Re: [Asterisk-Users] Digium Hardware Reliability
I would love to see some feedback on this as well. I've lost exact count now, but think I've seen about 5-6 failures on their cards TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with that many systems, which makes this really concerning. I've started a thread on the Asterisk Forum to get more feedback on the Sangoma cards as an alternative. I'm finding it hard to think this experience is a total fluke - it would be great to hear other people's experience though - good or bad.philippe<[EMAIL PROTECTED]><[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: "M.Hockings" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Thu, 29 Jun 2006 21:38:20 -0400Subject: [Asterisk-Users] Digium Hardware Reliability How reliable is Digium hardware in general.? My new TDM400P just died.I am trying to determine if I have a lemon. This a new PC with a Digium TDM400P in it with a single FXO and single FXS card just stopped working today. It has been running less than three weeks with the the FXS card and has the FXO card in it only for about a week. Today the power went out due to a mis-configuration on my part the UPS shut down before the machine shut down. Now, I would not think this should be a problem but the Digium card no longer responds. lspci does not show it either so I presume it deadSo, at over 2x the cost is Sangoma hardware more sturdy than the Digium stuff?Right now we are back using the POTS phones with the nice new SPA-922's looking like cute paperweights.Mike (totally UNimpressed with Digium)___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.___ --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] password on radius authentication
Well, I know to be compatible with porta-billing you need password to do ip based auth. It's a bit goody but they basically seem to expect if trusted ip and no Digest support then radius auth has username=src_ip and password=x. To put it another way it would be help full to porta-billing users to be able set username and password fields on auth being sent via radius to porta-billing. So in a round about way I would say yes I can probably test the module against some things for you. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dennis Nacino Sent: Wednesday, June 28, 2006 6:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] password on radius authentication Hi, It's kind of off-topic , but still within Asterisk. I developed an asterisk module that send an authentication to a radius server for call authorization and process its reply (limited to User-Name and Cisco or Quintum VSA h323 attribute). My question, is when it make sense to use or include the attribute Password/User-Password? Looking on PDF's of Quintum and Cisco none of it really make use of this attribute. Any comment? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] asterisk shutdown
On Thu, Jun 29, 2006 at 10:54:58PM -0500, Anton Krall wrote: So, no answers? Nobody knowd why this might be happening? Nobody else experiencing this? Is this a reproducable issue? Have you turned on verbosity and debug and log them (e.g. the full log)? If still no messages and this is reproducable, consider running asteriskunder strace (-f). Though I figure that this could be a major performance hit. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] voting,suggestiuon,your input needed to all
ok, We are building the perfect voip company..we are trying...we need input on end-users:reply to my email with --ENDUSER in subject.with anything you would like to see your current voip provider offer online/offline ( don't say.. support, an answer on phone etc) be constructive.. reply to myt email with --WHOLESALE in subject with anything you would like to see for wholesale (resellling itsp like services, rebranding,whitelable, per client rates. etcreply to our email [EMAIL PROTECTED] only not to start an endless thread..we WILL make those features happen, we actually got 4 engineers that are doing only requests from clients.Please occupy them as we them anyhow. ;) push things you want to see. ( no non asterisk things) EG: T38 NOT GOOD. as if we do then wont be branched and all loose...only things you would want your provider/partner to have ..Only the things we always hear around here.. I WISH..Thanks and let's make it happen.PS BTW contact us for custom IVR/PBX/ANYTHING programming.-- MikeSales Manager http://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --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] SIP reinvite still does not occour
Roger If transcoding happened (from G.729 to GSM, any to any) then Asterisk will issue no re-INVITE, for sure. Pls. change Disallow=all Allow=gsm (only one codec) Then test, you'll see it happen. Cheers Hoa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Schreiter Sent: Friday, June 30, 2006 8:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP reinvite still does not occour Hi, I have in my sip.conf disallow=all allow=alaw in order to avoid any codec problems disturbing reinvite. And of course I have: canreinvite=yes In extensions.conf there is only one Dial command. It has no qualifiers like t or T. Just Dial(SIP/[EMAIL PROTECTED]) Anyway, asterisk does not try to reinvite. asterisk tells -- Attempting native bridge of SIP/01234567 ... but in the debug output there no reinvite. Using tcpdump I can see, that the audio data are going via the asterisk box in the middle, not direct between the endpoints. Is there anything else, which can prevent a reinvite? dtmp-settings? nat-settings? Thanks for any hints! Roger. ___ --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] cheapest Cisco Smartnet contract?
Hello, I've got a few Cisco phones to maintain and need access to firmware files. Dealers here in .fr want unreasonable prices for a Smartnet subscription. Where can I get a better deal on the Net? Thanks, ___ --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] SIP reinvite still does not occour
Hoa Thai Duy schrieb: Roger If transcoding happened (from G.729 to GSM, any to any) then Asterisk will issue no re-INVITE, for sure. Pls. change Disallow=all Allow=gsm (only one codec) Hi, yes, to avoid transcoding problems I only have one codec, just alaw. Anything else is disallowed. That's why I don't understand, why there is no reinvite. Thanks for answering! Roger. ___ --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] Queue - Log if caller disconnects
Hello List, i'm wondering if there is any way to get a AGI executed if a caller disconnects while he is INSIDE the queue application. If so, i would like to log the call as missed. Hope someone can help. Greetings, Michael ___ --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] need help troubleshooting clipping and garbled VOIP calls
T. Shaw wrote: Hello all, I have a problem with call quality with my Asterisk setup. I'm doing VOIP only so far, but have a zaptel TDM400P in the box not being used. The problem i'm having is that when calls are placed, connected, and the far-end is reporting that they are experiencing clipping, choppy, and garbled voice conversations. So bad that we have to resort to using our cell phones. This entire setup is still being built, but any phone attached is experiencing this. Call volume is almost nil (under 20 total incoming calls a day). This is a small business setup. The server is used exclusively for Asterisk, so it isn't a fileserver, or anything else. The setup is as such: ipphone ---cisco 2900XL switch Cisco 2621 router --- dsl modem --DSL --- VOIPprovider I've configured the switch and the router to set priority and qos to prioritize voice traffic above data. Funny thing is, there is not data REALLY hitting the network. I have setup 2 vlans, data vlan, and voice vlan. There are two work stations on the network, and neither is being used to hit the internet heavily (office is still being setup). Any pointers or suggestions anyone have for me as to were to look for this poor quality? It seems only the Far-end (called party), is hearing this and not the calling party. I haven't tried switching out the phones because we only have 1 type, and any of the phones i used exhibit these problems. I will try softphones to see if it is truly a networking issue or Phone issue. Is anyone using a cisco 2900 switch or router and care to provide config samples of their COS/QOS setup? Thanks! Terrelle Shaw I've got a similar setup (which does have a TDM card and voip incoming and outgoing), for some reason an IAX provider (which provides most of our calls incoming and outgoing) has this problem, whereas a different SIP one doesn't seem to. I have checked my traffic shaping script, and everything seems fine, the same provider works flawlesly from home, with a simliar setup (only without a timing source and a cable modem). I'd be very interested to see what you find out. ___ --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] Avaya 4610sw SIP setup problem
Hi I tried that too, but the only useful thing I can change (besides the IP settings of the phone itself) is the "CallSv" parameter; I set it to the IP of the SIP registrar/proxy but it still doesn't work... Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of HenkSent: 29 June 2006 21:16To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Avaya 4610sw SIP setup problem Did you try to manually to change the parameters of the phone? When you power the phone up then are you able to enter manually the parameter when you hit *. I am using a 4610 with Release 2.2 but I am not using the capability to upload the settings from the server. Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herchi SilviuSent: donderdag 29 juni 2006 15:55To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Avaya 4610sw SIP setup problem I just tried serving the files off Apache, port 80, no change... Most parameters are taken into account by the phone, except for SIP proxy and SIP registrar... Coud someone post an excerpt from their 46xxsettings.txt where I could see the format they use? Thank you in advance, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 29 June 2006 00:33To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem I too am using 2.2.2, but I'm loading my config files via HTTP. I was having some difficulty when I was using TFTP. Things were not as reliable for me, so I switched to HTTP. I've been stable since. On 6/28/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom LynnSent: 28 June 2006 05:35To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying "Registering" for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN " sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219 " SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o= avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support. avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls
Martin Joseph wrote: On Jun 29, 2006, at 2:43 PM, T. Shaw wrote: thanks for all the responses. I feared that it might be a bandwidth issue. We have a (supposedly) business DSL line that is 1.5M - 3M down/ 512k up. might have to bump that up to a higher grade. If you are actually getting what you describe above you should have no need to upgrade in order to support 3 or 4 simultaneous calls... I have only 384K bit's upstream on my home DSL and that's fine for 2 uLaw calls. If you switch to GSM or G729 that should allow for even greater simultaneous call volume... Accorging to the calculator on asterisk-guru (which I know isn't perfect), you should be able to manage at least 30 calls with trunked IAX and G.729. I did take your suggestion and contact my VOIP provider. They suggested to two things: 1) Use SIP to trunk with them instead of IAX ( they said that lots of people complain about the conenction with IAX, but when they use SIP the issues get better) ? This sounds like they have an issue. With (presumably) a different provider, I seem to be getting a similar problem, at home (even with the same accounts) IAX-in and IAX-out causes no problems, with G.729, whereas on site IAX-in and IAX-out calls have clipping and buzzing, with same handsets, same codec only differences being that on site there is a timing source (TDM400) and the IAX channels are trunked. SIP-in doesn't seem to cause the same problems on-site. ___ --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] OH323 issue on AT320 Phones
