[asterisk-users] RTP suppress during calls - Asterisk 1.8.*
Im facing some problems with RTP during queue agent calls. Randomly during the call the agent can't hear the other side. This happens for two or three seconds and the the call continue without problems. The weird thing is that the recording for this call is fine, so both sides are recorded without interruption. I can hear both sides. When this problem happen, all agents that is on call get the problem. I've tried these versions: Asterisk 1.8.30.0 built by root @ weon264 on a x86_64 running Linux on 2014-09-23 16:32:21 UTC Asterisk 1.8.15-cert5 built by root @ Asterisk1.8Felipe on a x86_64 running Linux on 2014-06-24 19:35:14 UTC Asterisk 1.8.32.2 built by root @ weon264 on a x86_64 running Linux on 2015-02-19 19:46:43 UTC I have 5 different Asterisk servers on different networks facing the same problem. Again, the recording is fine. Both sides sent RTP to Asterisk. /deg -- _ -- 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] Realtime Extensions -- Comments?
Im not sure, but there is a "commented" column that could have 0(not commented) or 1(commented) as values. Is this right? P.S.: I got it from voip--info.org on the realtime Static page... D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1610A CEP: 80530-000 - Curitiba - PR +55 41 3252-2977 r 101 http://www.isolve.com.br On 22 de ago de 2006, at 11:20, Douglas Garstang wrote: The unofficial docs on the voip wiki for the realtime extensions table structure is: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Uhm... what abouts comments? What if I wanted to temporarily deactivate a couple of extensions? Without a comment flag, I'd have to completely remove those entries from the extensions table! That's not very friendly is it... Is there a better way? Doug. ___ --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 ___ --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: [asterisk-dev] Recent additions to the Digium Asterisk development team
Very good news. Really good to know about the success of companies(like Digium) and developers(like all mentioned by Kevin) that are working with and for the Asterisk community. I just have one thing to complain: When will Digium invite a developer to put the MFCR2 stack(channel) on Asterisk official core? Keep in mind that South American/Asian markets is growing UP pretty faster on VoIP, and of course Asterisk is one of the tools that have been used to get this grow. MFCR2 is almost on 90% of all telephony carriers in Brazil. I'm the founder of AsteriskBrasil.org(born on 2004), we have 5000 users and 2000 members on the email discussion list/IRC. All of them are using MFCR2, implemented by Steve Underwood that deserves all of AsteriskBrasil.org community's respect. The VERY GOOD work done by Steve on the chan_unicall, spandsp and libmfcr2 turn on the possibility to work with Asterisk in Brazil, but is a pain to apply a patch every time a new Asterisk version is announced, is pain to maintain two software trees. AsteriskBrasil.org has its own developers that is doing a very good work on translating, coding and recoding things to work in Brazil (some of limfr2 stuff, voicemail, grammar, etc -I'll prepare a full list-) that should help the Asterisk dev team to put some of our needs on the core. I'll not write more lines here, I just wanna know: "Is Digium interested to keep/grow business in South America/Asia?" Thanks for all of you specially for Steve(coppice). Denis Galvão AsteriskBrasil.org On 16 de ago de 2006, at 19:12, Kevin P. Fleming wrote: Some of you may have noticed some new people with '@digium.com' email addresses lately... yes, we have been hiring to expand our Asterisk development team and I should have made an official announcement some time ago :-) Joshua Colp joined our development team a few months ago. Josh (file on IRC/Mantis) has been working on Asterisk development for quite some time and had contributed many features and bug fixes as a volunteer community member, along with being very active on the IRC channels and issue tracker. Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements. Jason Parker joined our development team at the beginning of this week. Jason (qwell on IRC/Mantis) has been maintaining the chan_skinny driver for Cisco SCCP phones as well acting as a bug marshal and fixing various bugs in Asterisk for the past year or more. Russell Bryant has been a Digium part-time employee and an active Asterisk maintainer since before I got involved with Asterisk :-) His contributions are innumerable, and he has worked far more than the 'ten to twenty hours per week' he claims to have available outside of his school work! Russell (russellb on IRC/Mantis) will be joining us full time in Huntsville after the winter semester is complete, when he expects to graduate. Please join me in welcoming all these new members of our development team; they are helping to make Asterisk (and our other software products) better every day and will enable us to accelerate our products into the future. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --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] DEBUG[13314]: Didn't get a frame from channel: SIP/
Someone could help me on troubleshooting this error?
DEBUG[13314]: Didn't get a frame from channel: SIP/
When passing a fax over a PRI channel I got this error after the 4th
page. Evereything is ok if the fax has 3 pages, but on forth I got a
hangup and this message appeared on my full log:
DEBUG[13314]: Didn't get a frame from channel: SIP/
Is there some parameter that could handle this timeout? I saw
something on the channel.c:
/* Calculate the appropriate max sleep interval - in general, this is
the time,
left to the closest jb delivery moment */
if (jb_in_use)
to = ast_jb_get_when_to_wakeup(c0, c1, to);
who = ast_waitfor_n(cs, 2, &to);
if (!who) {
/* No frame received within the specified timeout - check if we
have to deliver now */
if (jb_in_use)
ast_jb_get_and_deliver(c0, c1);
if (c0->_softhangup == AST_SOFTHANGUP_UNBRIDGE || c1->_softhangup
== AST_SOFTHANGUP_UNBRIDGE) {
if (c0->_softhangup == AST_SOFTHANGUP_UNBRIDGE)
c0->_softhangup = 0;
if (c1->_softhangup == AST_SOFTHANGUP_UNBRIDGE)
c1->_softhangup = 0;
c0->_bridge = c1;
c1->_bridge = c0;
}
continue;
}
f = ast_read(who);
if (!f) {
*fo = NULL;
*rc = who;
ast_log(LOG_DEBUG, "Didn't get a frame from channel: %s\n",who-
>name);
break;
}
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1610A
CEP: 80530-000 - Curitiba - PR
+55 41 3252-2977 r 101
http://www.isolve.com.br
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Re: [Asterisk-Users] TE420P/TE415P?
Hi Kevin. Where could I get more information about those boards? Thanks, D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1610A CEP: 80530-000 - Curitiba - PR +55 41 3252-2977 r 101 http://www.isolve.com.br On 25 de jun de 2006, at 07:07, Kevin P. Fleming wrote: - C F <[EMAIL PROTECTED]> wrote: I like the TC400P card, how many T1s will that take? or is it just a Daughter card on the TE4xx ? How many channels can it transcode? Neither. It's a separate device, entirely unrelated to any TDM cards (which means it can be used for any type of channel, not just TDM). The final specs for the number of channels are not yet determined, but we expect to do at least 100 channels of G.729 and/or G.723.1 per board. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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 ___ --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] Using "#include" on zaptel.conf
But the zaptel.conf is an Asterisk file? Thanks for your reply and you're right about testing before. :) D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1610A CEP: 80530-000 - Curitiba - PR +55 41 3252-2977 r 101 http://www.isolve.com.br On 09 de jun de 2006, at 14:32, Kevin P. Fleming wrote: - Denis Galvão - iSolve <[EMAIL PROTECTED]> wrote: Is this possible to use an include parameter on zaptel.conf file? All Asterisk .conf files support #include, it's handled at the file- reading level. It would have taken less time to just try it, though, and you'd already have your answer :-) -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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 ___ --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] Using "#include" on zaptel.conf
Hi all. Is this possible to use an include parameter on zaptel.conf file? I mean, I want to have a bunch of files with zaptel configurations, each one with the configuration of one kind of board(TDM, analog, and so on). Thanks, Denis Galvão ___ --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] Polycom IP 301 is slow
The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. Denis. On 26 de mar de 2006, at 21:17, Avi Miller wrote: Nick Hoffman wrote: Hrm, well that's disappointing. If they're so slow, why are they so popular? They may be slow to startup, but they're great phones. :) Once the phone has started up, it works like a charm and the sound/call quality is fantastic. -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ .>> Open Source - Own it - Squiz.net ./> ___ --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 ___ --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] Google Analytics and voip-info.org
Damned! What is going on with voip-info.org this week? I think Google Analytics is the cause... Has anybody facing this problem too? Denis. ___ --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] Realtime Subscribecontext
Could DUNDI help him? Or maybe a OpenSER plus Asterisk environment... Denis. On 12 de dez de 2005, at 12:41, Kevin P. Fleming wrote: Douglas Garstang wrote: The issues of NAT, call limit handling and registration expiration don't sound quite so bad. I think we can live with those, if we can in fact just get a central location database. Do you have any suggestions or ideas about how this can be implemented with Asterisk? Because, honestly, right now this current limitation is proving to be a real thorn in our side. There is no known answer at this time; there are many discussions occurring about this topic and various ways of addressing it, but they are all theoretical at this point and nobody has come up with a solid design. ___ --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 ___ --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] Using RxFAX and TxFAX together
Steve, Im receiving FAXes from an IP connection... This is what Im talking about: Asterisk - RxFAX - VoIP provider - PSTN - FAX Denis. On 16 de nov de 2005, at 12:34, Steve Underwood wrote: app_rxfax and app_txfax do not work across VoIP channels. ___ --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] WiFi Phones
Wait for the next UTStarCom version... Called F3000, Im not sure, but something like that. It will have better battery performance and will have 802.11g support, and many other improvements. It will be available soon. Denis. On 07 de out de 2005, at 00:54, Andy Hamilton wrote: Anyone have good words to say about any of the WiFi handsets currently available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). The unit, I'm guessing, was designed somewhere in Asia, and the language translation shows it a little bit. Sound quality seems pretty good for the few calls I've passed through it. I only have one AP in my house, so I can't comment on roaming. The headset for my cell phone is stereo, and I think the phone would be most happy with a standard 3 conductor plug, but I imagine a headset on a phone is a headset on a phone. The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available "security" option. Overall: I would recommend purchasing one, for testing at the very least. They are well priced and of good quality. Battery life seems to be pretty good, too. -A ___ --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 ___ --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 not make call with Unicall (MFC/R2)
