RE: [asterisk-users] Does X100P decode caller ID?
A little googling made me realize that Asterisk demo may not be the best application to look for caller ID because it tries to pick up at first ring. So I zapped demo context with a plain one. This time, no more failed success. But Asterisk only receives New User, no matter which caller calls. (Callers can be correctly identified from other devices.) Even more discouraging - it seems that this value New User comes from inside Asterisk - users.conf says: fullname = New User So receiving New User is equivalent to receiving nothing. Yuan Liu ___ --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] Does X100P decode caller ID?
Yuan LIU wrote: A little googling made me realize that Asterisk demo may not be the best application to look for caller ID because it tries to pick up at first ring. So I zapped demo context with a plain one. This time, no more failed success. But Asterisk only receives New User, no matter which caller calls. (Callers can be correctly identified from other devices.) You need to know when does your carrier send caller ID, some carriers send between 1st and 2nd ring, others after 2nd ring. Try a Wait(1) before Answer to give asterisk a little more time to pick up the callerid. Leo ___ --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] Does X100P decode caller ID?
From: Leo Ann Boon [EMAIL PROTECTED] A little googling made me realize that Asterisk demo may not be the best application to look for caller ID because it tries to pick up at first ring. So I zapped demo context with a plain one. This time, no more failed success. But Asterisk only receives New User, no matter which caller calls. (Callers can be correctly identified from other devices.) You need to know when does your carrier send caller ID, some carriers send between 1st and 2nd ring, others after 2nd ring. Try a Wait(1) before Answer to give asterisk a little more time to pick up the callerid. Leo Asterisk no longer answers. I just let it ring at least 4 times but still no luck. Suggestion? Yuan Liu ___ --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] Does X100P decode caller ID?
On Fri, Jan 26, 2007 at 10:33:49PM -0800, Yuan LIU wrote: The SM56 MODEM manual says it does. But when used with zaptel 1.2.12, nothing shows up. Technically, it is Asterisk that does the decoding, regardless of the card's abilities. This is due to various technical reasons. Make sure you have in zapata.conf: usecallerid=yes callerid=asreceived before the: channel = 1 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Does X100P decode caller ID?
On Sat, Jan 27, 2007 at 12:02:20AM -0800, Yuan LIU wrote: Even more discouraging - it seems that this value New User comes from inside Asterisk - users.conf says: users.conf ? Which versions of asterisk do you work with? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] h323 compile error
I thinik the code is too new for your compiler... I remember reading about needing GCC 2.95 somewhere... I'm just about to post on a similar theme! I am getting the following compile error on h323. Its an old redhat 7.3 system with asterisk 1.2.14, zaptel 1.2.12 pwlib 1.5.2 and openh323 1.12.2 I have pwlib compiled and installed. I have openh323 compiled and installed. I went in the channels/h323 directory and did make opt What shall I do? Jerry ../../include/asterisk/utils.h: In function `void ast_slinear_saturated_divide (short int *, short int *)': ../../include/asterisk/utils.h:199: warning: `always_inline' attribute directive ignored ../../include/asterisk/utils.h: In function `int inaddrcmp (const sockaddr_in *, const sockaddr_in *)': ../../include/asterisk/utils.h:217: warning: `always_inline' attribute directive ignored In file included from ast_h323.cxx:51: ast_h323.h: At top level: ast_h323.h:159: type specifier omitted for parameter ast_h323.h:159: parse error before `*' ast_h323.cxx:957: type specifier omitted for parameter ast_h323.cxx:957: parse error before `*' ast_h323.cxx: In method `H323Channel *MyH323Connection::CreateRealTimeLogicalChannel (...)': ast_h323.cxx:959: `capability' undeclared (first use this function) ast_h323.cxx:959: (Each undeclared identifier is reported only once for each function it appears in.) ast_h323.cxx:959: `dir' undeclared (first use this function) ast_h323.cxx:959: `sessionID' undeclared (first use this function) make: *** [ast_h323.o] Error 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 ___ --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] convert URI string to lowercase
thank you for tip, seems that app_backticks will do exactly what I want,
but as asterisk addon, is it compatible with asterisk 1.4?
maybe this is also idea for developers to add something like this app to
main asterisk tree, because current asterisk lack function/application
for converting strings to lower od upper cases
I think it would be usefull when working with URI (incoming URI strings
should be normalized to further processing in dialplan, eg. lookup in
DB or ldap)
PJ
Ioan Indreias wrote:
Hello,
Maybe using app_backticks will solve your problem.