Hi all, I installed asterisk 1.2 branch, with oh323 channel support. Everything is fine, with netmeeting I can call and receive incoming calls, internal and external Then I tried to setup an AT320 phone , which is based on PA168S chip. I can receive call from internal or external phones, and talk to remote . I can place calls both to internal and external phones, but when remote answers (asterisk console sayng, on example SIP/944 answered OH323/945,@192.168.88.4 ) the AT320 phone continue ringing and saying calling; in other words asterisk is not able to notify back to the at320 the answering of its call. Is there any further debug I can enable ? thanks, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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] Work required - modify Asterisk + SEMS
Hi Jeremy, Thanks for your suggestion - but our project requires certain features that have to be additionally implemented, which means we cannot work with what is out there already. Best regards, Mike Jeremy McNamara wrote: Mike Puchol wrote: Hi all, I am looking for a developer or developers that can implement the following: - Modify an Asterisk server in order to support one inbound RTP and several outbound RTPs, I was thinking SEMS may provide a very good starting point. The idea is to make a PA system over IP. We do *not* want full-duplex audio. - Implement a client in Qt/C++, that allows to send audio to this platform, and plays back audio received from it (Windows-based). We are thinking about Speex for the codec, as there are no royalty issues. Interested parties please reply with your comments, capabilities, so we can start discussing the project. why not setup a listen only meetme for the 'listeners' and talk only for the 'talker'? Jeremy McNamara P.S. Cross posting is not a friendly way to generate discussion, just flames ___ --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] Limiting a group of phones available channels
Hi List I have 10 separate SIP phones, and I wish to limit the simultaneous available channels to 5 maximum for these. How would you go about it without setting up a separate * box? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] Limiting a group of phones available channels
On Fri, 2006-06-30 at 13:31 +0400, Jean-Michel Hiver wrote: Hi List I have 10 separate SIP phones, and I wish to limit the simultaneous available channels to 5 maximum for these. How would you go about it without setting up a separate * box? Cheers, Jean-Michel. you can limit it to the provider end by doing a limit, read the page below for 1.2 notes, as the naming changed. You can do a setgroup/checkgroup in the dialplan putting all 10 people into the same group. Lastly, and probably the least effective, is you can watch channel usage and when someone exceeds 5 run over to their desk and smack them with a rotten fish. http://www.voip-info.org/wiki-Asterisk+sip+incominglimit -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --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 kannel with asterisk
I don't use asterisk in combination with kannel. Actually we use nowsms as SMSC gateway to connect to our provider but we deside to replace it by kannel. so we store incoming messages in an sqlserver 2005 database in windows 2003 server . please let me what you need to combine kannel and asterisk ? thanks Regards issam - Original Message - From: Tomislav Vojvodic To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, June 29, 2006 11:00 AM Subject: RE: [Asterisk-Users] using kannel with asterisk Well kannel by itself doesen't use much resources as far as I remember.. it's all about actions taken upon receiving sms.. Please let me know your experiences since I'm also interested in kannel / asterisk combination.. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issamSent: Thursday, June 29, 2006 10:59 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] using kannel with asterisk hello I have an asterisk server with a te110pE1 digium card. the server is a hp ML370 3,2 Ghz 64bits, 1Mo L2 , 1Go Ram, 3 SCSI 73Go in raid5. I want to use in the same machine the kannel SMSC. i have no big trafic in the two gateway but I want to know if it generate a performence problem for asterisk I use fedora core4 with latest asterisk version . thanks Regards issam __ NOD32 1.1632 (20060629) Information __This message was checked by NOD32 antivirus system.http://www.eset.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] OH323 issue on AT320 Phones
[EMAIL PROTECTED] wrote: Hi all, I installed asterisk 1.2 branch, with oh323 channel support. Everything is fine, with netmeeting I can call and receive incoming calls, internal and external Then I tried to setup an AT320 phone , which is based on PA168S chip. Which version of the PA168S firmware are you using? (lastest is 1.52). Why are you using the H.323 firmware? (since the IAX firmware works, as does the SIP, although I'd recommend the SIP, since attended call transfer doesn't work with the IAX version). ___ --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] Limiting a group of phones available channels
trixter aka Bret McDanel wrote: Lastly, and probably the least effective, is you can watch channel usage and when someone exceeds 5 run over to their desk and smack them with a rotten fish. http://www.voip-info.org/wiki-Asterisk+sip+incominglimit I can't find the 'rotten fish' stuff documented anywhere on voip-info.org - was that some sort of red herring? ___ --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] Limiting a group of phones available channels
On Fri, 2006-06-30 at 10:44 +0100, Steve Langstaff wrote: trixter aka Bret McDanel wrote: Lastly, and probably the least effective, is you can watch channel usage and when someone exceeds 5 run over to their desk and smack them with a rotten fish. http://www.voip-info.org/wiki-Asterisk+sip+incominglimit I can't find the 'rotten fish' stuff documented anywhere on voip-info.org - was that some sort of red herring? Its an advanced asterisk management option. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --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] OH323 issue on AT320 Phones
I am using latest firmware, exactly 1.52 I am used to use PA168S phones in SIP mode (in the past I had problems using them as IAX., i.e. passing calls and so on) This is only for a test purpose, to test OH323 channel. It is not a crritical issue, i never will use H323 on PA168S phones in a production environment. I was only very suprised about this strange behaviour, and I would like to investigate it. Andrea Thomas Kenyon [EMAIL PROTECTED] ius.co.uk To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 30/06/2006 11.42 Re: [Asterisk-Users] OH323 issue on AT320 Phones Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com [EMAIL PROTECTED] wrote: Hi all, I installed asterisk 1.2 branch, with oh323 channel support. Everything is fine, with netmeeting I can call and receive incoming calls, internal and external Then I tried to setup an AT320 phone , which is based on PA168S chip. Which version of the PA168S firmware are you using? (lastest is 1.52). Why are you using the H.323 firmware? (since the IAX firmware works, as does the SIP, although I'd recommend the SIP, since attended call transfer doesn't work with the IAX version). ___ --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] IAX jitter / clocking problem
hello, I'm still trying to tune iax jitterbuffer in asterisk 1.2.9.1, but without success, I'm using idefisk-asterisk over cdma network, where rtt is about 100-500ms, so jitter about 400ms but sound is very jerky, in diection idefisk-asterisk, in reverse direction is sound relatively smoth, so, my question: has iax same problem as in sip/rtp, where packets are generated along incomming packets (what is probably solved in trunk with: http://bugs.digium.com/view.php?id=5374 0005374: [patch] Asynchronous generation of outgoing frames when timing device available my iax jitterbuffer settings (iax.conf): [general] jiterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=1500 resyncthreshold=-1 thanks for suggestions ;-) PJ ___ --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] ISDN: 3° incoming call
Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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] Problems with dial status...
Hello for the first time :-) I have a huge problem trying to create some sort of call back system. What am I trying to do? I call Asterisk, press 1 to call someone back and play announcement. Hanging up. Then I'm creating a file:- Channel: Zap/2-1/07966011122 Context: call-them-back Extension: s Priority: 1 And moving it into /var/spool/asterisk/outgoing The asterisk gets the file and goes to context 'call-them-back', which is:- [call-them-back] exten = s,1,Answer exten = s,n,Wait(1) exten = s,n,Playback(IVR/premium/calling-back) exten = s,n,Hangup() But the problem is asterisk executes Playback() before the call is actually connected. (On the console it says that Zap/2-1 answered while it's actually trying to ring on my mobile). How can I resolve this problem? I'm based in the UK if it matters. Thanks a lot for any help! Martin ___ --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] Queue - Log if caller disconnects
Asterisk logs very detailed information in /var/log/asterisk/queue_log file including abandoned calls. You can import this log to mysql with a simple perl script running periodically. -Original Message- From: Michael Konietzny [mailto:[EMAIL PROTECTED] Sent: Friday, June 30, 2006 11:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Queue - Log if caller disconnects Hello List, i'm wondering if there is any way to get a AGI executed if a caller disconnects while he is INSIDE the queue application. If so, i would like to log the call as missed. Hope someone can help. Greetings, Michael ___ --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] IAX jitter / clocking problem
Pavel Jezek wrote: hello, I'm still trying to tune iax jitterbuffer in asterisk 1.2.9.1, but without success, Here is my entries: jitterbuffer=yes dropcount=3 maxjitterbuffer=1000 maxjitterinterps=10 maxexcessbuffer=80 resyncthreshold=1000 minexcessbuffer=10 jittershrinkrate=1 -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Problems with dial status...
Marcin Lukasik wrote:
But the problem is asterisk executes Playback() before the call is
actually
connected.
(On the console it says that Zap/2-1 answered while it's actually
trying to
ring on my mobile).
This has been covered on the list many times, search the archives, the
Wiki and Google are your friend.
On a zap channel, Asterisk can't tell when a call has been answered, so
starts the playback immediately. Setup a loop asking the caller to
press a key. I have the following setup:
[voice-mail-callback]
;
; Set timeouts
;
exten = s,1,Set(TIMEOUT(response)=6)
exten = s,2,Set(TIMEOUT(digit)=3)
exten = s,3,Wait(5)
exten = s,4,Set(COUNT=0)
; ***
; Play, your attention is required, press 1 to
; collect voice mail
; ***
exten = s,5,Background(attention-required)
exten = s,6,Background(press-1)
exten = s,7,Background(to-collect-voicemail)
; *
; If 1 is pressed, then play transfer and
; then jump to voice-mail context.
; *
exten = 1,1,Playback(pbx-transfer)
exten = 1,2,Goto(voice-mail,s,1)
;
; Setup a variable to count the number of
; times the message has been played, when
; $COUNT reaches 5, play you've taken
; to long to dial and hangup.
;
exten = t,1,Set(COUNT=$[${COUNT} + 1])
exten = t,2,NoOP(${COUNT})
exten = t,3,GotoIf($[ ${COUNT} 5 ]?103)
exten = t,4,Goto(voice-mail-callback,s,5)
exten = t,103,Playback(local/tolong-todial)
exten = t,104,Playback(goodbye)
exten = t,105,Hangup()
exten = i,1,Playback(local/sorry-invalid-choice)
exten = i,2,Set(COUNT=$[${COUNT} + 1])
exten = i,3,NoOP(${COUNT})
exten = i,4,Goto(voice-mail-callback,s,5)
exten = h,1,NoOP(Hungup)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
___
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Re: [Asterisk-Users] Problems with dial status...