Put on the list the software version that you are using. D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 3252-2977 r http://www.isolve.com.br On 09 de set de 2005, at 02:29, Le Van Khoa wrote: Hi, I run the program testcall with one E1, it works fine; I receive DNIS and ANI for making calls and answering calls. When I start the Asterisk I receive call from outside correctly including DNIS and ANI, and receive the following messages: Sep 7 10:29:59 WARNING[12167]: Answer Call Sep 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 Call control(5) Sep 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 Answer call Sep 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 0101 -> [1/ 20/Group B /Accepted Paid] Sep 7 10:29:59 WARNING[12167]: Unicall/2 event Answered Sep 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 Channel echo cancel Sep 7 10:33:13 WARNING[12167]: Timeout, but no rule 't' in context 'aa_1' Sep 7 10:33:13 WARNING[12167]: MFC/R2 UniCall/2 Channel gains Sep 7 10:33:13 WARNING[12167]: MFC/R2 UniCall/2 Channel switching Sep 7 10:33:13 WARNING[12167]: MFC/R2 UniCall/2 Call control(6) Sep 7 10:33:13 WARNING[12167]: MFC/R2 UniCall/2 Drop call (cause=Normal Clearing [16]) Sep 7 10:33:13 WARNING[12167]: MFC/R2 UniCall/2 1101 -> [1/ 400/Answer/Accepted Paid] Sep 7 10:33:14 WARNING[12167]: MFC/R2 UniCall/2 <- 1001 [1/ 400/Clear back/Accepted Paid] Sep 7 10:33:14 WARNING[12167]: MFC/R2 UniCall/2 Call disconnected(cause=Normal Clearing [16]) - state 0x400 Sep 7 10:33:14 WARNING[12167]: Unicall/2 event Drop call Sep 7 10:33:14 WARNING[12167]: MFC/R2 UniCall/2 Call control(7) Sep 7 10:33:14 WARNING[12167]: MFC/R2 UniCall/2 Release call Sep 7 10:33:14 WARNING[12167]: MFC/R2 UniCall/2 1001 -> [1/ 1000/Clear back/Accepted Paid] Sep 7 10:33:14 WARNING[12167]: MFC/R2 UniCall/2 Release guard expired Sep 7 10:33:14 WARNING[12167]: MFC/R2 UniCall/2 Destroying call with CRN 32770 Sep 7 10:33:14 WARNING[12167]: Unicall/2 event Release call Sep 7 10:33:14 WARNING[12167]: MFC/R2 UniCall/2 Channel echo cancel Sep 7 10:33:32 WARNING[12167]: MFC/R2 UniCall/1 <- 0001 [1/ 1/Idle /Idle ] Sep 7 10:33:32 WARNING[12167]: MFC/R2 UniCall/1 Detected Sep 7 10:33:32 WARNING[12167]: MFC/R2 UniCall/1 Making a new call with CRN 32776 Sep 7 10:33:32 WARNING[12167]: MFC/R2 UniCall/1 1101 -> [2/ 2/Idle /Idle ] Sep 7 10:33:32 WARNING[12167]: Unicall/1 event Detected Sep 7 10:33:32 WARNING[12167]: MFC/R2 UniCall/1 <- 8 on [2/ 2/Seize ack /Seize ack] Sep 7 10:33:32 WARNING[12167]: MFC/R2 UniCall/1 5 on -> [2/ 2/Seize ack /Seize ack] Sep 7 10:33:32 WARNING[12167]: MFC/R2 UniCall/1 <- 8 off [2/ 2/Group A /Category req ] Sep 7 10:33:32 WARNING[12167]: MFC/R2 UniCall/1 5 off -> [2/ 2/Group A /Category req ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 <- 1 on [2/ 2/Group A /Category req ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 5 on -> [2/ 2/Group A /Category req ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 <- 1 off [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 5 off -> [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 <- 6 on [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 5 on -> [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 <- 6 off [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 5 off -> [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 <- 6 on [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 5 on -> [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 <- 6 off [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 5 off -> [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 <- 8 on [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 5 on -> [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 <- 8 off [2/ 2/Group A /ANI request ] Sep 7 10:33:33 WARNING[12167]: MFC/R2 UniCall/1 5 off -> [2/ 2/Group A /ANI request ] Sep 7 10:33:34 WARNING[12167]: MFC/R2 UniCall/1 <- 6 on [2/ 2/Group A /ANI request ] Sep 7 10:33:34 WARNING[12167]: MFC/R2 UniCall/1 5 on -> [2/ 2/Group A /ANI request ] Sep 7 10:33:34 WARNING[12167]: MFC/R2 UniCall/1 <- 6 off [2/ 2/Group A /ANI request ] Sep 7 10:33:34 WARNING[12167]: MFC/R2 UniCall/1 5 off -> [2/ 2/Group A /ANI request ] Sep 7 10:33:34 WARNING[12167]: MFC/R2 UniC
Re: [Asterisk-Users] unicall and cvs head
Did you use the 1.1.x version of the patch and chan_unicall.c ? Denis. On 05 de set de 2005, at 20:57, Anton Krall wrote: Guys. Anybody gotten unicall to compile under cvs-head? I get a lot of errors while under 1.0.9 everything compiled without a hickup. Any hints? ___ --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 ___ --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] unicall deploy
Hi Guilhermo. Could you share with us your experience? What is the hardware(CPU, RAM, etc) that are you using for this server? What is your Linux distribution? How many concurrent calls do you have in the high traffic moment? Which is the unicall version that are you using? Thanks a lot! D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 3252-2977 r http://www.isolve.com.br On 04 de set de 2005, at 01:06, Guillermo Freige wrote: I´m using an unicall box with 4 E1 lines getting between 6000-15000 calls per day, and between 15-30 operators using AgentLogin, all using R2 signaling to the telco and a local PBX. I´m using the Argentina variant, and using the last version of unicall 0.0.2 and asterisk 1.0.7 Guillermo From: acriollo <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non- Commercial Discussion To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] unicall deploy Date: Sat, 3 Sep 2005 15:04:20 -0500 Hi every one . There are any out there that have a unicall deploy working without problem ? Can give me some tips or referenece about his config ? Regards ___ --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 ___ --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 ___ --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] 100% CPU with Unicall and * head
But which packages are you using? libunicall spandsp asterisk zaptel D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 3252-2977 r http://www.isolve.com.br On 26 de jul de 2005, at 12:27, Jose Chiantera wrote: Hi denis I am using Country ve,10,4Venezuela 10 ani 4 dnis please let me know if I can do some test, or anything to help Thanks - Original Message - From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, July 25, 2005 11:39 PM Subject: Re: [Asterisk-Users] 100% CPU with Unicall and * head Hi Jose. What is the packages version that are you using? What MFCR2 variant are you using, I mean, wich country? Maybe Steve could help us on it. I told him about this problem. Keep in touch. Denis. On 25 de jul de 2005, at 15:36, Jose Chiantera wrote: Hi, I got the same error, when call from IP to digital link using MFCR2, I thinks the problem is a event not managed, If you find a correction for this problem please let me know. Maybe the error in the program channel.c, but I am not sure, now I put some traces to try find what kind of event is. regards Jose - Original Message - From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> To: "Asterisk Users" Sent: Monday, July 25, 2005 9:47 AM Subject: [Asterisk-Users] 100% CPU with Unicall and * head Hi all. When I place a call Im getting this error: Jul 25 09:50:07 WARNING[3200] channel.c: Exception flag set on 'UniCall/13-1', but no exception handler Lots of this messages appeared on my Asterisk full log and the CPU got 100%. Topology: Analog Phone - Hicom300H E1 - E1 Asterisk - IP Trunk Problem: 1. Calls from Analog Phone through Asterisk is ok, but the messages appeared. 2. Calls from IP Trunk to Analog Phone is not ok andd the messages appeared too. System: Fedora Core 2 - Asterisk CVS HEAD(last week) - unicall-0.0.3pre3 - spandsp-0.0.2pre18 unicall.conf [channels] language=br context=from-internal usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 loglevel=255 protocolclass=mfcr2 protocolvariant=br,10,13 protocolend=co group=1 callerid=asreceived channel=>1-15 channel=>17-31 -- zaptel.conf loadzone = us defaultzone=us span=1,0,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 -- Thanks. Denis Galvão ___ 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 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 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 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 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] 100% CPU with Unicall and * head
Hi Jose. What is the packages version that are you using? What MFCR2 variant are you using, I mean, wich country? Maybe Steve could help us on it. I told him about this problem. Keep in touch. Denis. On 25 de jul de 2005, at 15:36, Jose Chiantera wrote: Hi, I got the same error, when call from IP to digital link using MFCR2, I thinks the problem is a event not managed, If you find a correction for this problem please let me know. Maybe the error in the program channel.c, but I am not sure, now I put some traces to try find what kind of event is. regards Jose - Original Message - From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> To: "Asterisk Users" Sent: Monday, July 25, 2005 9:47 AM Subject: [Asterisk-Users] 100% CPU with Unicall and * head Hi all. When I place a call Im getting this error: Jul 25 09:50:07 WARNING[3200] channel.c: Exception flag set on 'UniCall/13-1', but no exception handler Lots of this messages appeared on my Asterisk full log and the CPU got 100%. Topology: Analog Phone - Hicom300H E1 - E1 Asterisk - IP Trunk Problem: 1. Calls from Analog Phone through Asterisk is ok, but the messages appeared. 2. Calls from IP Trunk to Analog Phone is not ok andd the messages appeared too. System: Fedora Core 2 - Asterisk CVS HEAD(last week) - unicall-0.0.3pre3 - spandsp-0.0.2pre18 unicall.conf [channels] language=br context=from-internal usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 loglevel=255 protocolclass=mfcr2 protocolvariant=br,10,13 protocolend=co group=1 callerid=asreceived channel=>1-15 channel=>17-31 -- zaptel.conf loadzone = us defaultzone=us span=1,0,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 -- Thanks. Denis Galvão ___ 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 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 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] 100% CPU with Unicall and * head
Hi all. When I place a call Im getting this error: Jul 25 09:50:07 WARNING[3200] channel.c: Exception flag set on 'UniCall/13-1', but no exception handler Lots of this messages appeared on my Asterisk full log and the CPU got 100%. Topology: Analog Phone - Hicom300H E1 - E1 Asterisk - IP Trunk Problem: 1. Calls from Analog Phone through Asterisk is ok, but the messages appeared. 2. Calls from IP Trunk to Analog Phone is not ok andd the messages appeared too. System: Fedora Core 2 - Asterisk CVS HEAD(last week) - unicall-0.0.3pre3 - spandsp-0.0.2pre18 unicall.conf [channels] language=br context=from-internal usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 loglevel=255 protocolclass=mfcr2 protocolvariant=br,10,13 protocolend=co group=1 callerid=asreceived channel=>1-15 channel=>17-31 -- zaptel.conf loadzone = us defaultzone=us span=1,0,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 -- Thanks. Denis Galvão ___ 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] Asterisk 1.2 is getting closer - please help
Maybe we can have a wiki section with success stories using Asterisk CVS HEAD. Some new features tested and succefully used. It could be a point to start a 1.2 documentation. I'm available to do it, or better, to put some success stories on it. Denis. On 23 de jul de 2005, at 09:52, Olle E. Johansson wrote: Dear Asterisk Community, Asterisk 1.0 was released at Astricon 2004, in September last year. It's been almost a year and we haven't been able to go ahead and release a new version. Now is the time to try to move forward again. As we've outlined before, the process is this: * Code freeze: At this point, we'll stop accepting new additions (new functions) to the source code. Bug fixes are more than welcome, but additions will be postponed until after release and added to the 1.3dev source code base (the new HEAD). * Release candidate: A release candidate will be produced as a tar.gz file on the FTP site. * Release of 1.2: The new release version of Asterisk, that replaces Asterisk 1.0 * Release of 1.2.1: The working version :-) of the new version of Asterisk So why 1.2.1? Well, the common feeling among developers is that "No one really tests anything until we release, so we will receive bug reports from the hour we release 1.2.0". Let's try to prove that they are wrong! What can you do to help this process? - * Set up a test system, and test CVS head in something that resembles your production environment. Scripts, phone, dialplan - make sure you use as many of the features as you can and use in production to make sure they work as expected in version 1.2 * Go wild and test at least two of the new features in 1.2 just for fun and make sure they work as documented. Or document how they work if it's not documented. Test the new realtime architecture, voicemail ODBC storage, AEL - the new scripting language, the new dialplan templates and constructs, the #exec config directive, attended transfers, native music on hold... The list is long. * If you have reported bugs or filed patches in the bugtracker (bugs.digium.com), make sure you reply quickly when a bug marshal or developer ask you questions or require more information. At this point, we're working very hard to clear out outstanding bugs and stabilize the additions that is waiting for inclusion in the CVS. We will close reports that we can't move forward if we do not get any responses. We can re-open later, but need to move forward. If we have a report of a proven bug that needs fixing, those will not be closed. Only unclear reports with no responses will be closed. * Visit the bug tracker at bugs.digium.com and help us test patches. Postitive and negative reports are both equally needed. There's no way a small team of core developers and bug marshals can test everything in there now. We need to decide which patches that are ready for inclusion, that are tested and documented. * If you find that we're missing documentation, please add to the readme files, write new ones. The Asterisk documentation team is ready to help you if you need assistance in this effort. * Disappoint the developers by making sure that the CVS head gets a thorough testing phase now, before release! * Update the Wiki on the 1.2 version. Make sure that you make it very clear that new features only work in 1.2 and releases after that so you won't confuse readers that use older versions. * Test Asterisk CVS head on other platforms than Linux: FreeBSD, OpenBSD, MacOS/X, Commodore VIC 20 - will it work? When is 1.2 scheduled to be released? - At usual with Open Source, we release when the software is ready for release. We do not release when it suits the marketing department, when we need a positive stock report or when customers require it. That said, we now are trying to focus on getting a release out of the door around September 1st. No promises, it all depends on your help and assistance to move forward. Please ask your boss for some time and resources to help the project with testing or dedicate resources within your company to help us. It's Open Source, meaning that everyone works together to make sure we get the software that works for our home, our company or our organization. Finding information --- If you have questions about the developer version, the base for the 1.2 release, use the #asterisk-dev channel on the freenode.net IRC. If you have questions about bug reports and patches, find a bug marshal in the #asterisk-bugs channel. To find out how to download or connect to the IRC channel, please visit http://www.asterisk.org Thank you for your assistance! /Olle - Astricon 2005 - With the Asterisk Solutions Showcase! * Conference, Exhibiti
[Asterisk-Users] Analog extensions behind E1, how to create them?
