Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)
[EMAIL PROTECTED] ha scritto: On Wed, Apr 12, 2006 at 09:32:12PM +0200, Sergio Chersovani wrote: [EMAIL PROTECTED] ha scritto: context = from-sccp-intenal I guess intenal is not the righe context :-) Sergio The from-sccp-internal is almost an exact copy of my from-sip-internal context, which works fine there's a typo in your sccp.conf intenal instead internal, so of course the context does not exists Sergio ___ --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: Double sip logins
On 4/8/06, Joe [EMAIL PROTECTED] wrote: Remove the SIP /400 entry from the Asterisk DB. Database del At asterisk prompt. Or look at the wiki for info on how to remove it. Or make sure the SIP/500 uses a different IP address than the old SIP/400. Joe thanks for your reply i've tried to remove the entry in the database, it works, but if i reboot the phone it still register itself with both 400 and 500 accounts!! do you know how to reset the phone settings? 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] AAstra 9133i register double account.. ??
On 4/10/06, William Harrison [EMAIL PROTECTED] wrote: How is the 9133i configured, through the .cfg file, the WebUI, or the Phone's own interface? The PhoneUI WebUI take precedence over the .cfg file. You can look at the WebUI and see what the current settings are, and clear them out if you'd rather use the .cfg file settings. the 9133i is configured through the webUI... sorry but where is this .cfg file? 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] playback soundfile in memory
I want to playback sound file loaded in memory not from a file...is this possible? _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --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] * 1.2.4 1.2.6: Ringing anamoly
12 apr 2006 kl. 18.38 skrev Ronald Lewis: I was alerted the other day by of all people, my mom, that she wasn't hearing a ring when she dialed my number. Puzzled, I tried calling myself. The call connects, but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I figured, another damned Broadvoice issue. For kicks, I upgraded to 1.2.6 today, and the end result is the same. So, I went to the dialplan playground, and removed a few lines for testing. It turns out that if I playback a file before ringing an extension, ringing works fine. Without, dead silence. Any ideas? Not really, but a hint: * Check if we receive a ringing indication on the outbound call from Asterisk * When you playback, you answer the call (unless you use the noanswer option). After that, Asterisk will generate the ringing internally in audio instead of sending an indication signal Seems like there is a signalling problem so that Asterisk does not get a ringing indication from the device that we ring before moving to voicemail, or that Broadvoice does not get the ringing indication we send to them. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] Problem with Voice Quality
[EMAIL PROTECTED] wrote: Hi All, We are making a VOIP application for Mobiles (PDA's) and we are using Asterisk for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP router and routes everything to Asterisk. We also have rtpproxy for SER. Our packet delivery from clients (Mobiles, PDA's) is inconsistent and ranges between 10 to 60 ms delay but the average is near to 20 ms. We use SIP. How can we solve this problem, is there any setting at the server end to handle this, as clients have very limited resources we have to manage this at the server end, please tell me how can I do this? Sanity check: a) What kind of connectivity - WiFi or GPRS, 3G etc? b) What's the ping time to your clients? From your ping values, I think you're running over WiFi? Always bear in mind, anything going over the air will have delay. It's pretty much out of your control. Some things that you can do to smooth it out: a. Use a low bit-rate codec that does PLC b. Use a large jitterbuffer c. Send more than 1 frame per packet. I don't think stock Asterisk can do this, but I remember there was mention of a patch for it. d. If you're using WiFi, you might want to check your cell planning. WiFi handover is a bitch for VoIP. a. and b. will pretty much depend on what your PDA softphone is capable of. 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] AAstra 9133i register double account.. ??
On Thu, 2006-04-13 at 09:00 +0200, nik600 wrote: On 4/10/06, William Harrison [EMAIL PROTECTED] wrote: How is the 9133i configured, through the .cfg file, the WebUI, or the Phone's own interface? The PhoneUI WebUI take precedence over the .cfg file. You can look at the WebUI and see what the current settings are, and clear them out if you'd rather use the .cfg file settings. the 9133i is configured through the webUI... sorry but where is this .cfg file? thanks All is explained in Aastra's excellent manual available from their web site. -- Dave Cotton [EMAIL PROTECTED] ___ --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] TE410P upgrade to TE411P = (solution to) no more fax carrier detection !