This has been covered on the list many times, search the archives, the Wiki and Google are your friend. On a zap channel, Asterisk can't tell when a call has been answered, so starts the playback immediately. Setup a loop asking the caller to press a key. I have the following setup: Doug, Thank you. I am using my friends, and I know it's.. well... not possible... But when I use Dial() to call on my mobile, it is not saying that call has been connected unless you actually answer it. So it makes me wonder why... that's why I'm asking. Thanks, Martin ___ --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] BLINDTRANSFER
Hi List, i'm fiddling around with a blindtransfers. (and 3PTY) a calls b a transfers b to c (blindtransfer) (c is not a party but a makro which puts b into a MeetMe conference) the conference should be dynamically created. and named after the callerid of a therefor b has to know who which callerid --transfered-- him. is there a VARIABLE or something else, where i can look up WHO transfered b? thx Kai ___ --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] ISDN: 3° incoming call
BRI ISDN is 2 channels, what would you want to do with a 3rd call? Julian On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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] Best GPL Gui?
Hi Guys With the profusion of different GUI's and Web interfaces out there could someone possibly save me a load of time and let me know which is the best one and why? Also is there an independent site reviewing asterisk GUI's anywhere. I'm looking at Cisco phones and TDM400 and X101P cards. Only GPL versions please. TIA Paul ___ --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] Queue - Log if caller disconnects
I am having the same problem with my IAX clients. I posted some issues that are causing my remote IAX agents to be disconnected due to errors in setting up the IAX stream. I have found that calls will abandon when a dynamic agent is logged into a down phone, the agent obviously cant logout if they cant call the switch back. The caller seems to be disconnected when being transferred to an agent that is logged into a down phone. I am using least recent routing. I had thought that asterisk at very worst would try to transfer to the agent, see the phone down, timeout on rings or not ring at all, and then log the agent out. I am definitely missing something or mis-reading my instructions. Please post your resolution and I will do the same. Jordan Novak Communications Technician ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Hardware Reliability
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Philippe Lindheimer wrote: I would love to see some feedback on this as well. I've lost exact count now, but think I've seen about 5-6 failures on their cards TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with that many systems, which makes this really concerning. I've started a thread on the Asterisk Forum to get more feedback on the Sangoma cards as an alternative. I'm finding it hard to think this experience is a total fluke - it would be great to hear other people's experience though - good or bad. H... I have around 10 TDM400 in the field with out a single failure. I also have 6-8 sangomas A200's in the field with no problems... Sangomas are not twice the price at all... Sangoma a200 with 2xFXO = $249.95 Digium TDM402b= $225.90 The only time the cost really goes up with the sangoma is when you add the echo cancellor. (1 time cost of $300 roughly) at that point you can only compare the card to the TDM2400 with echo cancellation and then it too is an even cost. Both cards in my experience are very reliable, the sangoma IMHO gives you a little more flexibility in terms of a smaller system and experimenting with echo cancellation or no echo cancellation. philippe From: M.Hockings [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Thu, 29 Jun 2006 21:38:20 -0400 Subject: [Asterisk-Users] Digium Hardware Reliability How reliable is Digium hardware in general.? My new TDM400P just died. I am trying to determine if I have a lemon. This a new PC with a Digium TDM400P in it with a single FXO and single FXS card just stopped working today. It has been running less than three weeks with the the FXS card and has the FXO card in it only for about a week. Today the power went out due to a mis-configuration on my part the UPS shut down before the machine shut down. Now, I would not think this should be a problem but the Digium card no longer responds. lspci does not show it either so I presume it dead So, at over 2x the cost is Sangoma hardware more sturdy than the Digium stuff? Right now we are back using the POTS phones with the nice new SPA-922's looking like cute paperweights. Mike (totally UNimpressed with Digium) ___ --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 -- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta. http://us.rd.yahoo.com/evt=42297/*http://advision.webevents.yahoo.com/handraisers -- ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEpRfj1Kolm8VQlAURAh8DAJwOCXhwFPMp9pcslNk9yW4TR8zLlQCgqj5R 8qtsHCpUDYIbqrOMWbcT0Mw= =mIkX -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Hardware Reliability
M.Hockings wrote: How reliable is Digium hardware in general.? My new TDM400P just died. I am trying to determine if I have a lemon. This a new PC with a Digium TDM400P in it with a single FXO and single FXS card just stopped working today. It has been running less than three weeks with the the FXS card and has the FXO card in it only for about a week. Today the power went out due to a mis-configuration on my part the UPS shut down before the machine shut down. Now, I would not think this should be a problem but the Digium card no longer responds. lspci does not show it either so I presume it dead So, at over 2x the cost is Sangoma hardware more sturdy than the Digium stuff? Right now we are back using the POTS phones with the nice new SPA-922's looking like cute paperweights. I've been using a TDM04b since it came out with no failure problems. Digium did replace the card and modules early on, when a card design issue was identified. That's been about two years or so. If you look at the components used on the Digium and Sangoma cards, they are almost identical. Therefore, from a heat generation perspective, both are likely to have the same issues without a reasonable air flow. Given the number of folks on the list that attempt to do things without the technical skills, knowledge, experience, configuration issues, plugging a telco line into a fxs port, poor power supplies, no ups, poor air flow, etc, etc; I don't think you're going to get any reasonable response from the list in terms of reliability. For those that seem to have the necessary skills, both Sangoma and Digium cards seem to be fine. Since the TDM card carries a two year warranty (or greater), I'd suggest you contact digium tech support and ask for their assistance in determining whether the card is actually bad. You've already paid for that assistance and they can handle warranty issues as well. R. ___ --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] Surge Protector for T1/PRI ?
Just recently a client of mine took a lightning hit, which in turn blew out their Digium TE411P board. This just so happened to be their main office where their call center was located. We had a backup card on hand, but this still meant downtime for the client until we got out there to replace the card. I was thinking - what if we put a surge protector device between the PRI card and the circuit itself? That way, the client themselves could replace surge protector units (if it got hit again) and protect our expensive telco equipment from getting damaged. Has anyone else experience surges on a T1/PRI circuit? What did you do to prevent further issues? Anyone from Digium - do you see a surge protector device causing interferrence or a problem with the equipment? Example device I'm looking at: http://www.apc.com/resource/include/techspec_index.cfm?base_sku=PDIGITEL Thanks!! ___ --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] Recommended FXO device
Chris Mason wrote: I have a client's installation that requires 4 lines PSTN interface only so I am looking at 4 port FXO units. What works well with Asterisk and is not exorbitant to purchase? Would a Sangoma remora be better? The Sangoma A200D card has better echo canceller (if needed) compared to the Digium TDM card, plus the A200D card does support modem calls (eg, faxing, POS) when the modem is plugged into a fxs port on the same card as the fxo ports. For external adapters, the only one that I've tested that functioned reliably was the Mediatrix 1204 at a rather heafty price. It did an excellent job with echo cancellation, audio levels, and audio quality; but, their implementation (about two years ago) was very non standard with absolutely no security, etc. That may have changed now, but at the time, it could only be configured using snmp and with an snmp community string of public (which could not be changed). Rather difficult to set up, but once configured it just worked. I've tested a large number of other external adapters and have not found a single one that had a reasonable echo canceller built in. Many of them work fine on short pstn lines (where echo is much less of a problem), but provided even reasonable service on longer pstn lines or lines that involve unusual telco configurations (eg, remote line concentrators). R. ___ --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] cheapest Cisco Smartnet contract?