I use it to call a script and saved the result into an Asterisk variable.
http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
Regards,
## nini @ www.modulo.ro ##
Pavel Jezek wrote:
any idea, how to do something like this, but in correct/functional
form? ;-)
Set(foo=System(echo ${EXTEN} | tr [:upper:] [:lower:]))
${EXTEN} is SomeStrinG
${foo} output should bee somestring
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Re: [asterisk-users] VIA EPIA DeadLock Issues
Via EPIA CN1 as well. Di you find any solutions? On 1/10/07, Raymond McKay [EMAIL PROTECTED] wrote: Greetings, I've been having a large number of deadlock issues lately on chan_sip occurring only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar issues. My Config (have multiple systems all running the same hardware with the same problem) VIA EPIA ML6000 1GB RAM 80GB HDD Various Digium Cards (T1 and TDM cards) Trixbox 1.2.2 (though running stock asterisk code) Asterisk Versions 1.2.12 - 1.2.14 - with and without metermaid patch Problem seems to happen more on systems that use parking lots. The system will run for around 24 hours or so fine, and then mysteriously, without any errors leading up to it, will stop being able to send calls to the chan_sip. System from that point on reports the following in the logs. Dec 13 12:07:04 DEBUG[16415] chan_zap.c: Took Zap/1-1 off hook Dec 13 12:07:04 VERBOSE[16415] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for '0x9896848', 10 retries! Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for '0x9896848', 10 retries! attempting to stop asterisk from the CLI causes the CLI to become unresponsive and a trace shows chan_sip goes into a mutex_wait state. Anybody seen this? Have a fix? Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Via EPIA channel_find_locked: Avoided initial deadlock
In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] X100P - zttools says red status
The card that I have got only has one port. I assume there are two versions? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: 27 January 2007 00:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] X100P - zttools says red status Charlie Grosvenor wrote: Yes the line is connected, a standard phone works fine when connected to the line. There're 2 ports on the card. Which port are you using? One of the ports is for connecting another phone in parallel to the card. Leo ___ --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] Simple question
Whats the difference between the following statements in extensions.conf include=inbound AND #include inbound/*.conf -- Regards Rizwan Hisham Software Engineer ___ --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] Nobody there, continuing...
Hi Alex - Running Asterisk 1.2.12 (a bit out dated, but it was fully operational until a few days ago), I'm seeing the following message in my logs, repeated literally millions of times: channel.c: Nobody there, continuing… We've started to see some odd behavior (incoming callers can hear us, we can't hear them, we can't dial out, etc). A reboot fixes the issues for about 3/4 of a day Can you tell us any more about your setup? What OS? What is your dialplan? What are your hardware devices? Have you tried with a newer version? There are some related bugs that have been fixed. I also read a report of this happening when using Nagios. - Noah ___ --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] Simple question
Rizwan Hisham wrote: Whats the difference between the following statements in extensions.conf include=inbound AND #include inbound/*.conf Hi, checkout this page: http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf With the #include filename statement in extensions.conf, other files are included. This way you can setup a system where extensions.conf is the main file, users.conf contain your local users, services.conf contain various services, like conferencing. This way, the dial plan may be easier to maintain, depending on the size of your setup. The #include filename statement is not the same as the include context statement. The #include statement works in all Asterisk configuration files. I believe that #include syntax works like a include in programming languages where the file or files listed are included as part of the file that references them. The include = context syntax is for including on context within another. If context A includes context B then calls going into Context A could possibly match extensions in context B. -- Warm Regards, Lee ___ --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] H.264 *Not Patented*
The H.264 codec patent by Qualcomm has been ruled invalid by a San Diego Federal jury: http://www.eetimes.com/news/semi/showArticle.jhtml?articleID=197001066 . That means that H.264 codecs can now be written, distributed and revised freely under any license their authors choose, including GPL, public domain, or any other, and $free now that royalties are no longer required. How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and audio quality at low bitrates? GSM is $free, but G.729 is higher quality (tho patented with at least $10 per running codec instance royalties). Will H.264 become the favorite high-quality Asterisk codec, or will it perhaps force G.729 to become free, or negligibly cheaper? -- (C) Matthew Rubenstein ___ --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] H.264 *Not Patented*