I will have some extensions behind an E1. All of them will need the features/applications of Asterisk. Analog Extensions <-> PABX E1 <-> E1 Asterisk IP <-> VoIP trunk ^ | | IP Phones How is the best way to create this users on Asterisk? Some of them will have a SIP account to have its extensions with mobile functionality when they will be out of office, others will not have this feature. Some examples will be great! Thanks. Denis. ___ 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] Unicall and Asterisk HEAD
Anybody using Asterisk HEAD with chan_unicall ? Denis. ___ 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] Re: Asterisk 1.1
We are using it too, withouta problem. SipGetHeader and realtime works like charm. I just didn't get spandsp working... It compiled ok, but doesn't work. Denis. On 06 de jul de 2005, at 13:56, Kevin P. Fleming wrote: Tony Mountifield wrote: Anyone here "in the know" about when HEAD will be branched to 1.2? Very soon. We are actively trying to clean up the open bugs and issues so we can prepare a release candidate. ___ 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 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] MeetMe hardware dimensioning
Hi William. On 07 de jul de 2005, at 18:39, William Boehlke wrote: If your users are business people they ratio to 1100 simultaneous business calls and you will need 6-9 Lintel servers, again depending on the conferencing load and the transcoding. I think that I will be in this case. That is a PalTalk like project. What is your opnion about the separation of the services? Would you use the 6-9 lintel to handle each one a separate service, or your plan is to have some redundancy? What is the hardware configuration that you recomend for each server? Xeon 3Ghz each? Thanks. Denis. ___ 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] MeetMe hardware dimensioning
Hi all. What is the best hardware configuration to handle this following scenario? - 4 IVR menu with conference applications for each option; - Only SIP/g711 user access - 3500 simultaneous users(800 at the beginning) - No ZAP channels Where is the most important point of failure? CPU? Ethernet? RAM? Im planning to separate in three servers: Server01: 01 Xeon 3Ghz getting the 1st level of the 4 IVR options. Server02: 01 Xeon 3Ghz with 2 IVR suboptions and 2 conference room Server03: 01 Xeon 3Ghz with 2 IVR suboptions and 2 conference room How it sounds to you? Denis. ___ 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] IAX DTMF Problem...
IAX doesn't use INBAND DTMF. Denis Galvão. On 01 de jul de 2005, at 03:23, Mark Edwards wrote: Hi. Probably been asked before, but my IAX provider assures me its not their problem I have a IAX connection to a peer providing a DID. I am dialing up my number, seeing the DTMF tones come down the line, and the * IVR is just ignoring them. IAX debug output is: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF Subclass: 1 Timestamp: 02608ms SCall: 00016 DCall: 3 [ 210.80.176.12:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02608ms SCall: 3 DCall: 00016 [210.80.176.12:4569] for a press of "1" I am assuming this is the DTMF inband problem, but I appear unable to convince my provider. Can I work around this on * or do I have to go back to SIP? Mark -- regards, Mark P. Edwards TEL:+61 408 601 107 SKYPE: mark.p.edwards ___ 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 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] Asterisk RSS list feeder ready
Where? Denis. On 28 de jun de 2005, at 19:42, Sjaak Nabuurs wrote: Hello Just for fun a rss newsreader for the asterisk users and biz list. Easy to use and now with the complete history to search. Just use it if you like Thanks Sjaak ___ 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 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] UTStarcom F1000 WiFi IP Phone Review
Ok. You're right. Denis. On 25 de jun de 2005, at 15:07, Dan Perik wrote: Not always. Some use a www capture page. When you log in through that page, it opens up that mac/ip for a specified length of time. We're doing that here using nocat (http://nocat.net) Without logging in, no traffic goes through from that mac/ip. - Dan Denis Galvão - iSolve wrote: Hi Steve. I think the proxy authorization is just for WWW access(tcp 80 and 443), if some VoIP port is open you will be able to access your provider without auth. Denis. On 25 de jun de 2005, at 02:22, Steve wrote: I keep getting asked by people if these types of wifi phones are capable at all of getting onto the type of wifi network where you have to login via http (web page) such as is typical at many hotels in the us. ___ 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 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 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] UTStarcom F1000 WiFi IP Phone Review
Hi Steve. I think the proxy authorization is just for WWW access(tcp 80 and 443), if some VoIP port is open you will be able to access your provider without auth. Denis. On 25 de jun de 2005, at 02:22, Steve wrote: I keep getting asked by people if these types of wifi phones are capable at all of getting onto the type of wifi network where you have to login via http (web page) such as is typical at many hotels in the us. ___ 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] Extension Configuration Best Practice
On 21 de jun de 2005, at 17:20, Andrew Kohlsmith wrote: How would you have asterisk know which IP to ring if nobody is registered until the phone rings?? You're right Andrew. I didn't thought about the ring... Honestly -- what's wrong with SIP/location1&SIP/location2&SIP/location3 ? For me, nothing. I would use some AGIs to solve that, or the serial rings like you told. Denis. ___ 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] Extension Configuration Best Practice
On 21 de jun de 2005, at 14:18, Jay Milk wrote: |Rich is indeed correct, Asterisk does not yet support multiple |registrations for a single peer entry. Thus when you register |the previous registration is discarded and the new one is |used. Thus like he said, the last one that registered gets the call. And asterisk will never do that, because that's not how SIP works. Is there a way to just register the phone when user pickup the phone!? In this way we can have two phones regitered with the same context. Denis Galvão AsteriskBrasil.org ___ 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_HEADER example
Hi all. Could someone point me an example to use SIP_HEADER function!? I want to read the "To:" and send this INVITE to an internal extension. Tks. Denis Galvão ___ 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] Peer to Peer calls
Be aware to the codec compatibility between peers. Direct calls to the peer has to be under the same codec and initiation protocol. And, yes, if you have (eg.) SIP and GSM, and careinvite=yes, the media path dont pass through Asterisk. Denis Galvao. On 29/05/2005, at 18:32, Cenk Yabas wrote: Can anybody please answer this. Both clients are behind different NAT's. One of them starts a SIP call to the other through Asterisk. Asterisk sets up the call. Issues reinvite and connects them together. After this point does the media stream flow through Asterisk or Peer to Peer? Does such a call use any system resources of Asterisk server after connection? Thank you in advance.___ 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 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] Asterisk Business Case - Who is using it!?
Hi all. Im participating of a project(a huge one) that will study Asterisk as its PABX base system. They ask me: "Who is using Asterisk as its base PABX!?" Now I ask you: "Anyone know about some important and big company that have been implemented Asterisk!?" Im not talking about VoIP providers... Maybe this question will be the point of a decision to this project. Thanks a lot! Denis. ___ 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] Options in Brazil
If you speck portuguese, visit AsteriskBrasil.org: http://www.asteriskbrasil.org Regards. Denis. Em Qui 03 Mar 2005 22:23, Paul Davidson escreveu: > All- > > I am considering an Asterisk implementation in Brazil. Unfortunately, > this presents something of a challenge to plan sitting in Chicago, > USA. I know there is a large section of Brazillian Asterisk users who > actively read this list- so I'd love to pump out a few questions- > note, I'm not necessarily a newbie, having successfully implemented a > few Asterisk boxes here in the US. > > My primary question revolves around connection hardware- I need to > plug in 8 POTS lines (I've no idea what they'd be called there) to an > Asterisk box. Is digium's TDM400 series availble down there? > Recommended? Undesirable? ATA's? (Sipura, presumably) - channel > banks? > > If anyone has any solid knowledge they can share- gotchas appreciated- > feel free to contact me off list. > > Thanks, > -pbd > ___ > 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 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] MozPhone
Em Qua 02 Mar 2005 16:52, skamp escreveu: > Thats kinda lame who uses their machine and runs apps as root > ughhh, can i install it as root and run it later as the user ? > I installed as normal user... But didnt get the app running Just dont appear... Is there anything else to do!? Denis. ___ 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] How could Asterisk help me on a Internet webcast speech!?
Hi All! I have the folowing need: We have a project in Brazil called Quinta Livre(Free Thursday) where we have one speech about some Open Source project, every last thursday of every month... We want to make this presentation avaliable to more people, so we have to "broadcast" this presentation for everybody that wants to watch it over the Internet (in real time). How could Asterisk help me on it!? We could have some meetme accounts to give to the remote participants, so they could make some questions alive. Something else!? What about video!? Is there anybody here that use it before!? I will apreciate any kind of help. Thanks. Denis Galvão. ___ 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] Conecting to asterisk server through NAT usingIAX
Send us your DIAX configuration. Denis. Em Seg 21 Fev 2005 07:29, Bartosz Wegrzyn - asterisk escreveu: > I did change the port 4569. > Also my router forwards those packets. > > If I start tcpdump port 4569 on my server I receive: > > 04:25:36.061292 IP 192.168.1.253.4569 > > beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 > 04:25:39.154871 IP beu164.neoplus.adsl.tpnet.pl.4569 > > 192.168.1.251.4569: UDP, length 24 > 04:25:39.155919 IP 192.168.1.253.4569 > > beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 > 04:25:44.063009 IP 192.168.1.253.4569 > > beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 > 04:25:46.063463 IP 192.168.1.253.4569 > > beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 24 > 04:25:46.063952 IP 192.168.1.253.4569 > > beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 > 04:25:49.119019 IP beu164.neoplus.adsl.tpnet.pl.4569 > > 192.168.1.251.4569: UDP, length 24 > 04:25:49.120272 IP 192.168.1.253.4569 > > beu164.neoplus.adsl.tpnet.pl.4569: UDP, length 12 > > It means that client is trying to comunicate with asterisk server. > But the client says that the server could not be contacted. > > On asterisk console with iax2 debuging enabled I receive > > Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: > REGREQ Timestamp: 7ms SCall: 1 DCall: 0 > [66.234.228.170:4569] USERNAME: nWv96gaD75 >REFRESH : 60 > > Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > REGAUTH >Timestamp: 00012ms SCall: 00055 DCall: 1 [66.234.228.170:4569] >AUTHMETHODS : 3 >CHALLENGE : 164462354 >USERNAME: nWv96gaD75 > > Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: > REGREQ Timestamp: 00049ms SCall: 1 DCall: 00055 > [66.234.228.170:4569] USERNAME: nWv96gaD75 >REFRESH : 60 >MD5 RESULT : 478939afef8fa0ec5b480cc939dedf6f > > Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: > REGACK Timestamp: 00047ms SCall: 00055 DCall: 1 > [66.234.228.170:4569] USERNAME: nWv96gaD75 >DATE TIME : 173363009 >REFRESH : 60 >APPARENT ADDRES : IPV4 69.208.170.240:4569 > > Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: > ACK Timestamp: 00047ms SCall: 1 DCall: 00055 [66.234.228.170:4569] > Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: > REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] > USERNAME: tester >REFRESH : 300 > > Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > REGAUTH >Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] >AUTHMETHODS : 1 >USERNAME: tester > > Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: > REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] > USERNAME: tester >REFRESH : 300 > > Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] > Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: > REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] > USERNAME: tester >REFRESH : 300 > > Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] > Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: > REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] > USERNAME: tester >REFRESH : 300 > > Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] > Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > REGAUTH >Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] >AUTHMETHODS : 1 >USERNAME: tester > > Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: > REGREQ Timestamp: 3ms SCall: 13354 DCall: 0 [83.28.32.164:4569] > USERNAME: tester >REFRESH : 300 > > Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > ACK Timestamp: 3ms SCall: 2 DCall: 13354 [83.28.32.164:4569] > Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: > LAGRQ Timestamp: 10022ms SCall: 2 DCall: 13354 [83.28.32.164:4569] > Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > REGAUTH >Timestamp: 00019ms SCall: 2 DCall: 13354 [83.28.32.164:4569] >AUTHMETHODS : 1 >USERNAME: tester > > Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: > LAGRQ Timestamp: 10022ms SCall: 2 DCall: 13354 [83.28.32.164:4569] > Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: > REGREQ Timestamp: 3ms SCall:
Re: [Asterisk-Users] DIAX 0.9.10d with Eutectics USB phone suport
With this version I cant use my ATCom usb phone. I didnt see it at the USB Phone options at the DIAX softphone menu. Only yealink and eutectics. Denis. Em Qui 17 Fev 2005 11:44, Dan escreveu: > Hi Denis, > > - Original Message - > From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> > > >> ' - audio delay when IAX bridging inside Asterisk > > > >Will it cover that problem of long delays that we talked before!? > > Yes, with a small remark. > In some situations is possible to loose the audio for the first 2-3s of a > call. > > Best regards, > Dan > > > ___ > 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 > > Spam detection software, running on the system "zeus.avanzada7.com", has > identified this incoming email as possible spam. The original message > has been attached to this so you can view it (if it isn't spam) or label > similar future email. If you have any questions, see > the administrator of that system for details. > > Content preview: Hi Denis, - Original Message - From: "Denis > Galvão - iSolve" <[EMAIL PROTECTED]> >> ' - audio delay when IAX > bridging inside Asterisk > >Will it cover that problem of long delays > that we talked before!? [...] > > Content analysis details: (0.1 points, 5.0 required) > > pts rule name description > -- > -- 0.1 FORGED_RCVD_HELO > Received: contains a forged HELO -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] DIAX 0.9.10d with Eutectics USB phone suport
Hi Dan. > ' - audio delay when IAX bridging inside Asterisk Will it cover that problem of long delays that we talked before!? Regards, Denis Galvão. ___ 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] AsteriskBrasil.org - We have an email list!!!