I've got a few Cisco phones to maintain and need access to firmware files. Dealers here in .fr want unreasonable prices for a Smartnet subscription. Where can I get a better deal on the Net? You probably can't legally. Cisco controls who is allowed to resell their contracts very very closely, and sets the pricing for those resellers. Been discussed numerous times over the last three years. So, you're really stuck paying their prices, or, doing something illegally. No other choices. ___ --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] FOSS, Science, and Public activism
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 (Sorry if you get more than one copy of this message, but I felt that it was urgent to get this important info out.) The values of freedom and openness are crucial to understanding itself, so that civilization and public welfare now depend on them, as I argue below. These values may find their best expression in the free and open source software (FOSS) movement, and the foresightful example of FOSS developers should now be beneficially applied to many other disciplines in the context of a global and public Internet. It is crucial that we occasionally take time to discuss the reasons _why_ we release our source code, and this is one of those occasions. There are good reasons for the freedom and openness which are characteristics of FOSS development, reasons which should receive wider attention now that they can be readily communicated to other arenas. The consequences of doing otherwise are often catastrophic. For example, it incomprehensible that Genentech could consider withdrawing a cheap cure for blindness (ARMD) from the market. http://lists.essential.org/pipermail/random-bits/2006-june/001374.html The mechanism of this drug is public knowledge. http://sourceforge.net/mailarchive/forum.php?thread_id=14183567forum_id=6042 This abhorrent situation is a great example of the kind of thing that will happen if people don't get behind the values of freedom and openness that we are espousing. Please let Genentech know that you find what they are doing offensive. Publicize the mechanism so that new compounds can be obtained as replacements. For the future, continued vociferous public activism is required to prevent such outrages from occurring in the future. It becomes clear that the compounds which come from common roots, fruits, and vegetables are a shared human heritage and the free and open source of the future. Tannins are another interesting case in point, because as molecules, and as anti-oxidents, they are similar to resveratrol (resV), and that molecular mechanism has been anchored to the public domain via a prior art declaration. It is a so-called CR-memetic, which may increase healthy human longevity by many decades. Here are some links about it. Resveratrol mechanism posts from GNU-Darwin list http://proclus.gnu-darwin.org/gdposts.html CR protocol for human bodies http://proclus.gnu-darwin.org/bootstrap.html Here is some important recent news about it. http://www.imminst.org/forum/index.php?s=act=printclient=printerf=237t=10749 It is exciting to suppose that people can get off the pharmaceuticals that they are taking with calorie restriction or CR-memetics. I personally am trying to get off the cholesterol drug Pravachol, a statin compound, starting a few of weeks ago. Write me, and I'll let you know how it turns out. From the article... Fontana says ... evidence of younger hearts in people on calorie restriction, suggest that humans on CR have the same adaptive responses as did animals whose rates of aging were slowed by CR. I think that it is time to look at the tannins in tobacco leaves. There may be other treasures lurking there too. As you may be aware there is ample public research into any possible beneficial compounds that may be obtained from tobacco leaves. The mechanisms are there waiting to be discovered. If you want to post them, just reply to me and I'd be delighted to host them. The public establishment of prior art is a time-honed method of entering inventions into the public domain. We now have other methods at our disposal as well. If you are planning to establish prior art against future CR-memetic related patents, you might want to have a look at www.creativecommons.org. Perhaps it goes without saying at this point that you should please choose a license that provides for free and broad public access to your memetic. In that way you will assure that the public health is served by anchoring them to the public common, where they cannot be exploited by those who would withhold them for their own profit. The DRM situation is precisely analogous to this. Can you imagine doing science in a world where your ability to read and write your data is filtered through secret protocols that are hidden from you? I recommend the Defective By Design campaign to fight the outrage of DRM, which is incompatible with the scientific pursuit. http://www.defectivebydesign.org/ It is clear that scientific tools must be demonstrably and penetratingly understood, or else our claims will likely be skewed and called into question. Free and open source software is a great example of how to make your science verifiable to the public. Establishing prior art against future patents is another good one, which is precisely analogous in method, making the result explicit to the public, free and open to all. Thank goodness for the free and open software movement, which gave us such a great example of how to serve
RE: [Asterisk-Users] SNOM Softphone on windows 2000
Having your users as admins on the local machine is generally a bad thing to do, that means that any virus and/or spyware can install itself into the machine without a problem. It would be nice to know in what key SNOM stores the reg info, so that one can simple grant full access to 'authenticated users' for the settings. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of SANS Sent: Thursday, June 29, 2006 10:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 Sorry had to jump in. I had a similar problem with Mozilla. Make sure the Users can write to the config file. I just made all the Users an Administrator at the local machine from Local Users menu, and that fixes write to issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Thursday, June 29, 2006 10:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 Well we do write to the registry... Sorry about that, but how would we otherwise store the information that is needed for the phone?! CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Thursday, June 29, 2006 4:01 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 W2K had problems with Security (Surprising huh?) You may need to grant write access for the user to the Folder where SNOM is installed. I don't think SNOM is writing to the registry if so you will need to open permissions up on those keys in the hive. ___ --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 ___ --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] SNOM Softphone on windows 2000
A config file in text would be nice. Oh wait this is windows based, config files don't exist anymore!!! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Thursday, June 29, 2006 10:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 Well we do write to the registry... Sorry about that, but how would we otherwise store the information that is needed for the phone?! CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Thursday, June 29, 2006 4:01 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000 W2K had problems with Security (Surprising huh?) You may need to grant write access for the user to the Folder where SNOM is installed. I don't think SNOM is writing to the registry if so you will need to open permissions up on those keys in the hive. ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Hardware Reliability
On Thursday 29 June 2006 21:38, M.Hockings wrote: How reliable is Digium hardware in general.? My new TDM400P just died. I have a number of Digium T1 products (T100P, TE410P, TE405P and TE406P) as well as a few TDM400 based boards. No failures in the last 2 years or so. So, at over 2x the cost is Sangoma hardware more sturdy than the Digium stuff? Not that I've seen. I also have a number of Sangoma products. Both work very well for me. As an engineer, I can also see that the protection on the interfaces is comparable. Mike (totally UNimpressed with Digium) I don't think this is a Digium problem, at least not yet. What did their customer service people say? Can you ask for a failure report? You note that power went out. Generally when this occurs there is a very high chance of transient voltage spiking or line swells not only on the residential electrical power grid but also on the telephone network. Do you have any telco line protection in place to protect the card from nasties coming in from the outside? Is the protection correctly installed? How about electrical protection? The MOVs in your power strip and UPS are only good for a few hits before they become ineffective (something they never tell you). Unless you know something more than you've presented here it is a little premature to start pointing fingers. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Hardware Reliability
On Friday 30 June 2006 02:24, Philippe Lindheimer wrote: I would love to see some feedback on this as well. I've lost exact count now, but think I've seen about 5-6 failures on their cards TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with that Then put proper telco line protection in place! Good lord, it's blindingly obvious to me that you seem to be in a particularly harsh environment and that the protection on the FXO modules was not designed for the type of transient disturbances you're experiencing. -A. ___ --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] IAX2 Jitterbuffer and trunking
Is there a fix for the problems with using the jitterbuffer on a trunked IAX2 in asterisk 1.4 ? ___ --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] Surge Protector for T1/PRI ?
The APC units work well, they have a rackmount module system also. Protect yourself from grounding mismatch with security and paging systems on channel banks also. Talk with your clients about emergancy repair/replacement. On 6/30/06, Dustin Wildes [EMAIL PROTECTED] wrote: Just recently a client of mine took a lightning hit, which in turn blew out their Digium TE411P board. This just so happened to be their main office where their call center was located. We had a backup card on hand, but this still meant downtime for the client until we got out there to replace the card. I was thinking - what if we put a surge protector device between the PRI card and the circuit itself? That way, the client themselves could replace surge protector units (if it got hit again) and protect our expensive telco equipment from getting damaged. Has anyone else experience surges on a T1/PRI circuit? What did you do to prevent further issues? Anyone from Digium - do you see a surge protector device causing interferrence or a problem with the equipment? Example device I'm looking at: http://www.apc.com/resource/include/techspec_index.cfm?base_sku=PDIGITEL Thanks!! ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] Integrate asterisk with Database
Hi All, I am plainging to give a solutions for a sports club. Follwing is the process that i need to achieve. If any body achieve this kind of setup pls give me a feedback, so that i can go through. Call flow start [for database operations please use an access database with suitably configured fields] Thank you for calling Sports World Press 1 for English, 2 for French Please key in your 12 digit membership number Your membership number is [repeat the digits that have been keyed in] Press 1 to confirm or 2 to key again [loop until confirmed] [exit after three invalid attempts and say] Invalid membership number. Please call customer services. [Access database and lookup the membership number and check validity field in database] [If missing or invalid bin] The membership number is invalid. Transaction terminated. [exit at this point] Please enter your mobile number Your mobile number is [repeat the digits that have been keyed in] Press 1 to confirm or 2 to key again [loop until confirmed] Please enter your activity code [4 digits] Your activity code is [repeat the digits that have been keyed in] Press 1 to confirm or 2 to key again [loop until confirmed] Please enter your activity duration [upto 5 digits] Your activity duration is [repeat the digits that have been keyed in] Press 1 to confirm or 2 to key again [loop until confirmed] [store mobile number, activity code and activity duration in the database] Your transaction is complete. Thank you for using sports world. ..Call flow end .. Thanks Regards, Vidura Senadeera. ___ --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 kannel with asterisk
If you'll use newer distribution of linux you'll probably jump into problems with libsqlite3 (libsqlite2 is needed for kannel).. it is well documented on kannel website.. you can contact me off-list about kannel since this isnt't kannel mailing list... I got kannel and asterisk running under CentOS 4.3 with forced sqlite2 install ;) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issam Sent: Friday, June 30, 2006 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] using kannel with asterisk I don't use asterisk in combination with kannel. Actually we use nowsms as SMSC gateway to connect to our provider but we deside to replace it by kannel. so we store incoming messages in an sqlserver 2005 database in windows 2003 server . please let me what you need to combine kannel and asterisk ? thanks Regards issam - Original Message - From: Tomislav Vojvodic To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, June 29, 2006 11:00 AM Subject: RE: [Asterisk-Users] using kannel with asterisk Well kannel by itself doesen't use much resources as far as I remember.. it's all about actions taken upon receiving sms.. Please let me know your experiences since I'm also interested in kannel / asterisk combination.. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issam Sent: Thursday, June 29, 2006 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] using kannel with asterisk hello I have an asterisk server with a te110pE1 digium card. the server is a hp ML370 3,2 Ghz 64bits, 1Mo L2 , 1Go Ram, 3 SCSI 73Go in raid5. I want to use in the same machine the kannel SMSC. i have no big trafic in the two gateway but I want to know if it generate a performence problem for asterisk I use fedora core4 with latest asterisk version . thanks Regards issam __ NOD32 1.1632 (20060629) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --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] Limiting a group of phones available channels
Snip On Fri, 2006-06-30 at 10:44 +0100, Steve Langstaff wrote: trixter aka Bret McDanel wrote: Lastly, and probably the least effective, is you can watch channel usage and when someone exceeds 5 run over to their desk and smack them with a rotten fish. http://www.voip-info.org/wiki-Asterisk+sip+incominglimit I can't find the 'rotten fish' stuff documented anywhere on voip- info.org - was that some sort of red herring? Its an advanced asterisk management option. Snip. The rotten fish AMO (Asterisk Management Option) had too many side effects, we are in Miami and the weather makes that particular AMO too perishable. We have since sub-contracted with the Soprano Family, at this point we have 100% Available channels, and users love the quality of the phones, they have found a new love for the Fisher Price/Sponge Bob SIP phones we installed last year, These guys are great they got complaints down to zero!!! Like the salesman Tony told me 'If it wasn't for employees, running a company would be fun.' The above is a parody and if you can't take the joke, laugh first then delete!! ___ --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] Surge Protector for T1/PRI ?