For the moment, if you need FAX tone detection, you will need to use 'vpmdtmfsupport=0' in your module configuration for loading the wct4xxp module; this will not disable the echo canceler, just stop using it for tone detection. Any idea if/when this will be addressed? We had been going with the VPM modules to get better DTMF detection and echo can but if we have to move DTMF detection back to Asterisk in order to get CNG detection then we're concerned. (We haven't added fax detection to our systems yet but it is planned.) -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --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] Urgent: Unable To Execute after updating from SVN
As a clarification for further posts, its wise to delete both the asterisk modules and header directory when having problems upgrading from 1.0 to 1.2 and depracated modules are in the way. As Rob T. pointed out the best way to do this is: # mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.oldand # mv /usr/include/asterisk /usr/include/asterisk.oldOn 1/30/06, Boris Bakchiev [EMAIL PROTECTED] wrote:Delete /usr/lib/asterisk/modules/app_md5.so and update your dialplan if you use MD5. It is now done in functions./usr/lib/asterisk/modules/app_md5.so is a leftover from your previous installation.[app_md5.so]Jan 29 02:49:10 WARNING[32424]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/app_md5.so: undefined symbol: option_priority_jumping Jan 29 02:49:10 WARNING[32424]: loader.c:555 load_modules: Loading module app_md5.so failed!___--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] Re: update - 512 Simultaneous Callswith DigitalRecording
[EMAIL PROTECTED] wrote: On Wed, 12 Apr 2006, Leo Ann Boon wrote: I'm not sure tmpfs is the right solution for the OP's problem - disk access slowing down the system. My understanding of tmpfs is that it will swap pages in and out to/from disk. Wouldn't that be as bad as directly writing to disk? I can see tmpfs will have some advantage over direct disk IO when the files are small and short-lived, i.e. less likely to be swapped. One way around this is to not have swap at all. Then there is no disk i/o to worry about. Everything will be in ram. This is what I do for embedded asterisk servers. tmpfs and no swap. Ram is cheap. Problem solved. I guess it really depends on the load. The OP wants to record 512 concurrent calls. A quick calculation would show that the system will need (1024 * 8Kb)/second, that's about 8Mb per second. Assuming a mobo with 12GB of RAM, we're looking at around 25 minutes of calls. Sounds like a fair deal. -Dan ___ --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 BRI in the USA
Hi Joe, In your mail you wrote that I've heard a few stories that reported partial success with an Eicon Diva Server card, but always with the caveat that it doesn't work quite right or something along those lines. I can ensure you that this is not the case. We are implementing a Diva Server card in our call centre with Asterisk - so it works perfectly on both BRI and PRI lines. You need to follow the instructions here though. http://www.eicon.com/support/helpweb/slnxen/asterisk.asp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: 12 April 2006 14:06 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk BRI in the USA I dunno if it's THAT bad. I had a BRI line in the (relatively) podunk town of Kalamazoo, Michigan back in 1998. Sure, it took the phone company a couple of weeks to provision the service, but it takes the phone company a couple of weeks to do most anything in my experience. The price was something like $45/mo for two channels and the same per-call/per-minute pricing scheme as POTS (no per-minute fee for incoming and local calls, regular LD pricing for LD, and 800 local outgoing calls included after which it was something like 6 cents per call). The switch on ILEC's end was a DMS-100 implementing National ISDN-1. I really put the ISDN line through its paces too -- voice, data, bonded data, automatic bonding and de-bonding to allow for voice calls -- and everything always worked flawlessly. I don't know what today's pricing is like for ISDN BRI what with all of the various mergers (at the time, I had service from Ameritech), but unless it has gone up significantly, BRI seems like the perfect type of trunk for an Asterisk system too small for a T1/PRI to be an affordable option. It's still similar. Out here, we get a lot of RF interference, and it turns out that BRI is actually cheaper than equivalent POTS lines with Caller-ID (a feature I require), and you can do neat stuff like having 56K dial-in with a USR I-Modem. However, CPE has always been very limited here in the States, and there was no good way to hook up direct to Asterisk. I've heard a few stories that reported partial success with an Eicon Diva Server card, but always with the caveat that it doesn't work quite right or something along those lines. CPE like the USR I-Modem won't deliver Caller-ID to the POTS port. Other CPE like the Motorola BitSurfr Pro is sensitive to RF noise. We were using Netgear RT338's for a number of years, but they are all burnt out now and impossible to replace (actually most CPE is virtually irreplaceable, as so few mfr's make ISDN gear anymore). And while most CPE was OK with our old POTS based phone system, almost none of it worked reliably with POTS-VOIP gateways, such as the Sipura SPA-3000. Further, BRI has two channels, and the U interface pretty much dictates that you feed both of them to the same place. Putting them into an Asterisk box, I would lose the ability to use the USR I-Modem, for example... Despairing, I thought I might have to abandon the beautiful digital delivery of ISDN, which is stupid when you have a digital (VoIP) phone system. But: After talking with a friend up in Minneapolis, I bought an Adtran Atlas 550 off of eBay, which is a versatile Swiss Army Knife for telecom needs. With a quad port ISDN BRI and an octal FXS, it's the killer CPE device, but the best part is that it also does T1/PRI, so you can /convert/ BRI to PRI, etc. I've not actually done that just yet, though I do have a Digium T1 card around here somewhere and want to try it out one of these days. So, I can't actually say it /works/, but it's supposed to. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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] web meetme instructions