Im proud to announce that our email list is already working!!! I want to invite all of you to participate in our community! http://www.asteriskbrasil.org We are almost complete with the development of our portal, that will include a lot of resources(translation os white papers, howtos, digium hardware specs, etc.) in brazillian portuguese. Thanks to all of you that support this iniciative and specially Mark Spencer and John Maddodg Hall that give us your support! Have a good discussion guys! Denis Galvão AsteriskBrasil.org ___ 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] Installation on Fedora 3
Em Qua 02 Fev 2005 17:34, Daniel del Castillo escreveu: > I'm having problems trying to run zaptel. I don't have the hardware, I > first want to test out asterisk. The problem is the usb-uhci/usb-ohci > module, it isn't present on the system as same as usbcore and I don't > know why. Any tip? Do you have any USB port!? Denis. ___ 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] *ASTERISK* Install and configure Step by Step.
Hi Max. We are providing a brazillian Asterisk comunity. Our domain is asteriskbrasil.org, and as soon as possible we are providing brazillian portuguese content of Asterisk and all of documents needed to assist you an other brazillians to install/configure and use Asterisk. Asteriskbrasil.org(and other companies) will support an event in Sao Paulo(April 2005) about Asterisk and OpenSource VoIP solutions. The official release will be delivered soon. If you need some help, we have a discussion forum(not yet like asterisk-users) to assist you. Please send me a private email. Um abraço. Denis. Em Qua 02 Fev 2005 08:13, Max escreveu: > Thanks, > > this is payed service in another state (private), I live in SC state > this is only in SP, also, this is not online public Comunity, > > :) > > Max > > - Original Message - > From: "listas iPfone" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, February 02, 2005 7:27 AM > Subject: Re: [Asterisk-Users] *ASTERISK* Install and configure Step by > Step. > > > Hi Max! > > > > "I am Begining in the ASTERISK IP-PABX world, and here in Brazil, have > > not > > > any Help to install and configure," > > > > Sure you have!: > > > > http://www.ipfone.com.br/curso.asp > > > > Miklos > > > > - Original Message - > > From: "Max" <[EMAIL PROTECTED]> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Sent: Tuesday, February 01, 2005 8:36 PM > > Subject: [Asterisk-Users] *ASTERISK* Install and configure Step by > > Step. > > > > > > Hello! > > > > I am Begining in the ASTERISK IP-PABX world, and here in Brazil, have > > not any Help to install and configure, > > > > If you know about any Good LINK contend HOW TO install and configure > > Asterisk to this hardware(minimal) > > > > OR if exist mini linux distro run asterisk in RAM, (similar at > > coyotelinux.com) > > > > bienvenidas todas las ideas! > > > > INTEL MMX CPU 166Mhz > > 32MB Ram > > HD 20GB > > Lan cart 10/100Mb > > Fax modem genius (Lucent chipset) > > Fax Modem USR 33.66 > > Sound OnBoard > > Disk Driver 1.44 > > CD 52X > > > > > > I need Send to my PABX, using only 1 FXS port all incoming Calls from > > Internet I have multiple SIP servers and providers(6 ip lines, vitual > > numbers) > > this is Posible using asterisk? > > > > Thanks in advace, > > > > Max Rivera > > > > > > > > > > --- > >--- > > -- > > > > ___ > > > 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 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 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 -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] Soft phones that _actually_ work under Linux?
Em Qua 02 Fev 2005 01:05, [EMAIL PROTECTED] escreveu: > > Surely there has to be one soft phone that works under Linux. > > > I've tried: > > kphone - it sometimes complains about the need to release the sound > > device > > linphone - lowww > > iaxcomm - needs some strange widgets > > various others - either only supplied as binaries, or just plain don't > > work, or won't compile. > > > > Is there just one out there that is guaranteed to work with adequate > > performance with FC2 or FC3. I don't mind whether its SIP or IAX2 - I > > just need it to _work_. > > > > iaxcomm worked right off the bat for me... FC2 on a MicronPC latop. It is "working" for me too. Im using IaxComm, sometimes it frezes, and I have to kill the proc and start it again... I have another problem too The ring tone is very poor... I dont know why!? Regards. Denis. ___ 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] IAX native transfers
Em Ter 01 Fev 2005 16:27, Bruno Hertz escreveu: > On Tue, 2005-02-01 at 16:27 +, Gareth Blades wrote: > > Unattended transfers just does nothing. I cannot get it to do anything. > > Not sure about this, but I'm under the impression that the # transfer > might need some client support. > > E.g. I tried gnomemeeting -> * -> NAT -> * -> firefly and # did nothing. > But when using sjphone instead of firefly it worked. So my guess is that > when sending the callee to a different extension, the callee's client > must support it. Or it may actually be an IAX problem, as sjphone is SIP > of course. Didn't try another IAX client, so a definitive answer would > interest me as well ... I believe that your problem is related to DTMF problems with your softphones. Denis. ___ 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] IAX Client
Hi Cesar. Try it out: http://iaxclient.sourceforge.net -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br Em Ter 01 Fev 2005 15:22, César Davi Ávila do Nascimento escreveu: > Hi All, > > I'd like to develop an IAX - client. > Does somebody know where can I get the source code for an IAX client? > > Regards > > César ___ 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] iaxComm version 1.0 released
Hi Michael. Any work to support some USB Phones!? The ability to dial using the phones keypad!? Thanks. Denis. Em Sáb 29 Jan 2005 01:11, Michael Van Donselaar escreveu: > iaxComm is an Open Source softphone for the Asterisk PBX. > > iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. > > Recent Changes: > * Improved jitterbuffer code > * Steve Underwood's Packet Loss Concealment Code > > Features Include: > > * iLBC support > * GSM support > * speex support > * ulaw and alaw support > * Blind Transfer. > * Custom Ringtones per CallerID > * Speakerphone mode. > * Register with multiple servers (ie enterprise server and iaxtel). > * Multiple call appearances. > * User selectable audio devices. > * User defined ringtones. > * Autoanswer intercom calls (with password protection). > > http://iaxclient.sourceforge.net/iaxcomm-win-1.0rc1.zip > http://iaxclient.sourceforge.net/iaxcomm-lin-1.0rc1.tar > > > ___ > 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 -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] IAX Softphone
Em Qui 27 Jan 2005 05:18, Dan escreveu: > Hi Denis, > > >- Original Message - > >From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> > > > > > Hey I tried DIAX today and the speech quality was rather poor > > > compared to X-lite. > > > >Dan, do you know wich iaxclient version firefly is build on!? > > > >I got better results(voice quality) using firefly, doesn't matter what > >CODEC > >I used. > > I don't know which library firefly uses. > Can you describe in more detail the difference regarding voice quality? > I mean... more distorted, drop-outs, tone, level, etc...? With Firefly I got better volume and the voice is more "polished", I mean, with DIAX I got more noise. This is my expirience, I tried a lot of softphones in different computers, Firefly win the contest, but I think DIAX is the better of all in features! Like I told you before, I really want to use DIAX! P.S.: Someone forgot to say that DIAX supports USB Phones with /u flag too! For it is great Regards, Denis. ___ 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] IAX Softphone
Em Qua 26 Jan 2005 20:01, Dan escreveu: > Hi, > > > Hey I tried DIAX today and the speech quality was rather poor compared > > to X-lite. > Dan, do you know wich iaxclient version firefly is build on!? I got better results(voice quality) using firefly, doesn't matter what CODEC I used. Regards. -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] AMP with SUSE 9.2
Same for me... No confirmation... Denis. Em Ter 25 Jan 2005 17:38, Keith Burns escreveu: > Ok, I signed up a few hours ago for the AMP mailing list, and no > confirmation. > > If anyone on this list has installed AMP with SUSE 9.2, if you wouldn't > mind emailing me with any gotchas at [EMAIL PROTECTED] I sure > would appreciate it. > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Keith Burns > > Sent: Tuesday, January 25, 2005 9:43 AM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: RE: [Asterisk-Users] AMP with SUSE 9.2 > > > > Cool, will do, thanks! > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > [mailto:asterisk-users- > > > > [EMAIL PROTECTED] On Behalf Of Jason Becker > > > Sent: Tuesday, January 25, 2005 9:24 AM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [Asterisk-Users] AMP with SUSE 9.2 > > > > > > Keith Burns wrote: > > > > *Hi,* > > > > > > > > *I have the newbie guide from AMP**'**s website and (fair enough) > > it > > > is > > > > > > all about whitebox linux.** Has anyone found any gotchas with the > > > > newbie > > > > > > guide relating to SUSE 9.2 ?* > > > > > > Please post to the amportal mailing list: > > > > > > http://lists.sourceforge.net/lists/listinfo/amportal-users > > > > > > or Help forum: > > > > > > http://sourceforge.net/forum/?group_id=121515 > > > > > > SUSE does some things differently - the main difference is the > > apache2 > > > > (httpd) configuration. > > > > > > Regards, > > > > > > -- > > > Jason Becker > > > Director & CEO > > > Coalescent Systems Inc. > > > 403.244.8089 > > > www.coalescentsystems.ca > > > ___ > > > 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 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 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 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] DIAX 0.9.9g more features and higher stability
Em Sáb 22 Jan 2005 07:51, Dan escreveu: > Hi all, > There is someone on this list having latency issues with DIAX who can > do this trace? I'm not able to dupplicate this behaviour here and as I'm > behind > a NAT I cannot use 2 DIAX phones connected to an external Asterisk > server (or there is a workaround for this?). Hi Dan. I could help on it, but I'll be able to get this trace only on wednesday 26... Tks. -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] Tips do update Asterisk and AMP
Sorry about the repost. I got an error in the first one. Denis. Em Qui 20 Jan 2005 14:48, Denis Galvão - iSolve escreveu: > Hi all. > > Somebody knows if AMP will work with the newest version of > asterisk(1.0.3)!? > > Somehting that I need to know before update!? How is the best way to get > my system updated!? > > Thanks. > > Denis. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: ___ 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] Tips do update Asterisk and AMP
Hi all. Somebody knows if AMP will work with the newest version of asterisk(1.0.3)!? Somehting that I need to know before update!? How is the best way to get my system updated!? Thanks. Denis. ___ 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] Tips do update Asterisk and AMP
Hi all. Somebody knows if AMP will work with the newest version of asterisk(1.0.3)!? Somehting that I need to know before update!? How is the best way to get my system updated!? Thanks. Denis. ___ 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] Newbie question: Can't start up asterisk
Em Ter 18 Jan 2005 21:27, beonice escreveu: > That _seems_ to be a possibility. But I'm not really > sure. I made sure that there is a symbolic link in > /usr/bin to mpg123 ... the actual version is in > /usr/local/bin. > > Thanks. By the way, I accidentally created a new post > with the details of the output instead of responding > to Matt's question right here ... but here is the > output again: Did you install mpg123 from source!? Or you're using a distro native version!? You have to get the mpg123 from its website and then get it compiled to your suystem. From AMP manual: SNIP Some linux distros have replaced the mpg123 application with another application, mpg321, and created a symbolic link to "mpg123", so it seems to work in the same way. Asterisk MusicOnHold only works with original mpg123. - Remove the symbolic links mpg123 located in /usr/bin and /usr/local/bin: - rm /usr/bin/mpg123 - rm /usr/local/bin/mpg123 - Then install mpg123 from http://www.mpg123.de SNIP Regards, Denis. ___ 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] Newbie question: Can't start up asterisk
Em Ter 18 Jan 2005 20:43, Matt Riddell escreveu: > beonice wrote: > > "Ouch ... error while writing audio data: : Broken > > pipe" > > What are the messages before this? Matt I think that is something related to mpg123... -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br pgpQ6K8S2agdZ.pgp Description: PGP signature ___ 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] DIAX 0.9.9g more features and higher stability