I have used these in the past, with only one issue. The T1 line was at the end of its tolerances as far as length from the repeater. The surge suppressor ntroduced enough resistance to make the T1 bounce, like Tigger. Having the Telco put in a repeater closer to our facility made the problem go away. Something to remember about lightning, it is lazier than a teenager with cableTV. It will always find the shortest (least resistive) path to ground. Today that may be your T1 card. Tomorrow it may be your Ethernet network. Look at placing one of these on your T1's as well as your Ethernet network, and Power Supply. I am in Florida and lightning is a way of life. You can protect yourself, you just have to think like a lightning bolt for a while. Alex Snip ___ --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] Integrate asterisk with Database
: Hi All, : : I am plainging to give a solutions for a sports club. Follwing : is the process that i need to achieve. : If any body achieve this kind of setup pls give me a feedback, so : that i can go through. Have you even _tried_ to create your dialplan? m. ___ --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] ISDN: 3° incoming call
Julian J. M. wrote: BRI ISDN is 2 channels, what would you want to do with a 3rd call? Julian On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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 I would to know when I'm using the second channel: there is some check method that I can do? So I can play a message for an incoming 3rd call or turn it in an other channel. For example, if I have all two channel busy, for an outgoing call I can use a SIP or IAX channel, or an other ISDN line. Thanks ___ --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] New Digium Card b410p
Who knows something interesting about the new BRI digium card b410p ? For example, will it use the misdn driver or the native zaptel? Any interesting links will be appreciated too. ___ --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] (no subject)
Dear I am using trixbox,I want ot disable and enable voicemail from command line At [EMAIL PROTECTED] v 2.8 I was using this command and was working successfully Database put AMPUSER/9990999 voicemail default And Database put AMPUSER.9990999 voicemail disables But at trixbox its not working Any ideas pleas Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Voicemail
Dear I am using trixbox,I want ot disable and enable voicemail from command line At [EMAIL PROTECTED] v 2.8 I was using this command and was working successfully Database put AMPUSER/9990999 voicemail default And Database put AMPUSER.9990999 voicemail disables But at trixbox its not working Any ideas pleas Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] directory
How can I isolate directory address book search *411 depending on context since context A user don't search at context B users regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Voicemail
Because probably the rows/table/database name changed. Connect to you mysql database and find what records you have to modify. m. - Original Message - From: Khaled Chehab To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Sent: Sunday, July 30, 2006 2:43 PM Subject: [Asterisk-Users] Voicemail Dear I am using trixbox,I want ot disable and enable voicemail from command line At [EMAIL PROTECTED] v 2.8 I was using this command and was working successfully Database put AMPUSER/9990999 voicemail default And Database put AMPUSER.9990999 voicemail disables But at trixbox its not working Any ideas pleas Regards *No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.* ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Problems with dial status...
On a zap channel, Asterisk can't tell when a call has been answered, so starts the playback immediately. Setup a loop asking the caller to press a key. I have the following setup: [..] I'm still wondering how to do it and I thought about BackgroundDetect(). Is there any way to use it to detect non-silence and non-beeps (calling signal) but speech? Martin ___ --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] Does anyone know what this means?
== Spawn extension (intqueue, 1004, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' Jun 30 15:18:34 WARNING[13523]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x81fe3f8', 10 retries! -- Stopped music on hold on Zap/2-1 ___ --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] New Digium Card b410p
I've contact Digium, and they told me they were finalizing the driver and so on. And all the info would soon be posted at digium's website. In fact it was supposed to be ready one week ago... At least they told me that. On 6/30/06, Tommaso Calosi [EMAIL PROTECTED] wrote: Who knows something interesting about the new BRI digium card b410p ? For example, will it use the misdn driver or the native zaptel? Any interesting links will be appreciated too. ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] ISDN: 3° incoming call
You should handle correctly Dial(...) return value in your dial plan, then playback(your busy channel msg) and then dial through IAX or SIP or whatever you want. If you use Freepbx would be easy to learn how to write your Dialplan Script... On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Julian J. M. wrote: BRI ISDN is 2 channels, what would you want to do with a 3rd call? Julian On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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 I would to know when I'm using the second channel: there is some check method that I can do? So I can play a message for an incoming 3rd call or turn it in an other channel. For example, if I have all two channel busy, for an outgoing call I can use a SIP or IAX channel, or an other ISDN line. Thanks ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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: New Digium Card b410p
Tommaso Calosi wrote: Who knows something interesting about the new BRI digium card b410p ? For example, will it use the misdn driver or the native zaptel? Any interesting links will be appreciated too. ___ --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 The new 8 channel BRI card uses mISDN. According to Digium the hardware is finalised and they are currently beta testing the drivers. I was talking to Matt, one of the Digium developers that has been working on the card, so this is all first hand information rather than rumour or hear-say. Should be available worldwide through Digium's normal distribution channels in the next few weeks. Like buses (so we say in the UK), decent BRI hardware comes all at once. Xorcom are just about to release BRI versions of their Asterisk specific channel banks as well. Best regards. David Cook ___ --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] ISDN: 3° incoming call
Marco Mouta wrote: You should handle correctly Dial(...) return value in your dial plan, then playback(your busy channel msg) and then dial through IAX or SIP or whatever you want. If you use Freepbx would be easy to learn how to write your Dialplan Script... On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Julian J. M. wrote: BRI ISDN is 2 channels, what would you want to do with a 3rd call? Julian On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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 I would to know when I'm using the second channel: there is some check method that I can do? So I can play a message for an incoming 3rd call or turn it in an other channel. For example, if I have all two channel busy, for an outgoing call I can use a SIP or IAX channel, or an other ISDN line. Thanks ___ --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 For an outgoing call I agree with your suggestion, but for an incoming call (witch i manage in remote context) how can I make this control? In this case I don't have a Dial return value to handle. Thanks a lot ___ --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] TE420P/TE415P?
On Jun 27, 2006, at 4:25 AM, Rob Lith wrote: On 25/06/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: 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, does the card include the licence for the codecs? Otherwise the card at +- $1994 SRP + codecs is quite expensive? Yes it does. The cost of the card includes the cost of the codec licenses. Matthew Fredrickson ___ --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] TDM400P bad echo problem, tried lots of things
Hi All, Also check that TDM400 not share interrups (yes, it sounds silly, but in some cases it were the answer for me). Fabio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Matthew Fredrickson Enviado el: Jueves, 29 de Junio de 2006 09:41 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] TDM400P bad echo problem, tried lots of things Try fxotune. That's the first thing you should have used. Matthew Fredrickson On Jun 20, 2006, at 11:55 AM, Carey O'Shea wrote: I have a bad echo problem on my TDM400P with one FXO module installed. I have tried a few things, such as: * setting rxgain and txgain to 0 * setting echocancelwhenbridged to no / yes * settting echocancel to 64 / no / yes * setting echocanceltraining to 800 / no / yes * MG2 echo cancellation * MARK2 echo cancellation * KB1 echo cancellation * AGGRESSIVE_SUPPRESSOR option of MARK2 Each time restarting Asterisk, then opening the Zap channel, and then speaking...only to hear my self played back almost instantly. None of these options changed the echo for me, it always sounded the same -- except for the AGGRESSIVE_SUPPRESSOR option, in which every time I spoke it made the other end a very low volume, so much that I couldn't hear the other end (ie: not useful). I don't have this problem with pure IP calls, it's only with my TDM400P and FXO that I have this echo problem. This means my headset and IP phones are fine (of course). So, what else can I try? :-) Any ideas why this is so consistent and persistent? Maybe it's something to do with my phone cable or something of that nature (hmm?)? Any input appreciated. Thanks, Carey O'Shea. ___ --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 ___ --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] ISDN: 3° incoming call
Incoming you mean arrivin from a SIP trunk or from ISDN? BRI card will be busy if you have already 2 calls running, so the caller party should get busy indication from your Telco... On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Marco Mouta wrote: You should handle correctly Dial(...) return value in your dial plan, then playback(your busy channel msg) and then dial through IAX or SIP or whatever you want. If you use Freepbx would be easy to learn how to write your Dialplan Script... On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Julian J. M. wrote: BRI ISDN is 2 channels, what would you want to do with a 3rd call? Julian On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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 I would to know when I'm using the second channel: there is some check method that I can do? So I can play a message for an incoming 3rd call or turn it in an other channel. For example, if I have all two channel busy, for an outgoing call I can use a SIP or IAX channel, or an other ISDN line. Thanks ___ --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 For an outgoing call I agree with your suggestion, but for an incoming call (witch i manage in remote context) how can I make this control? In this case I don't have a Dial return value to handle. Thanks a lot ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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: TE420P/TE415P?
