6125] pbx.c: Extension in_fax, priority 2 returned normally even though call was hung up Nov 23 22:36:31 VERBOSE[6125] logger.c: -- Executing Hangup(Zap/1-1, ) in new stack Nov 23 22:36:31 VERBOSE[6125] logger.c: == Spawn extension (ext-fax, h, 1) exited non-zero on 'Zap/1-1' Nov 23 22:36:31 DEBUG[6125] chan_zap.c: Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1 Nov 23 22:36:31 DEBUG[6125] chan_zap.c: disabled echo cancellation on channel 1 Nov 23 22:36:31 DEBUG[6125] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Nov 23 22:36:31 DEBUG[6125] chan_zap.c: Updated conferencing on 1, with 0 conference users Nov 23 22:36:31 VERBOSE[6125] logger.c: -- Hungup 'Zap/1-1' Nov 23 22:36:34 DEBUG[3374] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' /full log zaptel.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=15.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;busydetect=yes ;busycount=15 faxdetect=incoming /zaptel.conf -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AMP installation
On Monday 21 November 2005 17:12, Goran Donev wrote: How do you install AMP? I downloaded it and tried to run make or install and it doesn't work. Is there some trick to this? The trick is to run the install script and read the documentation. Just not in that order... -- Cheers Wayne ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't create iax channel
On Thursday 10 November 2005 10:55, Jason Walker wrote: The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. I don't know where I read it, apparently it is needed for timing or something, could be in the old handbook or hitchikers guide to asterisk as I havn't got far enough into the new handbook to comment. Based on your post, seems that you have an issue with codecs more than creating an IAX trunk. Thanks, yes I was disallowing all codecs, :( -- Cheers Wayne ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't create iax channel
Hi all Could somebody please give me an idea as to whats wrong here. I'm trying to connect 2 servers using IAX, I'm not trunking them because I read that you need zaptel hardware installed at both sides to do the trunking. Theregistration seems to have worked as the output of iax show peers on the side I'm working from is as follows Name/UsernameHost Mask Port Status wayne165.165.164.87 (D) 255.255.255.255 4569 Unmonitored and on the other side iax2 show users shows Username SecretAuthen Def.Context A/C Codec Pref waynepassword 001 default No Host When trying to call from this side to that side I get the following -- Executing Dial(SIP/301-2d50, IAX2/wayne:[EMAIL PROTECTED]/204) in new stack Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: chan_iax2.c:7745 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/wayne-5 -- Hungup 'IAX2/wayne-5' Nov 10 08:37:21 NOTICE[30785]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Congestion(SIP/301-2d50, ) in new stack == Spawn extension (from-internal, 204, 2) exited non-zero on 'SIP/301-2d50' Any ideas? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip phones on x86_64
On Tuesday 04 October 2005 00:42, Rajesh kumar wrote: I am using Kphone which works great for my purposes! You can look at twinklephone as well at http://www.twinklephone.com/ Hi, thanks all for the info, kphone does really wierd stuff and I can't get twinkle to compile. I'm looking into that gnomeeting CVS idea. -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip phones on x86_64
Hi all Can anyone recommend a good soft phone that can compile on x86_64 (linux) platform? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstriCon 2006 Location
How about someplace central like South Africa? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not enough lines available for Asterisk implemetation
Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt flood. 1) is my info okay so far? 2)What would be the best way for be to implement the other 22 lines? Is there hardware I'm not aware of? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)
On Thursday 08 September 2005 16:26, Simone Cittadini wrote: My boss is just asking me if it is possible to stuck 4* TE411P in a Doesn't that equal 16 lines, not 480 lines? Or did I miss something? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)
On Thursday 08 September 2005 13:12, asterisk groups wrote: You might want to offload some of that PRI termination to an external device such as a Cisco AS53XX, Lucent MAX TNT, Audio Codes or Redfone fonebridge device and then trunk it to your Asterisk servers. But putting more then 2 quad cards in a single server is not safe. 1 per server would be more acceptable. This link might be helpful to you: http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large Thanks for the info, its exactly what I'm looking for. I knew things didn't make sense. Thanks Wayne Gemmell ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Sendmail question
I think it would help if you sent an excerpt from your maillog. Cheers Wayne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