Em Seg 17 Jan 2005 19:13, Steve Kann escreveu: > I've already replied, asking for a trace.. If you get the trace, and > send it, we can look at what is actually happening: > > > What would really help, though, is a packet trace of the call. The > best way to get this is to use either ethereal or tcpdump. (there is an > ethereal for windows). > If you use ethereal for Windows, have it capture all udp, make the call, > and have it stay up for about 30 seconds, and save the file. You can > then send that file to me, and I'll be able to see what's going on a lot > better than guessing here.. > Hi Steve. I will do it, but I cant today. How could you get some info with a call trace from ethereal!? You will have a lot of traffic between 4569 UDP(IAX2) from both sides, how could you have a diagnostic of the problem!? Thanks and best regards. -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] DIAX 0.9.9g more features and higher stability
Em Seg 17 Jan 2005 16:47, Dan escreveu: > Hi Denis, > > >Same problem with version 0.9.8c > >After one minute aprox, delay disappear. > >Any ideas!? > > What's very strange is that I cannot reproduce this behaviour, trying > with different PCs and different DIAX versions and settings. > Which Asterisk version do you have installed? > I have now and it works in any circumstances the following: > CVS-D2004.09.20.21.00.00-11/11/04-16:24:55 > > It can be related to this? Thats the Asterisk version that I'm playing with: Asterisk CVS-v1-0-11/04/04-23:47:17 built by [EMAIL PROTECTED] on a i686 running Linux I dont know if it could be a problem... Anyone!? Someone that have the same version of mine could test it with two DIAX!? Tks. Denis ___ 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] Any interest in a Canadian Asterisk mailing list?
Here in Brazil we are creating our language email list. For us it will be great because we have a lot of different terms, technologies of US and Canada, even Europe. I think that for brazillians, the first place to solve a problem could be the brazillian list, after that a universal list(this asterisk-users list) could be the second choice. The local lists will be a good point to share local expiriences. I dont think that Canada and US have a huge difference of telecom standards like Brazil and US... Denis. Em Seg 17 Jan 2005 15:23, Jim Van Meggelen escreveu: > Gyrion, Larry M. wrote: > > I believe the US and Canada use the same methods for voice > > services, maybe we could make it a North America list serv > > instead. Just some thoughts here > > I see the value of regional lists primarily as relates to non-technical > items such as local service providers, regulatory issues, and so forth. > > For the technical stuff, I much prefer the single list. I don't mind at > all reading about POTS lines in the UK, or the joys of ISDN in the EU. > Maybe one day the sheer volume of messages will make that untenable, but > for now I love the international flavour of the list. > > As VoIP becomes ubiquitous, the value of regional lists might be less > and less relevant. > > Jim. > > > -Original Message- > > From: Jim Van Meggelen [mailto:[EMAIL PROTECTED] > > Sent: Monday, January 17, 2005 11:42 AM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: RE: [Asterisk-Users] Any interest in a Canadian > > Asterisk mailing list? > > > > [EMAIL PROTECTED] wrote: > >> On January 17, 2005 01:47 am, John Sellens wrote: > >>> Just on the off chance that Canadian Asterisk users might be > >>> interested in a place to discuss topics specific to the "great white > >>> north" (sources, services, telcos, etc.), I created the > >>> asterisk-canada mailing list: > >> > >> I know as a Canadian I'm not interested in a list "Just for > >> Canadians" -- It's just fragmenting the help available for very > >> little benefit. I do, however, appreciate the thought. > > > > I don't think the idea is to be "just for Canadians", but > > more as a forum for topics that relate to Asterisk in the > > Canadian environment. > > > > A very relevant example is the CRTCs deliberations on VoIP, > > which may have huge repercussions to Canadian Asterisk users, > > but is hardly relevant to the international version of the > > Asterisk-Users list. Bell and TELUS bashing might also be > > popular topics :-) > > > > I do agree that any subject that is not specific to the > > Canadian experience should remain in the international list. > > We are an international community; therein lies our power. > > > > Anyhow, I signed up, and am planning to start a thread about > > the CRTC VoIP deliberations (and the generous act performed > > there by Jeff Pulver), something I wouldn't feel was > > appropriate on Asterisk-Users. Time will tell how many topics > > there are to discuss. > > > > The way I see it, a Canadian mailing list will be no > > different than our country itself: visitors will always be welcome. -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] DIAX 0.9.9g more features and higher stability
Exactly the same problem for all of them(097a, 096d and 095). The delay is getting down by the time of conversation. After aprox 1 minute, or even less, the delay is totaly off. Denis. Em Seg 17 Jan 2005 15:59, Dan escreveu: > Hi, > > >From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> > > > >Same problem with version 0.9.8c > > > >After one minute aprox, delay disappear. > > > >Any ideas!? > > Can you check the older ones too? > > > http://www.laser.com/dante/diax/diax097a.zip > > http://www.laser.com/dante/diax/diax096d.zip > > http://www.laser.com/dante/diax/diax095.zip > > > > and see if the problem persist. > > Thank you and best regards, > Dan ___ 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] DIAX 0.9.9g more features and higher stability
I cant download from this location: Connection Refused wget http://www.laser.com/dante/diax/diax095.zip --16:02:08-- http://www.laser.com/dante/diax/diax095.zip => `diax095.zip' Resolving www.laser.com... 216.167.90.224 Connecting to www.laser.com[216.167.90.224]:80... failed: Conexão recusada. Denis. Em Seg 17 Jan 2005 15:59, Dan escreveu: > Hi, > > >From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> > > > >Same problem with version 0.9.8c > > > >After one minute aprox, delay disappear. > > > >Any ideas!? > > Can you check the older ones too? > > > http://www.laser.com/dante/diax/diax097a.zip > > http://www.laser.com/dante/diax/diax096d.zip > > http://www.laser.com/dante/diax/diax095.zip > > > > and see if the problem persist. > > Thank you and best regards, > Dan -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] DIAX 0.9.9g more features and higher stability
Hi Dan. Same problem with version 0.9.8c After one minute aprox, delay disappear. Any ideas!? -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br Em Seg 17 Jan 2005 13:43, Dan escreveu: > Hi Denis, > > >From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> > >... > > - Same problem with DIAX oldest DLL; > > It is not an old DLL, but the same DLL build with NEW JITERBUFFER 0 > > Please try an older version of DIAX, like 0.9.8c. > You can still download it from: > > http://www.laser.com/dante/diax/diax098c.zip > or even older: > http://www.laser.com/dante/diax/diax097a.zip > http://www.laser.com/dante/diax/diax096d.zip > http://www.laser.com/dante/diax/diax095.zip > > and see if the problem persist. > > If not, then it must be something in the new library and we will dig > further. > > Thank you and best regards, > Dan > P.S. Pls tell me the version working without delay... ___ 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] simple over view of the process
Digium is the company behind the Hardware to Asterisk. Try its website: http://www.digium.com They have a developers kit that could reach your needs. Denis. Em Seg 17 Jan 2005 14:13, [EMAIL PROTECTED] escreveu: > Hello All, > > Please forgive the lack of understanding as of yet but I have been trying > to follow the mailing list messages over the last few days and would like > to know if someone could wither point me into the right direction or > possibly give me a brief overview of the complete process. > > Basically, I see that the Asterisk PBX systems can run on linux and seems > to offer the engine base that is needed for the SIP clients to connect. > > Additionally, it seems that the various hardware (of which I have no > idea) if installed into the server will allow the SIP clients to > communicate with analog lines. > > What inexpensive hardware is need to set up a basic system? > > Thanks, > -Lonnie > > ___ > 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 -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] DIAX 0.9.9g more features and higher stability
Em Seg 17 Jan 2005 13:43, Dan escreveu: > Hi Denis, > > >From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> > >... > > - Same problem with DIAX oldest DLL; > > It is not an old DLL, but the same DLL build with NEW JITERBUFFER 0 > > Please try an older version of DIAX, like 0.9.8c. > You can still download it from: > > http://www.laser.com/dante/diax/diax098c.zip > or even older: > http://www.laser.com/dante/diax/diax097a.zip > http://www.laser.com/dante/diax/diax096d.zip > http://www.laser.com/dante/diax/diax095.zip > > and see if the problem persist. > > If not, then it must be something in the new library and we will dig > further. > > Thank you and best regards, > Dan > P.S. Pls tell me the version working without delay... Ok, Dan, I will try it out, and I'll inform you the results. Denis. ___ 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] DIAX 0.9.9g more features and higher stability
Em Seg 17 Jan 2005 13:51, Steve Kann escreveu: > Yes, it sounds like there's a discontinuity in the timestamps when you > set up your call, but it seems Dan can't reproduce this. > > The fix is probably: > > a) The jitterbuffer needs to be reset after the transfer, or > b) The timestamps sent need to be reset after the transfer. > c) Some changes to the jitterbuffer to automatically reset when it sees > this kind of discontinuity. > > (c can probably be combined with a and/or b). > > I forget if you tried setting "notransfer=yes" on asterisk to see what > that does? Yes, Im using notransfer=yes, like my iax extension: [1001] callerid="Ramal 1001" <1001> context=from-internal host=dynamic mailbox=1001 notransfer=yes port=4569 secret=1001 type=friend username=1001 > > What would really help, though, is a packet trace of the call. The > best way to get this is to use either ethereal or tcpdump. (there is an > ethereal for windows). > If you use ethereal for Windows, have it capture all udp, make the call, > and have it stay up for about 30 seconds, and save the file. You can > then send that file to me, and I'll be able to see what's going on a lot > better than guessing here.. Ok, I will do it. Thanks Steve. Denis. ___ 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] DIAX 0.9.9g more features and higher stability
Two more information: 1. I've played with all suported codecs, same problems for all of them. 2. After aprox. 1 minute of conversation the delay problem doesn't occur, or better, it is very less(some miliseconds) than the begining(10 seconds) of a call. Any ideas!? Denis. Em Seg 17 Jan 2005 11:51, Denis Galvão - iSolve escreveu: > Hi Dan, Steve, Michael, Bruno and others. > > I will try to describe my VoIP environment below: > > SERVER: > - FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17 > - iax.conf > [general] > bindport = 4569 > bindaddr = 0.0.0.0 > delayreject=yes > disallow=all > allow=ulaw > allow=alaw > allow=gsm > tos=lowdelay > jitterbuffer=no > dropcount=2 > maxjitterbuffer=100 > maxexccessbuffer=100 > mailboxdetail=yes > > [1001] > callerid="Ramal 1001" <1001> > context=from-internal > host=dynamic > mailbox=1001 > notransfer=yes > port=4569 > secret= > type=friend > username=1001 > > [1002] > callerid="Ramal 1002" <1002> > context=from-internal > host=dynamic > mailbox=1002 > notransfer=yes > port=4569 > secret= > type=friend > username=1002 > > CLIENT 1001: > - Windows XP > - DIAX 0.9.9g > - Firefly 1.9.6 Build 3944 > - USB Phone NTP200E - Compatible with ATCOM USB Phone > - AMD 1.8Ghz with 256Mb > > CLIENT 1002: > - Windows XP > - DIAX 0.9.9g > - Firefly 1.9.6 Build 3944 > - USB Phone NTP200E - Compatible with ATCOM USB Phone > - AMD 1.66Ghz with 256Mb > > > ADDITIONAL INFORMATION > - All machines are in the same network(192.168.*.*) no firewall in the > middle; > - With Firefly I have a VERY GOOD conversation, without any delay; > - With DIAX I have a one way delay of 10 sec. Only the person who recieve > the call get the delay, the person who make the call listen without > problems; > - Firefly in one side and DIAX in the other side, same delay problem; > - No problems with SIP; > - No problems(delay) with Linux clients runnig IaxComm 0.99pre11; > - Same problem with DIAX oldest DLL; > - Ping from clients to server: 0% packet loss and < 1ms; > - No problems calling PSTN, Voicemail, etc, just between DIAX clients; > > If you need something else, let me know! > > Thanks for your help! > > Denis Galvão. > > Em Dom 16 Jan 2005 19:52, Steve Kann escreveu: > > On Jan 16, 2005, at 2:53 PM, Dan wrote: > > > Hi Steve, > > > > > > - Original Message - From: "Steve Kann" <[EMAIL PROTECTED]> > > > > > >> On Jan 14, 2005, at 2:03 PM, Dan wrote: > > >>> Hi, > > >>> > > >>> \> Em Sex 14 Jan 2005 16:43, Dan escreveu: > > >>>>> > I dont have problems when calling PSTN extensions, and calling > > >>>>> > VoceMail, EchoTest, etc. The problem is related with the > > >>>>> > > >>>>> conversation > > >>>>> > > >>>>> > between two DIAX Softphones. > > >>>>> > > >>>>> Between 2 DIAX phone and the delay is in one direction only?? > > >>>> > > >>>> Yes. One direction only... Just who make the call get the delay. > > >>> > > >>> Then try > > >>> jitterbuffer=no > > >>> in iax.conf > > >>> to see if it solves this issue. > > >> > > >> Dan et. al, > > >> I think this might be a problem with native transfers, and needing > > >> to reset the jitterbuffer history when this happens, or something > > >> like this.. > > >> -SteveK > > > > > > But I have tried and I do don't have this problem here... > > > What can I do to make this happen here? > > > > I don't know... > > > > Maybe if we could get a packet trace of the situation that causes the > > problem? > > > > Maybe try notransfer or whatever the iax.conf parameter is, and see if > > that changes things. If it does, it points towards this being the > > problem. > > > > If the delay goes down after a couple of minutes after the transfer, > > this could be the problem. If it doesn't, there's something else > > really wrong.. > > > > (I'm assuming you're using the new JB code here..). Also, if you're > > using the new JB code, you should implement the stuff to get the > > network stats, so we can see if calculated jitter is substantially > > higher..) > > > > ___ > > 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 -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] DIAX 0.9.9g more features and higher stability