I assume that it would be 30 licenses, so you could fully use the card as E1. Is this correct? Can asterisk use these licenses for other calls as well? (sip G.729 to voicemail) -- -- Steven http://www.glimasoutheast.org Matthew Fredrickson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Jun 27, 2006, at 4:25 AM, Rob Lith wrote: On 25/06/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: 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, does the card include the licence for the codecs? Otherwise the card at +- $1994 SRP + codecs is quite expensive? Yes it does. The cost of the card includes the cost of the codec licenses. Matthew Fredrickson ___ --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] ISDN: 3° incoming call
On Fri, 30 Jun 2006, Marco Mouta wrote: Incoming you mean arrivin from a SIP trunk or from ISDN? BRI card will be busy if you have already 2 calls running, so the caller party should get busy indication from your Telco... No, the third call is signaled as call-waiting without attached to a b-channel. With chan-capi you can do actions in that case via the extentions.conf, like Busy() or deflect this call to another number. Armin On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Marco Mouta wrote: You should handle correctly Dial(...) return value in your dial plan, then playback(your busy channel msg) and then dial through IAX or SIP or whatever you want. If you use Freepbx would be easy to learn how to write your Dialplan Script... On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Julian J. M. wrote: BRI ISDN is 2 channels, what would you want to do with a 3rd call? Julian On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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 I would to know when I'm using the second channel: there is some check method that I can do? So I can play a message for an incoming 3rd call or turn it in an other channel. For example, if I have all two channel busy, for an outgoing call I can use a SIP or IAX channel, or an other ISDN line. Thanks ___ --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 For an outgoing call I agree with your suggestion, but for an incoming call (witch i manage in remote context) how can I make this control? In this case I don't have a Dial return value to handle. Thanks a lot ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] Recommended FXO device
Rich Adamson wrote: I've tested a large number of other external adapters and have not found a single one that had a reasonable echo canceller built in. Many of them work fine on short pstn lines (where echo is much less of a problem), but provided even reasonable service on longer pstn lines or lines that involve unusual telco configurations (eg, remote line concentrators). What about devices from audiocodes, ipgear/boscom and vegastream? Can you give a list of products you have tested and your results as well as your testing environment and methodology? ___ --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 SIP Registrations
Doug, If you'd be willing to share the patch and AGI, I would be happy to help test your solution. I know that myself and several others have been looking for a way to make Asterisk do this for quite some time. regards, David On 6/29/06, Doug G [EMAIL PROTECTED] wrote: Well, to dial a peer direclty the only thing that is missing in realtime is the status of the sip peer. (registered, Unregistered, unknown, reachable). If you dial a peer via ip and it is unavaliable you get dead air. So you need to know the status of the peer before dialing it. The change basicly updates realtime with the peers status. I did the same thing for IAX as well.. Doug From: [EMAIL PROTECTED] on behalf of Mike Lynchfield Sent: Thu 6/29/2006 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations can you elaborate on modify sip to update the status on the sip friends in realtime thanks On 6/29/06, Doug G [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: What I did was modify sip to update the status on the sip friends in realtime. Then via FAGI dial them directly with the data found in real-time. (ie dial ( SIP/[EMAIL PROTECTED]:5060) Of course you need to check the status in realtime data before you dial. This allows MANY Asterisk servers to share the same SIP data.I then load balance with DNS SRV.. Yes I have tested in failover it works. I too have been told that by many that this will not work. So I keep expecting to hit some problem with it, but to date I have not... Doug From: [EMAIL PROTECTED] on behalf of David Thomas Sent: Thu 6/29/2006 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations I think lots of us know about it... We're just not sure how to go about fixing it. :-( I know it's been a thorn in my side since I started using Asterisk. I would suspect that many of those saying works for me have never actually tested their system in failure scenarios, or they are working in a controlled environment without NAT and such... regards, David On 6/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Aaron Daniel [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Thursday, June 29, 2006 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime SIP Registrations On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.) 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 ___ --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 -- Mike Sales Manager http://www.theclubvoip.com Making it happen 1.888.470.7253 ___ --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] ISDN: 3° incoming call
Armin Schindler wrote: On Fri, 30 Jun 2006, Marco Mouta wrote: Incoming you mean arrivin from a SIP trunk or from ISDN? BRI card will be busy if you have already 2 calls running, so the caller party should get busy indication from your Telco... No, the third call is signaled as call-waiting without attached to a b-channel. With chan-capi you can do actions in that case via the extentions.conf, like Busy() or deflect this call to another number. I'm using chan_modem[i4l]: what actions can I do with this? thanks Armin On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Marco Mouta wrote: You should handle correctly Dial(...) return value in your dial plan, then playback(your busy channel msg) and then dial through IAX or SIP or whatever you want. If you use Freepbx would be easy to learn how to write your Dialplan Script... On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Julian J. M. wrote: BRI ISDN is 2 channels, what would you want to do with a 3rd call? Julian On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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 I would to know when I'm using the second channel: there is some check method that I can do? So I can play a message for an incoming 3rd call or turn it in an other channel. For example, if I have all two channel busy, for an outgoing call I can use a SIP or IAX channel, or an other ISDN line. Thanks ___ --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 For an outgoing call I agree with your suggestion, but for an incoming call (witch i manage in remote context) how can I make this control? In this case I don't have a Dial return value to handle. Thanks a lot ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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 ___ --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] recording all calls patch through asterisk
Basically I will have a call come in a PRI trunk and be routed out the same PRI trunk. The point of this is so I can use asterisk to record the call. Has anyone set up a system like this? I know how to get asterisk to record a call from and extension, but not a call that is just passing through the system. -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --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] SIP reinvite still does not occour
On Fri, 2006-06-30 at 10:39 +0200, Roger Schreiter wrote: Hoa Thai Duy schrieb: Roger If transcoding happened (from G.729 to GSM, any to any) then Asterisk will issue no re-INVITE, for sure. Pls. change Disallow=all Allow=gsm (only one codec) Hi, yes, to avoid transcoding problems I only have one codec, just alaw. Anything else is disallowed. That's why I don't understand, why there is no reinvite. Thanks for answering! Iirc if you have something like a t or T in your Dial command in extensions.conf than canreinvite will not work because Asterisk has to stay in the middle to take care of the t or T. Remove these (and maybe othger) options from the Dial command and give it a try again. Regards, Patrick ___ --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
Octasic for TDM2400P and TDM400P? was: [Asterisk-Users] TE420P/TE415P?
When will Digium include the octasic on the TDM2400P? And maybe the TDM400P? Also how does the TE415P and TE420P differ from the TE412P card? ___ --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 SIP Registrations
I'm intensely curious why it doesn't currently work. You have multiple Asterisk systems, all referring to a common table for SIP peer information. The fact that there is multiple Asterisk systems accessing the same MySQL data should be completely transparent to each of them, and I don't understand why this doesn't work. Anyone? Doug. -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Friday, June 30, 2006 9:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations Doug, If you'd be willing to share the patch and AGI, I would be happy to help test your solution. I know that myself and several others have been looking for a way to make Asterisk do this for quite some time. regards, David On 6/29/06, Doug G [EMAIL PROTECTED] wrote: Well, to dial a peer direclty the only thing that is missing in realtime is the status of the sip peer. (registered, Unregistered, unknown, reachable). If you dial a peer via ip and it is unavaliable you get dead air. So you need to know the status of the peer before dialing it. The change basicly updates realtime with the peers status. I did the same thing for IAX as well.. Doug From: [EMAIL PROTECTED] on behalf of Mike Lynchfield Sent: Thu 6/29/2006 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations can you elaborate on modify sip to update the status on the sip friends in realtime thanks On 6/29/06, Doug G [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: What I did was modify sip to update the status on the sip friends in realtime. Then via FAGI dial them directly with the data found in real-time. (ie dial ( SIP/[EMAIL PROTECTED]:5060) Of course you need to check the status in realtime data before you dial. This allows MANY Asterisk servers to share the same SIP data.I then load balance with DNS SRV.. Yes I have tested in failover it works. I too have been told that by many that this will not work. So I keep expecting to hit some problem with it, but to date I have not... Doug From: [EMAIL PROTECTED] on behalf of David Thomas Sent: Thu 6/29/2006 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations I think lots of us know about it... We're just not sure how to go about fixing it. :-( I know it's been a thorn in my side since I started using Asterisk. I would suspect that many of those saying works for me have never actually tested their system in failure scenarios, or they are working in a controlled environment without NAT and such... regards, David On 6/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Aaron Daniel [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Thursday, June 29, 2006 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime SIP Registrations On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.) 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 ___ --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 -- Mike Sales Manager http://www.theclubvoip.com Making it
Re: [Asterisk-Users] ISDN: 3° incoming call
On Fri, 30 Jun 2006, francesco giuliani wrote: Armin Schindler wrote: On Fri, 30 Jun 2006, Marco Mouta wrote: Incoming you mean arrivin from a SIP trunk or from ISDN? BRI card will be busy if you have already 2 calls running, so the caller party should get busy indication from your Telco... No, the third call is signaled as call-waiting without attached to a b-channel. With chan-capi you can do actions in that case via the extentions.conf, like Busy() or deflect this call to another number. I'm using chan_modem[i4l]: what actions can I do with this? I4L itself has nothing to do with that. The low-level driver reports the amount of channels to I4L. And I don't know any I4L low-level BRI driver which handles/reports more than two channels. So I think you don't have any action available. Armin On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Marco Mouta wrote: You should handle correctly Dial(...) return value in your dial plan, then playback(your busy channel msg) and then dial through IAX or SIP or whatever you want. If you use Freepbx would be easy to learn how to write your Dialplan Script... On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Julian J. M. wrote: BRI ISDN is 2 channels, what would you want to do with a 3rd call? Julian On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote: Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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 I would to know when I'm using the second channel: there is some check method that I can do? So I can play a message for an incoming 3rd call or turn it in an other channel. For example, if I have all two channel busy, for an outgoing call I can use a SIP or IAX channel, or an other ISDN line. Thanks ___ --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 For an outgoing call I agree with your suggestion, but for an incoming call (witch i manage in remote context) how can I make this control? In this case I don't have a Dial return value to handle. Thanks a lot ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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 ___ --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] Asterisk -x option in 1.2.9.1
This really looks like a bug. It seems as though the '-x' option is broken as of 1.2.9.1 Sometimes the output of the -x command will be only a single line: hestia:(pbx1)~ # asterisk -rx 'database show' //Agents/80014054 : [EMAIL PROTECTED];80014054 and sometimes it will display many or all lines. A buffering issue of some sort maybe? 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
RE: [Asterisk-Users] cheapest Cisco Smartnet contract?
Email me off list with the phone part numbers, and I'll see what I can do.. It probably depends on the level of cisco certification the company has. I dont know if we can do better, but I'll see! Steve [EMAIL PROTECTED] From: Louis-David Mitterrand [mailto:[EMAIL PROTECTED] Sent: Fri 6/30/2006 4:30 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cheapest Cisco Smartnet contract? Hello, I've got a few Cisco phones to maintain and need access to firmware files. Dealers here in .fr want unreasonable prices for a Smartnet subscription. Where can I get a better deal on the Net? Thanks, winmail.dat___ --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] asterisk to mobile phone
what brand of gsm gateway do you think works well with asterisk? voismart.it - quadgsm -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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] Cannot get back chan_zap.so module!??