On Sat, Jan 27, 2007 at 10:20:40AM -0500, Matthew Rubenstein wrote: How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and audio quality at low bitrates? GSM is $free, but G.729 is higher quality (tho patented with at least $10 per running codec instance royalties). Will H.264 become the favorite high-quality Asterisk codec, or will it perhaps force G.729 to become free, or negligibly cheaper? G.729 is $10 from Digium. If you want to go license several thousand codecs (or probably more like 10's of thousands) I think the Sipro license is more like a couple of bucks. Unfortunately you have to license a large number in one go, so the initial set-up is very high. Digium have done a deal (I presume) whereby they've taken the intial hit and are just sub-licensing at a cost which make it whorth while for them. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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
Re: [asterisk-users] Does X100P decode caller ID?
From: Tzafrir Cohen [EMAIL PROTECTED] On Sat, Jan 27, 2007 at 12:02:20AM -0800, Yuan LIU wrote: Even more discouraging - it seems that this value New User comes from inside Asterisk - users.conf says: users.conf ? Which versions of asterisk do you work with? 1.4.0. Anything I need to know about this ver? It's a bit odd to run zaptel 1.2 with asterisk 1.4, but it's otherwise working. (Basically I ran into known issues that zaptel 1.4 has problems with older kernel/gcc/make.) Yuan Liu -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] H.264 *Not Patented*
Matthew Rubenstein wrote: The H.264 codec patent by Qualcomm has been ruled invalid by a San Diego Federal jury: http://www.eetimes.com/news/semi/showArticle.jhtml?articleID=197001066 . That means that H.264 codecs can now be written, distributed and revised freely under any license their authors choose, including GPL, public domain, or any other, and $free now that royalties are no longer required. How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and audio quality at low bitrates? GSM is $free, but G.729 is higher quality (tho patented with at least $10 per running codec instance royalties). Will H.264 become the favorite high-quality Asterisk codec, or will it perhaps force G.729 to become free, or negligibly cheaper? Although I wouldn't complain about a free G.729 codec, I have to be honest in saying that $10.00 isn't that great of an expense considering the better call quality you get. -- Warm Regards, Lee ___ --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] International Carriers
Hi Facundo I´m working with carriers Voip/TDM in Brazil with quality in our terminations. My net is in Brazilian Territory. Visit www.ositelecom.com.br Best Regards Josué 2007/1/26, Rafael Canchola [EMAIL PROTECTED]: Hi: I am working in a VoIP Carrier Company, I could provider you the service for your internationals calls. Please visit www.fonetglobal.com and call me, my phone number is +52 442 167 08 00 x214 Rafael Canchola. Thanks. At 09:54 a.m. 26/01/2007, Facundo Ameal wrote: Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Rafael*Canchola *Product Development Engineer*, FonetGlobal Inc. [EMAIL PROTECTED] http://www.fonetglobal.com *Ph. *+ 52 800 022 10 21 ext. 214 + 52 442 167 08 00 *VoIP* 523663899 *d00d! *cyberalph ___ --Bandwidth and Colocation provided by Easynews.com http://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] International Carriers
Hello Facundo, i have an entreprise in Peru, if you want i can give you a best price for call in peru. My traffic is on net. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Facundo Ameal Enviado el: Viernes, 26 de Enero de 2007 10:55 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] International Carriers Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --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] X100P - zttools says red status
Charlie Grosvenor wrote: The card that I have got only has one port. I assume there are two versions? Thanks NOPE There really is only one port. two modular jacks, one marked line, which should go to the PSTN or FXS device, one with battery on it. The second, upper? port is simply in parallel, and in some modems ( can't say for sure on the X100 ) is opened when the modem goes off hook. If you have a card with only ONE modular jack, then perhaps you don't really have an X100?? Remember the X100 is really a simple WinModem that has a specific chipset that the Zaptel driver was written for, If you were to remove it and place it in a Windows machine with the proper drivers, you would have a run of the mill 56K dial up modem Red alarm in the X100 means it doesn't see battery from the PSTN -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: 27 January 2007 00:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] X100P - zttools says red status Charlie Grosvenor wrote: Yes the line is connected, a standard phone works fine when connected to the line. There're 2 ports on the card. Which port are you using? One of the ports is for connecting another phone in parallel to the card. Leo ___ --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] max tnt pri voice channels 56k or 64k, does it matter, selection parameter?
Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Does X100P decode caller ID?
Yuan LIU wrote: From: Yuan LIU [EMAIL PROTECTED] The SM56 MODEM manual says it does. But when used with zaptel 1.2.12, nothing shows up. Debug level 6 (Asterisk 1.4.0) only shows: [Jan 26 22:56:46] ERROR[25603]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-14) [Jan 26 22:56:46] WARNING[25603]: chan_zap.c:6389 ss_thread: CallerID feed failed: Success [Jan 26 22:56:46] WARNING[25603]: chan_zap.c:6489 ss_thread: CallerID returned with error on channel 'Zap/1-1' If you are in a country that uses FSK Caller*ID, this error usually indicates that the rxgain is too high or too low. ___ --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] Simple question
Whats the difference between the following statements in extensions.conf include=inbound AND #include inbound/*.conf The first one includes a context the second one includes a file(s). -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.12/654 - Release Date: 1/27/2007 5:02 PM ___ --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] Does X100P decode caller ID?
From: Nilesh Londhe [EMAIL PROTECTED] On ebay, I have seen x100p (or clone) with two different chipsets; 1) has motorols chip 2) has something else I dont call. My experience says that the x100p/clone with motorola chipset shows caller id with default * settings. This one (SM56) is Motorola. I did get an authentic X100P from DigitNetworks that uses Intel chipset but haven't tested on this line. Yuan Liu On 1/27/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Tzafrir Cohen [EMAIL PROTECTED] On Sat, Jan 27, 2007 at 12:02:20AM -0800, Yuan LIU wrote: Even more discouraging - it seems that this value New User comes from inside Asterisk - users.conf says: users.conf ? Which versions of asterisk do you work with? 1.4.0. Anything I need to know about this ver? It's a bit odd to run zaptel 1.2 with asterisk 1.4, but it's otherwise working. (Basically I ran into known issues that zaptel 1.4 has problems with older kernel/gcc/make.) Yuan Liu -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] H.264 *Not Patented*
On Sat, Jan 27, 2007 at 10:20:40AM -0500, Matthew Rubenstein wrote: The H.264 codec patent by Qualcomm has been ruled invalid by a San Diego Federal jury: http://www.eetimes.com/news/semi/showArticle.jhtml?articleID=197001066 . That means that H.264 codecs can now be written, distributed and revised freely under any license their authors choose, including GPL, public domain, or any other, and $free now that royalties are no longer required. How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and audio quality at low bitrates? GSM is $free, but G.729 is higher quality (tho patented with at least $10 per running codec instance royalties). Will H.264 become the favorite high-quality Asterisk codec, or will it perhaps force G.729 to become free, or negligibly cheaper? H264 is video, not audio, right? Are those all the patents relevant to H264 or are there any others that Broadcom has aready licensed? Anyway, GSM is nice. SPEEX is nicer and has no patent issues. It also works in other sample rates, which will eventually become handy. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] H.264 *Not Patented*
H.264 is a video encoding standard, not an audio encoding standard. It won't help with phone calls too much, unless you're running video phones that support it. Matthew Fredrickson On Jan 27, 2007, at 9:20 AM, Matthew Rubenstein wrote: The H.264 codec patent by Qualcomm has been ruled invalid by a San Diego Federal jury: http://www.eetimes.com/news/semi/showArticle.jhtml?articleID=197001066 . That means that H.264 codecs can now be written, distributed and revised freely under any license their authors choose, including GPL, public domain, or any other, and $free now that royalties are no longer required. How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and audio quality at low bitrates? GSM is $free, but G.729 is higher quality (tho patented with at least $10 per running codec instance royalties). Will H.264 become the favorite high-quality Asterisk codec, or will it perhaps force G.729 to become free, or negligibly cheaper? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote: Unless you are monitoring calls, want full CDR etc, then that's what you want anyway. CDR are not affected by how the audio flows. While technically true, I believe (it may have changed in 1.4) that if you allow reinvites, the signalling path follows the audio path, and you end up with reported calls lasting 3 seconds. So, if you want full (ie accurate as to the length of time) CDR, then I think asterisk has to remain in the call path. J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to fix error when paging
I am trying to page my Grandstream GXP-2000 phones
and keep getting the error message:
Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete
destination '' supplied.