Hi,so how do I contribute the translation? (this would be one of my first contrib to an open source project).For the translation, I used Translation2.php from the pear repository and put the translated text in an xml file. Please informe me on how to contribute.benqOn 4/3/06, Dan Austin [EMAIL PROTECTED] wrote: Do you have an idea when this new submission will be available?I received the update over the weekend and will be looking at it this week.If possible I would like to include Ben's French and Germantranslations, which will likely take sometime.A hairy guess would be in third week on April. Dan Austin [EMAIL PROTECTED] wrote: Sorry for the late reply, I was away on vacation. Version 1.2 was created by Areski and I extended it to include the scheduling functions.I guess I should get an account on the Wiki and make some updates. If all you need is a tool for monitoring conferences, version 1.2 is the way to go. If you want scheduling features with optional granular access controls, thenthe latest version from www.fitawi.com/Asterisk is the correct choice. I am expecting a fairly large code contribution soon, from a nice gent with muchmore PHP experience than myself.Once I get a chance to review it I will postan update to 2.1.If you'd like I cannotify you before I release it and we can work on getting your translations in. Dan___--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] Background music in call
The thing I absolutely need is. To play a background music in call. If I have the opportunity to stop it via entering a dtmf combination is would be very very nice also. Does anybody know some application do this. NZR __ 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
Re: [Asterisk-Users] Asterisk BRI in the USA
I'm seeing Diva Server V-BRI running close to $1K/card. There are other Diva cards running around $700. A little pricy but not impossible to do. I remember back in the 90's I had ISDN into my home for internet access. The netgear router I used cost me about $350 back then, and it worked great. I still have it as a matter of fact. However internet access is not what I need. I'm still waiting for the ILEC (HawaiianTelcom) to get back to me to find out if it is even possible to do BRI into my office. The nearest ISDN capable CO is located a bit of a distance from my office (actually its closer to my home). The local CO dosen't have BRI capablities. From what I'm hearing when you bundle together all the costs BRI PRI are gonna be close in price (from a H/W point of view.) Maybe I should just look into going the PRI route and try to find some people willing to buy on my extra DiD's? Any one what a phone number in Hawaii? :) Its such a shame I can't leave well enough alone and suck it up on POTS (eck). I'll keep you informed as to my progress (or lack there of). Mark Coccimiglio n3whx @amsat.org sip:[EMAIL PROTECTED] Walt Reed wrote: I'm in a similar situation. Being on the end of a long loop, POTS sucks - echo / static / crappy calling features. Paying around $2K-3K for BRI solution is a non-starter though. It needs to get down to the $200-400 / port level (more ports = cheaper per port) to be viable. Soho / Very small business (under 12 people) is definately a 1-2 port market which my guess would be the bulk of sales for BRI. It would be awesome to see a Sangoma BRI card. It's hard to say what the market would be since the US telco companies have really tried to kill BRI service. Considering what a full PRI costs, there is also a point where too many BRI ports no longer makes sense, but that number is probably 4-6 BRI's. I was in a situation where I really only wanted 4 BRI's, but had to look at a PRI instead which ended up wasting a lot of money in the long run. POTS was a non-option. ___ --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 use external smtp server for sending mail
is it possibile to set up an external smtp server for the relay to the users of the mails? 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] Any way to prevent this from happening