Hi Dan, Steve, Michael, Bruno and others. I will try to describe my VoIP environment below: SERVER: - FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17 - iax.conf [general] bindport = 4569 bindaddr = 0.0.0.0 delayreject=yes disallow=all allow=ulaw allow=alaw allow=gsm tos=lowdelay jitterbuffer=no dropcount=2 maxjitterbuffer=100 maxexccessbuffer=100 mailboxdetail=yes [1001] callerid="Ramal 1001" <1001> context=from-internal host=dynamic mailbox=1001 notransfer=yes port=4569 secret= type=friend username=1001 [1002] callerid="Ramal 1002" <1002> context=from-internal host=dynamic mailbox=1002 notransfer=yes port=4569 secret= type=friend username=1002 CLIENT 1001: - Windows XP - DIAX 0.9.9g - Firefly 1.9.6 Build 3944 - USB Phone NTP200E - Compatible with ATCOM USB Phone - AMD 1.8Ghz with 256Mb CLIENT 1002: - Windows XP - DIAX 0.9.9g - Firefly 1.9.6 Build 3944 - USB Phone NTP200E - Compatible with ATCOM USB Phone - AMD 1.66Ghz with 256Mb ADDITIONAL INFORMATION - All machines are in the same network(192.168.*.*) no firewall in the middle; - With Firefly I have a VERY GOOD conversation, without any delay; - With DIAX I have a one way delay of 10 sec. Only the person who recieve the call get the delay, the person who make the call listen without problems; - Firefly in one side and DIAX in the other side, same delay problem; - No problems with SIP; - No problems(delay) with Linux clients runnig IaxComm 0.99pre11; - Same problem with DIAX oldest DLL; - Ping from clients to server: 0% packet loss and < 1ms; - No problems calling PSTN, Voicemail, etc, just between DIAX clients; If you need something else, let me know! Thanks for your help! Denis Galvão. Em Dom 16 Jan 2005 19:52, Steve Kann escreveu: > On Jan 16, 2005, at 2:53 PM, Dan wrote: > > Hi Steve, > > > > - Original Message - From: "Steve Kann" <[EMAIL PROTECTED]> > > > >> On Jan 14, 2005, at 2:03 PM, Dan wrote: > >>> Hi, > >>> > >>> \> Em Sex 14 Jan 2005 16:43, Dan escreveu: > >>>>> > I dont have problems when calling PSTN extensions, and calling > >>>>> > VoceMail, EchoTest, etc. The problem is related with the > >>>>> > >>>>> conversation > >>>>> > >>>>> > between two DIAX Softphones. > >>>>> > >>>>> Between 2 DIAX phone and the delay is in one direction only?? > >>>> > >>>> Yes. One direction only... Just who make the call get the delay. > >>> > >>> Then try > >>> jitterbuffer=no > >>> in iax.conf > >>> to see if it solves this issue. > >> > >> Dan et. al, > >> I think this might be a problem with native transfers, and needing to > >> reset the jitterbuffer history when this happens, or something like > >> this.. > >> -SteveK > > > > But I have tried and I do don't have this problem here... > > What can I do to make this happen here? > > I don't know... > > Maybe if we could get a packet trace of the situation that causes the > problem? > > Maybe try notransfer or whatever the iax.conf parameter is, and see if > that changes things. If it does, it points towards this being the > problem. > > If the delay goes down after a couple of minutes after the transfer, > this could be the problem. If it doesn't, there's something else > really wrong.. > > (I'm assuming you're using the new JB code here..). Also, if you're > using the new JB code, you should implement the stuff to get the > network stats, so we can see if calculated jitter is substantially > higher..) > > ___ > 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 -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] DIAX 0.9.9g more features and higher stability
Em Sex 14 Jan 2005 18:03, Bruno Hertz escreveu: > On Fri, 2005-01-14 at 16:27 -0200, Denis GalvÃo - iSolve wrote: > > Em Sex 14 Jan 2005 16:11, Dan escreveu: > > > > I dont have problems when calling PSTN extensions, and calling > > VoceMail, EchoTest, etc. The problem is related with the conversation > > between two DIAX Softphones. > > With * in the middle or direct calls? I had problems with > iaxcomm -> * -> firefly communication when * attempted a transfer. > Huge latencies (10 sec or so). Might be a bug in the iaxclient library, > just don't know. Asterisk in the middle. Try out Firefly -> * -> Firefly, for me it worked out. I really want to use DIAX, Im working to solve(detect the problem) this issue. Any help will be great. Denis. ___ 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] DIAX 0.9.9g more features and higher stability
I tried IaxComm in two Linux boxes. Everything work fine, with USB Phones + IaxComm. So, the problem should be related to Windows OS!? Wich version of Windows are you using Dan!? Denis. Em Sex 14 Jan 2005 17:16, Denis Galvão - iSolve escreveu: > Same problem with jitterbuffer=no > > I tried IaxComm, same problem of DIAX. > > This is related with iaxclient... > > Denis. > > Em Sex 14 Jan 2005 17:03, Dan escreveu: > > Hi, > > > > \> Em Sex 14 Jan 2005 16:43, Dan escreveu: > > >> > I dont have problems when calling PSTN extensions, and calling > > >> > VoceMail, EchoTest, etc. The problem is related with the > > >> > conversation between two DIAX Softphones. > > >> > > >> Between 2 DIAX phone and the delay is in one direction only?? > > > > > > Yes. One direction only... Just who make the call get the delay. > > > > Then try > > jitterbuffer=no > > in iax.conf > > to see if it solves this issue. > > > > BR, > > Dan > > ___ > 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 -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] DIAX 0.9.9g more features and higher stability
Same problem with jitterbuffer=no I tried IaxComm, same problem of DIAX. This is related with iaxclient... Denis. Em Sex 14 Jan 2005 17:03, Dan escreveu: > Hi, > > \> Em Sex 14 Jan 2005 16:43, Dan escreveu: > >> > I dont have problems when calling PSTN extensions, and calling > >> > VoceMail, EchoTest, etc. The problem is related with the > >> > conversation between two DIAX Softphones. > >> > >> Between 2 DIAX phone and the delay is in one direction only?? > > > > Yes. One direction only... Just who make the call get the delay. > > Then try > jitterbuffer=no > in iax.conf > to see if it solves this issue. > > BR, > Dan ___ 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] DIAX 0.9.9g more features and higher stability
Em Sex 14 Jan 2005 16:43, Dan escreveu: > > I dont have problems when calling PSTN extensions, and calling > > VoceMail, EchoTest, etc. The problem is related with the conversation > > between two DIAX Softphones. > > Between 2 DIAX phone and the delay is in one direction only?? Yes. One direction only... Just who make the call get the delay. > The phones are connected to the same PBX? Yes they are in the same Asterisk. > The problem is the same independent of the codec used? Yes. I tried out all of the codecs available. Driving me nuts... Denis. ___ 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] DIAX 0.9.9g more features and higher stability
Em Sex 14 Jan 2005 16:11, Dan escreveu: > I have modified the CallMe feature for DIAX to provide an Echo test. > Just use it with 0.9.9g and see the result. To pass the explanation or to > end the echo test just press '#'. You can still leave me a message after > that. I got the echo test. The result was fine, just a very SHORT delay, but nothing like my problem. I dont have problems when calling PSTN extensions, and calling VoceMail, EchoTest, etc. The problem is related with the conversation between two DIAX Softphones. Thanks for your help. Denis. ___ 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] DIAX 0.9.9g more features and higher stability
Em Sex 14 Jan 2005 15:11, Michael Van Donselaar escreveu: > The iaxclient default latency for windows was changed about two months > ago to 40. > > There were a couple of reports of audio distortion, so it was kicked up > to 67. > > I think you can get pretty agressive with this, just remember to check on > the latency if you get distortion. > I think the problem is not related to latency. I tried from 20 to 200 latency time, but the problem is the same: When: Jon -> call -> Fred Fred listen Jon without problems, but Jon listen Fred with 10 seconds of delay. When: Fred -> call -> Jon Jon listen Fred wihtout problems, but Fred listen Jon with 10 seconds of delay With Firefly Softphone(IAX2) I dont get this problem, everything works great. Thanks for any help. Denis. ___ 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] DIAX 0.9.9g more features and higher stability
Em Sex 14 Jan 2005 11:18, Dan escreveu: > Hi, > > >- Original Message - > >From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]> > > > >I have the same delay problem with diax 0.9.9g. > > > >This problem just happen with DIAX softphone, > > with others(iaxcomm, firefly, > > etc.) doesn't occur. > > > > Im using an ATCOM compatible USB Phone. > > > >Is there anything that I can do to solve this issue!? > > Have you trioed to play with the 'Latency' parameter > in Audio Configuration form? > Try between 40 and 200. > IAXCOM I think use the default which is 200. Doesn't have effect, the problem continue. The strange thing is that the delay is aprox 10-20 seconds!!! Too high!! Is there another thing to do!? I really want to use DIAX because it supprts my USB Phone and IAX protocol. Thanks. Denis. ___ 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] High delay with diax099f + Asterisk
Hi Matt. Same problem with 0.9.9g... Thanks. Denis. Em Sex 14 Jan 2005 03:17, Matt Riddell escreveu: > Denis Galvão - iSolve wrote: > > Hi all! > > > > Somebody knows something to do with a high delay using Asterisk + > > DIAX!? > > Try grabbing 099g released today... > > QUOTE: DIAX 0.9.9g is available for download (including the updated help > file and web page) from the following locations: > > http://www.laser.com/dante > > or > > http://www.geocities.com/tdanro ___ 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] DIAX 0.9.9g more features and higher stability
Hi Dan. I have the same delay problem with diax 0.9.9g. This problem just happen with DIAX softphone, with others(iaxcomm, firefly, etc.) doesn't occur. Im using an ATCOM compatible USB Phone. Is there anything that I can do to solve this issue!? Thanks in advance. Denis. Em Sex 14 Jan 2005 04:05, Dan escreveu: > Hi all, > > DIAX 0.9.9g is available for download (including the updated help file > and web page) from the following locations: > http://www.laser.com/dante > or > http://www.geocities.com/tdanro > > What's new in 0.9.9g (from 0.9.9f): > > - during a call, accept DTMF tones as monitored events to trigger output > commands > - call timer on the phone display > - Swedish language added > - can run a command from the monitoring definition form, to test it > - ENTER key validate all fields in the Registration form > - you can select both preffered and accepted codecs > - do not autoresize main form when receiving a call and monitoring > activated - use /m switch to start DIAX minimized > - saving only main form position, all others auto positioning relative to > the main form > > solved bugs: > - crash when trying to dial without registration server defined > - Config Audio form positioning issue > - not saving the main form when closing the app from the systray > - X10 send error if CM11/12 interface has some commands in the receiver > buffer > - error if trying to delete for the second time the log file > - unexpected crashes when registered with IAXTEL and/or other remote > servers > > > As usual, please send me your feedback. > > > Best regards, > Dan > > > ___ > 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 -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] long delays in list posts?