How can I fix this error?
The two contexts below do either one-way paging or two-way paging to all
Grandstream phones in a list.
[One_Way_Page_GROUP]; one to many page
exten = **1,1,VERBOSE(ENTERING [default] **1 in extensions.conf 1)
exten = **1,n,SIPAddHeader(Call-Info: answer-after=0)
exten = **1,n,Page(${One_Way_Paging_List}|)
exten = **1,n, Hangup
; Note: Above configuration will allow user to one way page(broadcast)
to all ; the extensions defined in variable One_Way_Paging_list
One_Way_Paging_List = SIP/420SIP/422/SIP/400SIP/413SIP/410
[Two_Way_Intercom_GROUP]; one to many intercom
exten = **2,1,VERBOSE(ENTERING [default] **2 in extensions.conf 1)
exten = **2,n,SIPAddHeader(Call-Info: answer-after=0)
exten = **2,n,Page(${Two_Way_Intercom_List}|d)
exten = **2,n, Hangup
Two_Way_Intercom_List = SIP/420SIP/422/SIP/400SIP/413SIP/410
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] BarCampUSA Tickets go on sale this Thursday the 1st of February 2007
I know most people on the Asterisk list are already aware of Barcamp so skip to the notice below. For those that don't read on. Barcamps traditionally are 'unconferences', meaning there is no set format or agenda apart from turn up on the day and be prepared to participate, people who attend are encouraged to speak or rather 'chair' a session, because as always happens when you get a bunch of smart people in the room, people in the room know as much if not more about a topic as you do and there is no audience or panel separation at barcamp, barcamp is all about participation. These 1 or 2 day events take place in cities around the world and are organized from a general site http://barcamp.org http://barcamp.org/ You can also read a review of BarcampNY2 that I attended last year http://deancollinsblog.blogspot.com/2006/10/barcampnyc2.html http://deancollinsblog.blogspot.com/2006/10/barcampnyc2.html People from all over the world are coming to Wisconsin in August so make sure you are there to participate, learn and enjoy. On the 23rd of August 2007 one of the most momentous IT events this year will be occurring www.BarCampUSA.org http://www.barcampusa.org/ For 4 days only from the 23rd to the 26th of August an expected 5-10,000 geeks will be descending on Jefferson, Wisconsin. It has been compared by a number of IT writers already as the 'Woodstock' of the IT world and the tickets haven't even gone on sale yet. Non stop 24 hours a day, for 4 straight days some of the brightest most influential people in IT will be discussing/demo'ing'/networking about what's hot/what's not, what they are working on today and more importantly discussing what's important to them in the future. Discount and other benefits for tickets purchased prior to 1st of May 2007 so purchase from http://www.barcampusa.org/wiki/index.php?title=Tickets from this Thursday. Want to Advance the Asterisk Cause? During BarcampUSA there will be a 'track' or series of events involving Asterisk. If you are interested in discussing or chairing a particular talk please go to www.barcampusa.org/wiki/index.php?title=Asterisk and register your interest or email me with BarCampUSA in the subject. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph ___ --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] FXS - Init Indirect Registers UNSUCCESSFULLY.