Typical user error, one user forwards his calls to another using CFwdAll on Cisco 7940, but the user receiving the call has done the reverse. -- Called 117 -- Got SIP response 302 Moved Temporarily back from 10.139.2.15 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/117-df17) -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten) in new stack -- Executing Set(Local/[EMAIL PROTECTED],2, LANGUAGE()=en) in new stack -- Executing SetMusicOnHold(Local/[EMAIL PROTECTED],2, aviareps) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/114|20|t) in new stack -- Called 114 -- Got SIP response 302 Moved Temporarily back from 10.139.2.14 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/114-2036) -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten) in new stack -- Executing Set(Local/[EMAIL PROTECTED],2, LANGUAGE()=en) in new stack -- Executing SetMusicOnHold(Local/[EMAIL PROTECTED],2, aviareps) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/117|20|t) in new stack -- Called 117 -- Got SIP response 302 Moved Temporarily back from 10.139.2.15 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/117-3adf) -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten) in new stack -- Executing Set(Local/[EMAIL PROTECTED],2, LANGUAGE()=en) in new stack -- Executing SetMusicOnHold(Local/[EMAIL PROTECTED],2, aviareps) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/114|20|t) in new stack -- Called 114 -- Got SIP response 302 Moved Temporarily back from 10.139.2.14 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/114-e4df) -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten) in new stack -- Executing Set(Local/[EMAIL PROTECTED],2, LANGUAGE()=en) in new stack -- Executing SetMusicOnHold(Local/[EMAIL PROTECTED],2, aviareps) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/117|20|t) in new stack -- Called 117 -- Got SIP response 302 Moved Temporarily back from 10.139.2.15 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/117-5ff1) -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten) in new stack -- Executing Set(Local/[EMAIL PROTECTED],2, LANGUAGE()=en) in new stack -- Executing SetMusicOnHold(Local/[EMAIL PROTECTED],2, aviareps) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/114|20|t) in new stack -- Called 114 -- Got SIP response 302 Moved Temporarily back from 10.139.2.14 -- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/114-45e0) -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten) in new stack -- Executing Set(Local/[EMAIL PROTECTED],2, LANGUAGE()=en) in new stack -- Executing SetMusicOnHold(Local/[EMAIL PROTECTED],2, aviareps) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/117|20|t) in new stack -- Called 117 Does anyone know of a way to prevent this from happening by dialplan manipulation? Also, same thing happens when a user forwards all to him or herself. Regards toall, Joe ___ --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] free tollfree termination
Hi ! Anybody know if 1800 free termination services from trxtel are in troubles? I can´t reach it, and don´t know why. Thanks a lot gus At 06:49 a.m. 26/03/2006, you wrote: Hi there, Thanks for the tip ! I am happily using this service now. One question though : I cannot get DTMF to work. Is there anything I can do in my asterisk setup to fix this ? Thanks, Lukas trixter aka Bret McDanel wrote: http://www.trxtel.com/index.php?page=Tollfree_Termination This is a free service, I am not selling anything with this service. I just thought that individuals that read this list may enjoy getting tollfree access free this way (yet another way) given that it lets you send your caller id and some of the other free providers dont let you do that. Starting a test service now, for individuals free north american tollfree termination. For carriers that do large quantities of minutes (a not really defined term as yet, more a negotiated value) we will share revenue with you for sending calls to us. If you set up IP PBX systems for customers, add a route in and make residuals off those customers. Run a ITSP? Get revenue for each minute that a customer dials a north american toll free. If anyone has any problems using the service I would appreciate hearing about it, the service will remain free even after the test period, however to get compensation requires an account so that it can be uniquely tracked. Granted tollfree traffic isnt usually the bulk of a provider, but at least now you can provide it free to your customers without losing on costs like bandwidth :) ___ --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] Music on hold problem
Hi, i'm having problems with the MOH module. In a queue sometimes it just stop playing, does anyone have some idea what could be wrong? No verbose data... Tks, Daniel Korndorfer ___ --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] Music on hold problem
What version of Asterisk? On Thu, 2006-04-13 at 12:38, Daniel Korndorfer wrote: Hi, i'm having problems with the MOH module. In a queue sometimes it just stop playing, does anyone have some idea what could be wrong? No verbose data... Tks, Daniel Korndorfer ___ --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] Hangupcause to handle Called party disconnect ? PSTN----E1----OldPBX---E1--Asterisk
Hi,I've been debuging the call disconnection problem in our
architecture:PSTN---E1---OldPBX---E1---AsteriskThis is our
problem:-SIP user agent A calls a pstn phone B.-B hangs up the
call.-SIP user agent A starts listenning busytones... But the call still
on. (and being payed).- Call only ends when it is correctly hanged up in the
SIPphone.I've been tracing the communications between the OldPBX
(NETWORK) and Asterisk (USER SIDE) and i found this:M03 PROGRESS
I08 CauseCoding Std=CCITTLocation=Private net-remoteCause
Code=16I1E Progress indicatorCoding Std=CCITTLocation=Public
net-localProgress desc=Inband info availI28 DisplayInfo=CHAMADA
DESLIGADA08 02 00 02 03 08 0285 90 1E 02 82 88 2811 43 48 41 4D
41 4441 20 44 45 53 4C 4947 41 44 41RXB From User Side
00:45:29.902 Fr.25L2: Sapi=0 Tei=0INFOpf=0 Nr=84 Ns=6900 01
8A A8L3: PD=08 CR(D)=2M7D STATUS I08 CauseCoding
Std=CCITTLocation=UserCause Code=98I14 Call stateCoding
Std=CCITTState=1008 02 80 02 7D 08 0280 E2 14 01
0AThis trace reports to a called party that hanged up the call,
then our old PBX talked to Asterisk with :PROGRESSCause
Code=16and Asterisk answered with Location=UserCause
Code=98I've been looking ISDN cause Codes and i found:Cause No.