Em Qui 13 Jan 2005 21:06, Steven Critchfield escreveu: > On Thu, 2005-01-13 at 16:41 -0600, Matthew Boehm wrote: > > OMG! 1 hour?!?! I just now got this at 4:40PM. It takes an hour for my > > emails to get posted to the list? Geez.. > > If you need responses in faster time than 1 hour you need to familiarize > your self with consultants in your area and sign some form of support > contract with them. Otherwise, chill out. Drink a beer or 3. Then try > and remember that this is a free support forum even though it costs > Digium and I think 2 other companies money to run this list. Your replies are too funny... We do not need any beer to get high with Steven kind of posts!!! Best Regards Dude! Denis. ___ 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] High delay with diax099f + Asterisk
Hi all! Somebody knows something to do with a high delay using Asterisk + DIAX!? When I used IAXComm(Linux) in both sides(peer and me) no problems. Whan I used DIAX099f(WinXP) in both sides(peer and me) I have a delay in the voice coming from the person that I called. I don't have delay in my voice to the peer phone. CODEC: u-law (I tried with all available codecs) Thanks for your help! -- Denis Galvão ___ 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] I need your feedback related to the DIAX 0.9.9f stability
Dan, I'm using DIAX(Portuguese Language) for one week too... Without any problems. I've tested with all suported CODECs... But Im using with a-law and u-law for now. If you need some help to translate to Brazillian Portuguese, call me! I like the incoming calls ring... ;) Denis. Em Seg 10 Jan 2005 05:46, Dan escreveu: > Hi all, > > I kindly ask DIAX users to send me a feedback related to the stability of > the new version (0.9.9f), > comparing with the older versions (especially 0.9.8). > I ask this because I have DIAX runing for one week now without any crash. > It is used mainly to control some X10 devices through a regular phone. > > Thank you and best regards, > Dan > > > ___ > 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 -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] New verision of AMP - 1.10.004
Hi Jason. First of all, thanks to become AMP a GPL software! What about zap channels support in AMP!? Denis. Em Qui 23 Dez 2004 14:08, Jason Becker escreveu: > Kanuri, Seshu (Company IT) wrote: > > Has this version improved the install process from what it was, to > > something where a guy with average intelligence (AKA dummy) can install > > without the need of a consultant. > > The Installation Guide provides step-by-step instructions: > > http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.1.pdf > > It might be better to characterize AMP as a blueprint or set of > instructions for building your own *-based telephony server. You do need > to invest some time and effort to build it - or you can pay someone else > to do it for you. > > We do of course welcome efforts from the community to make the project > better. An installation routine that handles all the popular Linux > distributions is non-trivial. At present we are focused on providing the > user base oft-requested features - like support for VoIP trunks which we > delivered in this version. > > Regards, -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ 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] Integrating Asterisk and Siemens Hicom 300E with TDM04B
Hi all. Does anyone integrated a Siemens Hicom 300E with Asterisk using FXO interfaces!? I created an extension group in Hicom and connected my 4FXO(TDM04B) into the telefony internal network. What issues I have to care about it!? Thanks for any help! Regards, Denis. ___ 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] Email to Fax?
Em Ter 07 Dez 2004 00:37, Denis Galvão escreveu: > Em Seg 06 Dez 2004 09:44, Michael Nolan escreveu: > > On Sat, 4 Dec 2004 21:24:05 -0500, Jason Lixfeld > > > > <[EMAIL PROTECTED]> wrote: > > > I've read about Fax to Email, but is there such a beast as email to > > > fax? If not, what do people use to take care of outbound faxing? > > > > HylaFAX can be made to do email to fax in a number of different ways. > > It can be a pain so setup, but once it's running, it should be > > reliable. > > Think the same, once runnig, forget it! > > Just install your faxserver like a printer and you will be able to send > emails with every program that could print... Sorry. "...send faxes" and not "...send emails" Denis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Email to Fax?
Em Seg 06 Dez 2004 09:44, Michael Nolan escreveu: > On Sat, 4 Dec 2004 21:24:05 -0500, Jason Lixfeld > > <[EMAIL PROTECTED]> wrote: > > I've read about Fax to Email, but is there such a beast as email to > > fax? If not, what do people use to take care of outbound faxing? > > HylaFAX can be made to do email to fax in a number of different ways. > It can be a pain so setup, but once it's running, it should be > reliable. Think the same, once runnig, forget it! Just install your faxserver like a printer and you will be able to send emails with every program that could print... Denis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] full duplex sound card
Em Seg 06 Dez 2004 01:49, [EMAIL PROTECTED] escreveu: > Hello, > I have installed asterisk on fedora core 2. > Can anybody suggest me a good full duplex sound card > supported on linux. Check this out! http://www.alsa-project.org/alsa-doc/ Some Googling will help you too ;) Denis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TE110P + Asterisk
Just remember you, that you can have Digium support too! Regards, Denis. Em Qui 02 Dez 2004 11:10, Leonardo Tramontina escreveu: > Hi, > > I've just got a TE110P card and installed at Asterisk. > I configured zapata.conf, according to > www.digium.com/index.php?menu=configuration, but the following error is > happening: > > > ... > ... > ... > [chan_phone.so] => (Linux Telephony API Support) > == Parsing '/etc/asterisk/phone.conf': Found > == Registered channel type 'Phone' (Standard Linux Telephony API > Driver) [chan_features.so] => (Feature Proxy Channel) > == Registered channel type 'Feature' (Feature Proxy Channel Driver) > [chan_zap.so] => (Zapata Telephony w/PRI) > == Parsing '/etc/asterisk/zapata.conf': Found > -- Registered channel 1, PRI Signalling signalling > -- Registered channel 2, PRI Signalling signalling > -- Registered channel 3, PRI Signalling signalling > -- Registered channel 4, PRI Signalling signalling > -- Registered channel 5, PRI Signalling signalling > -- Registered channel 6, PRI Signalling signalling > -- Registered channel 7, PRI Signalling signalling > -- Registered channel 8, PRI Signalling signalling > -- Registered channel 9, PRI Signalling signalling > -- Registered channel 10, PRI Signalling signalling > -- Registered channel 11, PRI Signalling signalling > -- Registered channel 12, PRI Signalling signalling > -- Registered channel 13, PRI Signalling signalling > -- Registered channel 14, PRI Signalling signalling > -- Registered channel 15, PRI Signalling signalling > -- Registered channel 17, PRI Signalling signalling > -- Registered channel 18, PRI Signalling signalling > -- Registered channel 19, PRI Signalling signalling > -- Registered channel 20, PRI Signalling signalling > -- Registered channel 21, PRI Signalling signalling > -- Registered channel 22, PRI Signalling signalling > -- Registered channel 23, PRI Signalling signalling > Dec 2 11:07:15 ERROR[5209]: chan_zap.c:6447 mkintf: Channel 24 is > reserved for D-channel. > Dec 2 11:07:15 ERROR[5209]: chan_zap.c:9274 setup_zap: Unable to > register channel '1-15' > Dec 2 11:07:15 WARNING[5209]: loader.c:396 ast_load_resource: > chan_zap.so: load_module failed, returning -1 > == Unregistered channel type 'Zap' > -- Unregistered channel 1 > -- Unregistered channel 2 > -- Unregistered channel 3 > -- Unregistered channel 4 > -- Unregistered channel 5 > -- Unregistered channel 6 > -- Unregistered channel 7 > -- Unregistered channel 8 > -- Unregistered channel 9 > -- Unregistered channel 10 > -- Unregistered channel 11 > -- Unregistered channel 12 > -- Unregistered channel 13 > -- Unregistered channel 14 > -- Unregistered channel 15 > -- Unregistered channel 16 > -- Unregistered channel 17 > -- Unregistered channel 18 > -- Unregistered channel 19 > -- Unregistered channel 20 > -- Unregistered channel 21 > -- Unregistered channel 22 > Dec 2 11:07:15 WARNING[5209]: loader.c:500 load_modules: Loading module > chan_zap.so failed! > > > > > > > > > I've tried many combinations on the channel parameter: > channel = 1-15,17-31 > channel => 1-15,17-31 > channel = 1-15;17-31 > channel => 1-15;17-31 > channel = 1-15 > channel = 17-31 > channel => 1-15 > channel => 17-31 > > but it still is not working... > Could someone help me? > > > Thanks > Leonardo > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] asterisk based bbs
Im just trying to don't freeze the future... We have to think about it. If
you think that a ADSL is too slow to put voice on... it happens today...
but in the future!?
Nobody knows, neither me, what will happen, but we must have some
brainstorming about newer models.
Denis.
Em Qua 01 Dez 2004 02:59, Joe Greco escreveu:
> > You're right! But I wrote that voice will be the next content that we
> > will use in networks environments...