I'm having problems with a TDM22B. The FXO modules work fine. Both FXS modules fail to initialized. The error messages I see when the module loads: Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. I already RMA'ed a TDM22B because of this problem. Now that the replacement shows a similar issue, I want to see if anyone else can think of something to try; at least until Monday when I can get an RMA number for this card. If it helps, I have a TDM20B (just FXS modules) that does not see this problem when I place it in the same slot. Here is what dmesg shows for the TDM22B when the system boots, or when I issues modprobe wctdm: Freshmaker version: 73 Freshmaker passed register test !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. Module 0: FAILED FXS (FCC) !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. Module 1: FAILED FXS (FCC) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) Thanks, MikeC ___ --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] How to fix error when paging
Larry Alkoff wrote:
I am trying to page my Grandstream GXP-2000 phones
and keep getting the error message:
Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete
destination '' supplied.
How can I fix this error?
The two contexts below do either one-way paging or two-way paging to all
Grandstream phones in a list.
[One_Way_Page_GROUP]; one to many page
exten = **1,1,VERBOSE(ENTERING [default] **1 in extensions.conf 1)
exten = **1,n,SIPAddHeader(Call-Info: answer-after=0)
exten = **1,n,Page(${One_Way_Paging_List}|)
exten = **1,n, Hangup
; Note: Above configuration will allow user to one way page(broadcast)
to all ; the extensions defined in variable One_Way_Paging_list
One_Way_Paging_List = SIP/420SIP/422/SIP/400SIP/413SIP/410
[Two_Way_Intercom_GROUP]; one to many intercom
exten = **2,1,VERBOSE(ENTERING [default] **2 in extensions.conf 1)
exten = **2,n,SIPAddHeader(Call-Info: answer-after=0)
exten = **2,n,Page(${Two_Way_Intercom_List}|d)
exten = **2,n, Hangup
Two_Way_Intercom_List = SIP/420SIP/422/SIP/400SIP/413SIP/410
I've found and fixed the problem.
Basically, I had to put the two Lists in [globals] and change the ' = '
to a simple = with no spaces. It works now.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] HFC-card and TDM400 with bristuff
Hi, is it possible to run an HFC-card with bristuff and an TDM400 in one PC? best regards Thomas ___ --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] H.264 *Not Patented*
Steve Kennedy wrote: On Sat, Jan 27, 2007 at 10:20:40AM -0500, Matthew Rubenstein wrote: How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and audio quality at low bitrates? GSM is $free, but G.729 is higher quality (tho patented with at least $10 per running codec instance royalties). Will H.264 become the favorite high-quality Asterisk codec, or will it perhaps force G.729 to become free, or negligibly cheaper? G.729 is $10 from Digium. If you want to go license several thousand codecs (or probably more like 10's of thousands) I think the Sipro license is more like a couple of bucks. Unfortunately you have to license a large number in one go, so the initial set-up is very high. Digium have done a deal (I presume) whereby they've taken the intial hit and are just sub-licensing at a cost which make it whorth while for them. Digium is providing a product that includes all the required software and patent licenses. You could buy a million sipro licenses but you still need to develop, buy or license software to implement the codec. ___ --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] Sangoma card dying after 1hour
Do you see anything in /var/log/messages? I am having a similar problem but I'm also getting some pci fatal error! messages. I had sangoma connect to the box and he couldn't find any config errors so we're leaning towards a hardware problem. - Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky Sent: Friday, January 26, 2007 7:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Sangoma card dying after 1hour Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with zaptel 1.2.12 :) Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 12:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour Which asterisk versions etc etc? On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: I am running the newest version, from the sangoma wiki. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded as there are some resource leak fixes in that version. Regards, Steve ___ --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 ___ --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] HFC-card and TDM400 with bristuff
On Sat, Jan 27, 2007 at 11:32:24PM +0100, Thomas Winter wrote: Hi, is it possible to run an HFC-card with bristuff and an TDM400 in one PC? Yes. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] NAT solutions
On Sat, Jan 27, 2007 at 01:55:31PM -0500, Jerry wrote: On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote: Unless you are monitoring calls, want full CDR etc, then that's what you want anyway. CDR are not affected by how the audio flows. While technically true, I believe (it may have changed in 1.4) that if you allow reinvites, the signalling path follows the audio path, and you end up with reported calls lasting 3 seconds. So, if you want full (ie accurate as to the length of time) CDR, then I think asterisk has to remain in the call path. That would have been an odd bug. Signalling in SIP only moves when you do a REFER or similar. Reinvites can't change it. Having the signalling flow differently from the audio is a feature, not a bug, a very important one. With SIP INFO (or its planned successor) you can get the DTMF without having to get the audio, which is highly valuable. Right now Asterisk needs to stay in the audio stream to get DTMF, and that is one of the prime reasons it does. (The others are NAT, recording and meetme, the latter 2 of which should be a small minority of calls.) This is an important thing. Done properly, audio should almost never flow through the switching machine, or only flow for a portion of the call. The result can be orders of mangitude difference in bandwith requirements. ___ --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] Sample Config.