Hey list! I keep getting the error: "Unable to create channel of type 'Zap' (cause 66 - Channel not implemented)" error. In looking on my filesystem, I seemed to have "lost" the chan_zap.so module from /usr/lib/asterisk/modules. I've re-compiled Zaptel and Asterisk, but it doesn't show up. Zaptel: # make clean # make linux26 # make install This is good. I've modprobe'd the cards, and everything comes up: # lsmod | grep zaptel zaptel 196740 1 wcte11xp crc_ccitt6081 2 zaptel,hisax So, I then re-compiled asterisk, so it can build the chan_zap.so: # make clean # make make install But the chan_zap.so module never gets built. What could I be missing? Thanks! ~~Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk x Qsig - messages
Hi All. Somebody already caught the messages below? -- Executing Dial(SIP/3347-9360, zap/g1/3384|60) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/3384 -- Zap/1-1 is proceeding passing it to SIP/3347-9360 -- Zap/1-1 is ringing!! Not yet handling pre-handle message type SEGMENT (96)!! Don't know how to post-handle message type SEGMENT (96) !! Not yet handling pre-handle message type SEGMENT (96)!! Don't know how to post-handle message type SEGMENT (96)!! Not yet handling pre-handle message type SEGMENT (96)!! Don't know how to post-handle message type SEGMENT (96) !! Not yet handling pre-handle message type SEGMENT (96)!! Don't know how to post-handle message type SEGMENT (96) -- Zap/1-1 answered SIP/3347-9360 -- Hungup 'Zap/1-1' == Spawn extension (default, 3384, 1) exited non-zero on 'SIP/3347-9360' -- Executing Dial(SIP/3347-4c61, zap/g1/3384|60) in new stack The call is completed, but it does not have audio, is dumb.Best Regards Josué ___ --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: Digium Hardware Reliability
Andrew Kohlsmith wrote: On Thursday 29 June 2006 21:38, M.Hockings wrote: How reliable is Digium hardware in general.? My new TDM400P just died. I have a number of Digium T1 products (T100P, TE410P, TE405P and TE406P) as well as a few TDM400 based boards. No failures in the last 2 years or so. So, at over 2x the cost is Sangoma hardware more sturdy than the Digium stuff? Not that I've seen. I also have a number of Sangoma products. Both work very well for me. As an engineer, I can also see that the protection on the interfaces is comparable. Mike (totally UNimpressed with Digium) I don't think this is a Digium problem, at least not yet. What did their customer service people say? Can you ask for a failure report? You note that power went out. Generally when this occurs there is a very high chance of transient voltage spiking or line swells not only on the residential electrical power grid but also on the telephone network. Do you have any telco line protection in place to protect the card from nasties coming in from the outside? Is the protection correctly installed? How about electrical protection? The MOVs in your power strip and UPS are only good for a few hits before they become ineffective (something they never tell you). Unless you know something more than you've presented here it is a little premature to start pointing fingers. -A. Point taken. I was not so much point fingers but asking what my expectation should be and maybe shedding some frustration. I don't really have a lot of experience with this kind of communications gear and it could very well be that one should keep spare daughter boards in stock. I was finally able to get the thing going again but I do not know what I did to accomplish that. I had tried the card in different PCI slots, reseated the daughter cards, powered the machine with and without the card, checked BIOS settings then after half a day of fiddling it just started responding again. Who knows what the problem was? As far as heat and stuff go, the card is in the only card in a new IBM/Lenovo box and has plenty of air on all sides. The box itself is powered by an AVR type UPS, which according to the graphs it shows is keeping the power pretty stable even though dips. One weakness is the incoming PSTN line, what is the best way to protect that beyond the device at the premises entry ? So now it appears to be working again, don't know what failed, don't know what made it work. and afraid of the next power outage at this rural SOHO. Mike ___ --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] Switchtype
Our PRI vendor is using a Nortel DMS500 switch. Which switch type should I use. I have been using national but we are having issues with our connectivity. national dms100 4ess 5ess euroisdn ni1 qsig Thank You James Hawks ___ --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] Auto NOTIFY
Hey everyone, I wrote in last week about our Polycom phones rebooting. I had a nice theory with it being the PoE switch but that was thrown out the window today when phones even with a power supply rebooted. So my question now points back to Asterisk. Is there any feature on Asterisk that sends a NOTIFY signal to the phones that is automatically enabled? Or is it only manual? Thanks, Kevin ___ --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] recording all calls patch through asterisk
Michael Sampson wrote:
Basically I will have a call come in a PRI trunk and be routed out the
same PRI trunk. The point of this is so I can use asterisk to record the
call. Has anyone set up a system like this? I know how to get asterisk
to record a call from and extension, but not a call that is just
passing through the system.
I'm assuming the call comes in through one PRI line (Zap group 1), and then goes
out again via another PRI line (Zap group 2) into some other device.
[incoming]
exten = _X.,1,MixMonitor(${UNIQUEID}.gsm))
exten = _X.,2,Dial(Zap/g2/${EXTEN})
[outgoing]
exten = _X.,1,MixMonitor(${UNIQUEID}.gsm)
exten = _X.,2,Dial(Zap/g1/${EXTEN})
make sure to set Zap group 1 to the incoming context and set zap group 2 to
the outgoing context
Flynn
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Re: [Asterisk-Users] Re: Digium Hardware Reliability
M.Hockings wrote: Mike (totally UNimpressed with Digium) Point taken. I was not so much point fingers but asking what my expectation should be and maybe shedding some frustration. I don't really have a lot of experience with this kind of communications gear All the more reason for you to fully inform yourself *first*, and then start posting negative drivel to a public mailing list. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Auto NOTIFY
The following command on the Asterisk console will reboot a polycom phone: sip notify polycom-check-cfg ip-addr but in sip.conf, voIpProt.SIP.specialEvent.checkSync.alwaysReboot needs to be set to 1. otherwise... beats the heck out of me! -Original Message- From: Kevin Smith [mailto:[EMAIL PROTECTED] Sent: Friday, June 30, 2006 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Auto NOTIFY Hey everyone, I wrote in last week about our Polycom phones rebooting. I had a nice theory with it being the PoE switch but that was thrown out the window today when phones even with a power supply rebooted. So my question now points back to Asterisk. Is there any feature on Asterisk that sends a NOTIFY signal to the phones that is automatically enabled? Or is it only manual? Thanks, Kevin ___ --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] Switchtype
I would work that out with your vendor, as the settings must be the same on both sides. If national won't work for you, ask them if they can change to something else. What kinds of connectivity issues? Could be line problems too. - Original Message - From: James Hawks To: asterisk-users@lists.digium.com Sent: Friday, June 30, 2006 2:45 PM Subject: [Asterisk-Users] Switchtype Our PRI vendor is using a Nortel DMS500 switch. Which switch type should I use. I have been using national but we are having issues with our connectivity. national dms100 4ess 5ess euroisdn ni1 qsig Thank You James Hawks ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Digium Hardware Reliability
Andrew,you seem to be assuming a lot. These were spread out across different parts of the country (US), on projects I was involved with but deployed by more than compentent telco and engineering colleagues of mine. And ... in the majority of the cases, they were DOA (not a transient issue, noisy line or not). The warranty is there and Digium or their resellers make good - but the delays in the project and the lossed time are still real. Once working, they do seem to continue working fine.So ... don't try to read too much into it. That is why I am very interested in seeing what others are finding.pFrom: Andrew Kohlsmith [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Fri, 30 Jun 2006 08:49:07 -0400Subject: Re: [Asterisk-Users] Digium Hardware Reliability On Friday 30 June 2006 02:24, Philippe Lindheimer wrote: I would love to see some feedback on this as well. I've lost exact count now, but think I've seen about 5-6 failures on their cards TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with thatThen put proper telco line protection in place! Good lord, it's blindingly obvious to me that you seem to be in a particularly harsh environment and that the protection on the FXO modules was not designed for the type of transient disturbances you're experiencing.-A. Yahoo! Music Unlimited - Access over 1 million songs. Try it free. ___ --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] Auto answer an IAXY how
Can an IAXY be setup to auto answer? If so how? I mean any call coming into it automatically connect it to the phone and send voice traffic. Thanks, Jerry ___ --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] Auto NOTIFY
Hey Doug, That's what I figured, but correct me if I am wrong. Isn't 1 will always set the phones to reboot on a NOTIFY command regardless of any changes in the configuration file? I thought 0 would means it requires both a notify request and a change in the configuration file. But you are right, I'm out of ideas. Seeing today one phone reboot with a power supply really threw me for a loop. Thanks, Kevin Douglas Garstang wrote: The following command on the Asterisk console will reboot a polycom phone: sip notify polycom-check-cfg ip-addr but in sip.conf, voIpProt.SIP.specialEvent.checkSync.alwaysReboot needs to be set to 1. otherwise... beats the heck out of me! -Original Message- From: Kevin Smith [mailto:[EMAIL PROTECTED] Sent: Friday, June 30, 2006 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Auto NOTIFY Hey everyone, I wrote in last week about our Polycom phones rebooting. I had a nice theory with it being the PoE switch but that was thrown out the window today when phones even with a power supply rebooted. So my question now points back to Asterisk. Is there any feature on Asterisk that sends a NOTIFY signal to the phones that is automatically enabled? Or is it only manual? Thanks, Kevin ___ --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 ___ --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] Voicemail
Khaled Chehab wrote: I am using trixbox,I want ot disable and enable voicemail from command line At [EMAIL PROTECTED] v 2.8 I was using this command and was working successfully snip But at trixbox its not working Any ideas pleas Did you try checking with the people who _wrote_ trixbox? Perhaps they have a forum or at least mailing list of some sort that could answer your question(s)? Flynn ___ --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] IAX jitter / clocking problem
thanks Dough, seems, that you mix options for old and new jitterbuffer implementation (according to iax.conf.sample), I think, that now is by default in compile time selected new jitterbuffer, so only these four options are in efect and rest are ignored PJ new jitterbuffer options: jitterbuffer=yes maxjitterbuffer=1000 maxjitterinterps=10 resyncthreshold=1000 [This option is not applicable to, and ignored by the new jitterbuffer implementation] dropcount maxexcessbuffer minexcessbuffer jittershrinkrate Doug Lytle wrote: Pavel Jezek wrote: hello, I'm still trying to tune iax jitterbuffer in asterisk 1.2.9.1, but without success, Here is my entries: jitterbuffer=yes dropcount=3 maxjitterbuffer=1000 maxjitterinterps=10 maxexcessbuffer=80 resyncthreshold=1000 minexcessbuffer=10 jittershrinkrate=1 ___ --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] trunk rollover
Jon Scottorn wrote:
What kind of line is being used?
in zapata.conf:
group = 1
channel = 1,3,5,6
I create a zap group will all your lines and dial out using the zap
group ie...