>
> Um, voice was pretty much the first content used in networked
> environments (telegraph doesn't count because it wasn't generally
> "networked", at least in an automated manner). Even today, most of the
> Internet runs over data circuits originally envisioned as carrying
> digital voice traffic.
>
> > of course, the model of exchanging information will upgrade too!
> >
> > You will make a GOOGLE search, just talking with your "voice browser",
> > something like:
> > "- Asterisk _plus_ BBS"
> > "- We found 30 references for Asterisk and BBS..."
> >
> > You will have your hands free to write an old email to your friens at
> > the same time that you ask and listen for a file stored in your
> > server...
> >
> > We gonna have another interface to the information, that means: more
> > information per second(i/s).
> >
> > BBS is an old idea, we must update its concepts.
>
> You're talking about a voice based PDA, not a voice based BBS. BBS is an
> old idea, and it's better to not morph its concepts to mean something
> completely different than what it has historically meant, when more
> modern concepts exist that fit much better.
>
> That all said: There's nothing wrong with that idea. I'll note that
> services like "inphone" currently accomplish some basic features along
> this line via a human operator interface; the natural evolutionary
> direction for this is to be a more virtualized voice PDA service of some
> sort like what you're describing.
>
> Regardless, while it may be handy in some circumstances, it doesn't
> really translate to more information per second. Lots of people have
> cable modems and a phone line; I find very few of them running a modem on
> the phone line in order to increase their overall transfer speed to the
> Internet. The trivial bit of added speed usually isn't of value. The
> speed differential between cable and modem is roughly similar to the
> speed differential between eye and ear, and then there's the notable bit
> that many people don't efficiently {read,type} /and/ {listen,speak}
> simultaneously anyways. Humans are not naturally capable of concentrating
> on two things and doing so proficiently.
>
> I don't think that you're actually going to find people using a voice PDA
> to do Google searches while writing e-mail on their computer... it's
> easier and faster to simply open another browser tab and go to Google,
> and then flip back to e-mail.
>
> Focus on when it'd be /really/ useful and usable: when you don't have
> instant Internet access at hand, but you do have voice communications
> (I'll include the visually impaired as a class of people who don't
> necessarily have instant Internet access at hand, at least not in the
> same way most other people do)
>
> ... JG
--
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
+55 41 252-2977
http://www.isolve.com.br
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Re: [Asterisk-Users] asterisk based bbs
Em Ter 30 Nov 2004 20:07, Joe Greco escreveu:
> > Em Dom 28 Nov 2004 13:55, Michael Vogel escreveu:
> > > lenz schrieb:
> > > > I was wondering: anybody ever wrote an asterisk based bbs? not a
> > > > bbs about asterisk, but a vocal bbs that runs on asterisk, so that
> > > > people can call, hear the discussion of the day, leave messages,
> > > > etc.
> > >
> > > It doesn't really make sense to me. It only makes sense for some very
> > > limited fields. e.g. when somebody cannot read or write (he hasn't
> > > learned it or is blind). Everybody else could use the computer. Most
> > > people who would use such a system do have a computer at home I
> > > guess.
> >
> > A lot of people thought the same before WWW, when text based contents
> > were the basis of networks.
>
> The real problem with bulletin board systems (speaking as someone who
> authored bulletin board systems back in the days of 110 and 300 baud) is
> bandwidth. Information transfers slowly at low bit rates, and we
> optimized the hell out of the online experience, doing things like
> shortening the message headers ("To:", "Fr:", "Su:", etc), making sure
> that character inputs to move on to the next message could be entered at
> any time, using short menus but always leaving more detailed help options
> available, etc.
>
> A vocal BBS would inherently involve a very slow transfer of information,
> and there's just so much time people are willing to commit to
> participating.
You're right! But I wrote that voice will be the next content that we will
use in networks environments... of course, the model of exchanging
information will upgrade too!
You will make a GOOGLE search, just talking with your "voice browser",
something like:
"- Asterisk _plus_ BBS"
"- We found 30 references for Asterisk and BBS..."
You will have your hands free to write an old email to your friens at the
same time that you ask and listen for a file stored in your server...
We gonna have another interface to the information, that means: more
information per second(i/s).
BBS is an old idea, we must update its concepts.
Dont try to put a V12 Ferrari engine in your 41 Ford T.
Denis.
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Re: [Asterisk-Users] IAXy power source from Radio Shack
Em Seg 29 Nov 2004 17:48, Wilson Pickett escreveu: > Anyone using this 110-220v 9v 1500ma supply? If so, which DC plug > adapter does the IAXy need? A friend of mine toild me: Easy: It is the type "L" plug, set to a positive tip. Denis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Problems starting Asterisk with TDM22B
1. Try to run: # ztcfg -v What happened!? 2. Did you plug the TDM at the power supply!? Denis. Em Ter 30 Nov 2004 00:40, DB escreveu: > Im getting the following error when I try to start Asterisk > > WARNING[16384]: chan_zap.c:765 zt_open: Unable to specify channel 1: No > such device or address > Nov 29 21:40:43 ERROR[16384]: chan_zap.c:6195 mkintf: Unable to open > channel 1: No such device or address > > I'm using a TDM22B card. > > = > # Zaptel Configuration File > # > loadzone=us > defaultzone=us > fxoks=1-2 > fxsks=3-4 > == > > Do I need to supply more config files or does some one recognize this > error? Please help. > > DB > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] asterisk based bbs
Em Dom 28 Nov 2004 13:55, Michael Vogel escreveu: > lenz schrieb: > > I was wondering: anybody ever wrote an asterisk based bbs? not a bbs > > about asterisk, but a vocal bbs that runs on asterisk, so that people > > can call, hear the discussion of the day, leave messages, etc. > > It doesn't really make sense to me. It only makes sense for some very > limited fields. e.g. when somebody cannot read or write (he hasn't > learned it or is blind). Everybody else could use the computer. Most > people who would use such a system do have a computer at home I guess. A lot of people thought the same before WWW, when text based contents were the basis of networks. A like the idea, maybe not exactly the Lenz proposal, but something similar... We will have a next step at the network communities. I will think about it. Thanks! Denis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Lock the phone when no using it
Em Sex 12 Nov 2004 13:11, Andrew Thompson escreveu: > Denis Galvão wrote: > > Is there a way(or some AGI app) to lock out the phone when user are not > > using it!? > > > > Thanks. > > You can set up an extension that does a dbput to a variable which > "locks" the phone. > > When someone tries to dial anything but 911 (or local equivalent), your > context should test for if the phone is locked and if so, fail it. Same > thing for calls going toward the phone if that's required. > > You would also need an extension and possibly a password to "unlock" the > phone, by setting the variable to another value. Nice Thanks Andrew. Denis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lock the phone when no using it
Is there a way(or some AGI app) to lock out the phone when user are not using it!? Thanks. Denis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] No sound with kphone 4.05 on SuSE 8.2 and asterisk
Check out your sound board specs, it has to support full duplex mode. Denis. Em Qua 10 Nov 2004 18:57, Volker Jahns escreveu: > Problem to get kphone 4.05 working w/ SuSE 8.2 > -- > > I amtrying to get asterisk running but I do get stuck at the first steps. > > asterisk installation OK. > UA kphone 3.13 (on SuSE 9.1 system) > UA kphone 4.05 (on SuSE 8.2 system) > > When connecting to the asteriask server with kphone both systems show up > on the CLI : > -- > *CLI> sip show peers > Name/usernameHostDyn Nat ACL Mask Port > Status 2000/marlies 10.XX.YY.ZZ D 255.255.255.255 5060 > OK (18 ms) 1000/100010.XX.YY.PP D > 255.255.255.255 5060 OK (21 ms) -- > > When trying to make connection between the 2 systems ringing is OK, but > once the connection is coming up, there is _no_ sound. > > -- > *CLI> sip show channels > Peer User/ANRCall ID Seq (Tx/Rx) Format > 10.XX.YY.ZZ marlies 6ee8c36c665 00104/0 ULAW > 10.XX.YY.PP 1000[EMAIL PROTECTED] 00103/01321 ULAW > 2 active SIP channel(s) > -- > > > n the SuSE 8.2 machine kphone (4.05) murmurs the following stuff: > -- > t=0 0 > m=audio 32772 RTP/AVP 0 97 3 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > > CallAudio: Using G711u for output > CallAudio: Sending to remote site 10.XX.YY.ZZ:32772 > UDPMessageSocket::SetTOS: Operation not permitted > CallAudio: Opening OSS device /dev/dsp for Input > CallAudio: Creating OSS->RTP Diverter > CallAudio: Sending to remote site 10.XX.YY.ZZ:32772 > UDPMessageSocket::SetTOS: Operation not permitted > CallAudio: Opening OSS device /dev/dsp for Input > CallAudio: Creating OSS->RTP Diverter > Attempt to start a thread already running > -- > > + the application evidently hangs: Thw windows don't get redrawn, > asterisk claims this client to be UNREACHABLE. > > > > ( The kphone rpm which is presented by SuSE will not come up as there > is a symbol from a qt library which is missing. So I had to compile > from src rpm. ) > > > What can I do? -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RES: [Asterisk-Users] Asterisk Brazillian Community
Hi Geraldo! We are at Latinoware 2004, Asterisk Brasil is getting high each day! Everybody wants to participate in our community, and it is great! Maddog bring his IAXy extension to our stand, and we are talking a lot about VoIP solutions at the open source world. We have his support, and it is great too! We(my company - iSolve) are showing a complete Asterisk implementation, with analog phone, and softphones. We are not restrict to just talk about Asterisk, but we can demonstrate its awesome power! Welcome Geraldo! ASAP we are posting some pictures. Thanks. Denis. Em Seg 08 Nov 2004 17:34, Geraldo Fco. do Espírito Santo Jr. escreveu: > Hi Denis, congratulations for the initiative. > > I would be glad to help. Feel free to contact me PVT. > > > Regards > > Geraldo > > -Mensagem original- > De: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Em nome de Denis Galvão > Enviada em: sexta-feira, 5 de novembro de 2004 17:55 > Para: [EMAIL PROTECTED] > Assunto: [Asterisk-Users] Asterisk Brazillian Community > > Hi all!!! > > Im proud to announce that we are creating an Asterisk Brazillian > Community! We are working hard to bring this "wonderful" piece of > software in our native language, brazillian portuguese. > > The community will get start after Latinoware 2004 > (http://www.latinoware.org) where we will give a lecture about VoIP and > Open Source. > > Im inviting all of portuguese speakers to join with us the "Asterisk > Brasil" > > community. > > If you wnat to go to Latinoware, we will give some Asterisk stickers for > all > > of you that want to participate in our community. > > As soon as possible I will send the wesite URL and other information. > > For now, if you want to contribute, send me an email. > > Best regards! > > Denis Galvão. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Brazillian Community
Hi all!!! Im proud to announce that we are creating an Asterisk Brazillian Community! We are working hard to bring this "wonderful" piece of software in our native language, brazillian portuguese. The community will get start after Latinoware 2004 (http://www.latinoware.org) where we will give a lecture about VoIP and Open Source. Im inviting all of portuguese speakers to join with us the "Asterisk Brasil" community. If you wnat to go to Latinoware, we will give some Asterisk stickers for all of you that want to participate in our community. As soon as possible I will send the wesite URL and other information. For now, if you want to contribute, send me an email. Best regards! Denis Galvão. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] how can I test canreinvite effectivness?
Try IPTRAF or TCPDUMP. Denis. Em Qui 14 Out 2004 19:12, Matthew Boehm escreveu: > I'm not running X or any kind of GTK/GUI abilities on our asterisk > server. I need some sort of ability to test wether sip canreinvite is > working. > > If it is, then the network usage should be minimal/nonexistant because > all voice packets should be going phone-to-phone. > > If it is not, then network usage would be high because all voice packets > would be going phone-to-asterisk-to-phone > > Does anyone know of a nice ncurses or terminal based realtime network > usage app? > > Or is there some other way in asterisk I can tell if the phones are > talking to each other directly? > > Thanks, > Matthew > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