Hello Mihaela, my name is Catalin. Thank you for you advices, I still hev problems with my spa3102, I just wanna use a xlite to log in to sipura... is that possible? I try to make small steps to understand all options. Thank you for your support. Catalin S. On 1/27/07, Token PBX [EMAIL PROTECTED] wrote: Hi! I don't understand what you mean by : „configure voice part on it, but I can give general guidelines: First you setup SPA3000 web UI: 1) Line1 Tab: Sip settings: SIP port : 5060 Proxy and Registration: Proxy: Asterisk IP Subscriber Information: Display Name: FXS_username Password: FXS password User ID: FXS_username 2) PSTN Line Tab: SIP Settings: SIP port: 5061 Proxy and Registration: Proxy: Asterisk IP Subscriber Information: Display Name: FXO_username Password: FXO_password User ID: FXO_username Dial Plans: Dial Plan 1: (S0:[EMAIL PROTECTED] IP:5060)(may be any other dial plan) VoIP-To-PSTN Gateway Setup: VoIP-To-PSTN Gateway Enable: Yes Line 1 VoIP Caller DP: 1 (or any other setup like Dial Plan 1) VoIP Users and Passwords (HTTP Authentication) VoIP User 1 Auth ID: asterisk VoIP User 1 DP: 1(same as above) PSTN-To-VoIP Gateway Setup: PSTN-To-VoIP Gateway Enable: Yes Then Asterisk sip.conf: [ FXO_username] disallow=all allow=alaw type=friend fromuser= FXO_username username= FXO_username secret= FXO_password host=dynamic dtmfmode=rfc2833 canreinvite=no qualify=1000 context=incoming port=5061 [FXS_username ] disallow=all allow=alaw type=friend username= FXS_username secret= FXS_password host=dynamic dtmfmode=rfc2833 canreinvite=no qualify=1000 context=outgoing Best regards Mihaela MJ On 1/26/07, Jonson Player [EMAIL PROTECTED] wrote: Hello, I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to configure voice part on it. I cannot get it if I can use like peer for my asterisk. Please help me with some tips. Thank you guys. Regards, Jonson. ___ --Bandwidth and Colocation provided by Easynews.comhttp://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 http://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] max tnt pri voice channels 56k or 64k, does it matter, selection parameter?
FWIW My TNT gateway also connects at 56k on voice calls. Tom At 11:55 AM 1/27/2007, you wrote: Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Adding 4 more POTS lines
Jim Freeze wrote: I suppose that is my alternative - remove the 4FXO card and add an 8FXO card. But I'm not seeing the prices you list. The Digium TDM2402B is listed at $837.00. Am I missing something? Digium is releasing a new 8 FXO/FXS card TDM800P, based on the same expansion cards used for TDM2400. This new card has been announced on the IT Expo East, which ended last Friday in Fort Lauderdale, FL. ___ --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] Response on dialin - no extension
On a SIP phone is it possible to enter the dialplan when the user picks up the phone without having to wait for the user to press an extension? Is is possible to do something like [sip-test] s,1,Answer s,2,Playback(welcome) s,3,WaitExten(30) 1,1,Noop(exten 1) ... t,1,Goto[s,2] - Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games.___ --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