Dial(Zap/g1/${EXTEN})
By using the group it dials on the first available line.
If you want a more complex setup I have that as well.
I have an agi script that looks at the number dialed and determins if
it is a local call if so, dial out the ZAP line, if all ZAP lines are
busy dial out an IAX provider, I all IAX lines are busy, then roll to
my SIP provider.
Took a bit to figure it all out and get working but it is very useful.
Jon
Hi Jon,
Thanks, One of the lines is a sip connection to Telasip, the other is a
ZAP line. I'd appreciate any help I could get. I don't know what an
agi script is, so be gentle. :)
Thanks,
Jim.
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RE: [Asterisk-Users] Digium Hardware Reliability
Im working on quantifying an overall defect rate for both Digium and Sangoma products, based upon overall number of units deployed over a 12 month period versus overall number of units RMA replaced. I believe both products to have very low DOA rates, well below acceptable industry standards for electronic components, but the data will tell the story. More to come shortly. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer Sent: Friday, June 30, 2006 3:04 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Digium Hardware Reliability Andrew, you seem to be assuming a lot. These were spread out across different parts of the country (US), on projects I was involved with but deployed by more than compentent telco and engineering colleagues of mine. And ... in the majority of the cases, they were DOA (not a transient issue, noisy line or not). The warranty is there and Digium or their resellers make good - but the delays in the project and the lossed time are still real. Once working, they do seem to continue working fine. So ... don't try to read too much into it. That is why I am very interested in seeing what others are finding. p From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 30 Jun 2006 08:49:07 -0400 Subject: Re: [Asterisk-Users] Digium Hardware Reliability On Friday 30 June 2006 02:24, Philippe Lindheimer wrote: I would love to see some feedback on this as well. I've lost exact count now, but think I've seen about 5-6 failures on their cards TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with that Then put proper telco line protection in place! Good lord, it's blindingly obvious to me that you seem to be in a particularly harsh environment and that the protection on the FXO modules was not designed for the type of transient disturbances you're experiencing. -A. Yahoo! Music Unlimited - Access over 1 million songs. Try it free. ___ --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] Auto answer an IAXY how
I see on http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels that an option 'a' is available meaning 'request autoanswer'. Never tested this before, so please do. Another possibility might be setting immediate=yes in iax.conf for the iaxy? just a guess. Moj Jerry Geis wrote: Can an IAXY be setup to auto answer? If so how? I mean any call coming into it automatically connect it to the phone and send voice traffic. Thanks, Jerry ___ --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 !DSPAM:500,44a5778f215925167217508! -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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] Auto answer an IAXY how
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 It would not be the iaxy... it would be the phone that is attached to it... there are plenty of phones/answering machines /other FXS signalling devices that can do auto answer... the iaxy is not capable of doing that... Sean Jerry Geis wrote: Can an IAXY be setup to auto answer? If so how? I mean any call coming into it automatically connect it to the phone and send voice traffic. Thanks, Jerry ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEpXvo1Kolm8VQlAURAt5BAJ91hIBpkCABT5buMVqiau5K61pL2ACfYLwG WCp55L0L4OHM64pASfWJCgg= =frDI -END PGP SIGNATURE- ___ --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] Cannot get back chan_zap.so module!??
I get the chan_zap.so if I recompile under asterisk-1.2.7.1, but not under subversion TRUNK Anyone able to do this? - Original Message - From: Aaron Paxson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, June 30, 2006 1:44 PM Subject: [Asterisk-Users] Cannot get back chan_zap.so module!?? Hey list! I keep getting the error: "Unable to create channel of type 'Zap' (cause 66 - Channel not implemented)" error. In looking on my filesystem, I seemed to have "lost" the chan_zap.so module from /usr/lib/asterisk/modules. I've re-compiled Zaptel and Asterisk, but it doesn't show up. Zaptel: # make clean # make linux26 # make install This is good. I've modprobe'd the cards, and everything comes up: # lsmod | grep zaptel zaptel 196740 1 wcte11xp crc_ccitt6081 2 zaptel,hisax So, I then re-compiled asterisk, so it can build the chan_zap.so: # make clean # make make install But the chan_zap.so module never gets built. What could I be missing? Thanks! ~~Aaron ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Auto answer an IAXY how
I see on http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels that an option 'a' is available meaning 'request autoanswer'. Never tested this before, so please do. Another possibility might be setting immediate=yes in iax.conf for the iaxy? just a guess. Moj I tried both of those just now and it did not work. I am trying to use the IAXY to connect to an analog intercom system. I can put a normal analog line on the intercom system, pick up the phone (off hook), and select my zone and talk. I want to do this with an IAXY. So when I call into the IAXY it comes off hook and I would be connect to the intercom. Thanks for any other suggestions. Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Hardware Reliability
To get an accurate portrayal of defect rate, a very large sample size will obviously result in a more accurate calculation. I calculated a defect rate of between 1-2% for Digium products, based on an arbitrary sample size of 5000 units. These included ALL Digium products, not just TDM products. This does not account for shipping mishandling, or onsite mishandling leading to failure. Excluding those factors, Id offer an educated assessment of around 1% DOA/Failure rate. Waiting on Sangoma data which is likely about the same. I cant find an industry standard defect rate for general electronic components, but 1% seems pretty low. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Friday, June 30, 2006 3:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Digium Hardware Reliability Im working on quantifying an overall defect rate for both Digium and Sangoma products, based upon overall number of units deployed over a 12 month period versus overall number of units RMA replaced. I believe both products to have very low DOA rates, well below acceptable industry standards for electronic components, but the data will tell the story. More to come shortly. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer Sent: Friday, June 30, 2006 3:04 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Digium Hardware Reliability Andrew, you seem to be assuming a lot. These were spread out across different parts of the country (US), on projects I was involved with but deployed by more than compentent telco and engineering colleagues of mine. And ... in the majority of the cases, they were DOA (not a transient issue, noisy line or not). The warranty is there and Digium or their resellers make good - but the delays in the project and the lossed time are still real. Once working, they do seem to continue working fine. So ... don't try to read too much into it. That is why I am very interested in seeing what others are finding. p From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 30 Jun 2006 08:49:07 -0400 Subject: Re: [Asterisk-Users] Digium Hardware Reliability On Friday 30 June 2006 02:24, Philippe Lindheimer wrote: I would love to see some feedback on this as well. I've lost exact count now, but think I've seen about 5-6 failures on their cards TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with that Then put proper telco line protection in place! Good lord, it's blindingly obvious to me that you seem to be in a particularly harsh environment and that the protection on the FXO modules was not designed for the type of transient disturbances you're experiencing. -A. Yahoo! Music Unlimited - Access over 1 million songs. Try it free. ___ --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] Auto answer an IAXY how
Ah, the problem is that you are connecting FXO to FXO. The IAXy provides dialtone and o does your Intercom system. You can try to use an FXO to FXS converter or simply replace it with an FXO adapter. I would also check the documentation on your intercom device. There may be a way to switch the port type around. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Friday, June 30, 2006 3:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Auto answer an IAXY how I see on http://www.voip- info.org/wiki/index.php?page=Asterisk+IAX+channels that an option 'a' is available meaning 'request autoanswer'. Never tested this before, so please do. Another possibility might be setting immediate=yes in iax.conf for the iaxy? just a guess. Moj I tried both of those just now and it did not work. I am trying to use the IAXY to connect to an analog intercom system. I can put a normal analog line on the intercom system, pick up the phone (off hook), and select my zone and talk. I want to do this with an IAXY. So when I call into the IAXY it comes off hook and I would be connect to the intercom. Thanks for any other suggestions. Jerry ___ --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] SIP qualify time - best practices?
For the typical home user who has a SIP ATA behind (usually) a Linksys home router/firewall, what's the best practice qualify= time we should be running on the server, to keep the home user's NAT happy? The default, 2 seconds, is way too short (generates too much net traffic). I am wondering how high we can go, and still make the majority of our customers' home nat's happy. 1 minute? 2 minutes? 10 minutes? Thanks for opinions, Bryan ___ --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] SOLVED: IAX jitter / clocking problem
I found my mistake jiterbuffer=yes vs. jitterbuffer=yes ;-) currently I have this settings, and seems this working quite well, only sometimes gaps appears, when jitter changes too much eg. 500ms - jitterbuffer probably can't adapt so quick, maybe good idea to set some minimum jitterbuffer value, but this is not possible in current new jitterbuffer implemenation PJ jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=1500 maxjitterinterps=10 resyncthreshold=2000 Pavel Jezek wrote: hello, I'm still trying to tune iax jitterbuffer in asterisk 1.2.9.1, but without success, I'm using idefisk-asterisk over cdma network, where rtt is about 100-500ms, so jitter about 400ms but sound is very jerky, in diection idefisk-asterisk, in reverse direction is sound relatively smoth, so, my question: has iax same problem as in sip/rtp, where packets are generated along incomming packets (what is probably solved in trunk with: http://bugs.digium.com/view.php?id=5374 0005374: [patch] Asynchronous generation of outgoing frames when timing device available my iax jitterbuffer settings (iax.conf): [general] jiterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=1500 resyncthreshold=-1 thanks for suggestions ;-) PJ ___ --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