98 - message not compatible with call state or message type
non-existent.This cause indicates that the equipment sending this cause has
received a message such that the procedures do not indicate that this is a
permissible message to receive while in the call state, or a STATUS message was
received indicating an incompatible call state.I hope you can advice me.
Is it affordable to use Hangupcause?what we need is that, if the called
party hangs, asterisk should hang (safety reasons on billing)..exten
= _2,1,Dial(Zap/g1/${EXTEN})exten =
_2,2,gotoif,$[${HANGUPCAUSE} = 16]?9|1exten =
9,1,HangupI'm not sure if this is possible neither recommended,
should be HangupCAUSE=16 or =98 ??Best regards,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] ast_sched_runq ran 281 scheduled tasks all at once
Just noticed that I occasionally get these messages:- Apr 12 09:27:03 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 281 scheduled tasks all at once Apr 13 09:13:18 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 1987 scheduled tasks all at once Apr 13 12:47:56 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 1804 scheduled tasks all at once Are they anything to be concerned about? ___ --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] Music on hold problem
I had this problem with 1.2.5, 1.2.6 and now with 1.2.7... On 4/13/06, Gareth Blades [EMAIL PROTECTED] wrote: What version of Asterisk? On Thu, 2006-04-13 at 12:38, Daniel Korndorfer wrote: Hi, i'm having problems with the MOH module. In a queue sometimes it just stop playing, does anyone have some idea what could be wrong? No verbose data... Tks, Daniel Korndorfer ___ --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] IP logging
Hi,I need to have the information about the current IP address of the user. I want to know IP address from which user is registered to Asterisk server. Is it possible with Asterisk to log this information to the database or file? Does anyone can give me some info about this issue? Thanks in advance Cheers Andrew ___ --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 terminate ringing call before it is answered?
Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because the current number it is not being answered, and you don't want to hangup before dialling another number. /Obelix ___ --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 BRI in the USA
On Wed, Apr 12, 2006 at 11:16:10PM -1000, Mark Coccimiglio said: I'm seeing Diva Server V-BRI running close to $1K/card. There are other Diva cards running around $700. A little pricy but not impossible to do. I remember back in the 90's I had ISDN into my home for internet access. The netgear router I used cost me about $350 back then, and it worked great. I still have it as a matter of fact. However internet access is not what I need. I'm still waiting for the ILEC (HawaiianTelcom) to get back to me to find out if it is even possible to do BRI into my office. The nearest ISDN capable CO is located a bit of a distance from my office (actually its closer to my home). The local CO dosen't have BRI capablities. From what I'm hearing when you bundle together all the costs BRI PRI are gonna be close in price (from a H/W point of view.) Maybe I should just look into going the PRI route and try to find some people willing to buy on my extra DiD's? Any one what a phone number in Hawaii? :) Its such a shame I can't leave well enough alone and suck it up on POTS (eck). I'll keep you informed as to my progress (or lack there of). From my research, the problem with PRI's is that you generally pay a lot for the circuit - especially if you only need 8 channels or so. While you can go ahead and get a full PRI for not much more than a partial PRI, the cost of the taxes on the unused channels kills the budget when you look at a 2 year cost. I found a telcom broker in San Francisco that works with all the top providers, and while a lot of the competitors to the ILEC's had lower up-front prices, they got you by not including the costs of the taxes and had other fees too that killed the savings. This was especially true for less than 2 year commitments. Telco's hate BRI's because it takes more cable pairs and repeater hardware when you need more than one. In some cases they end up putting in a T1 / micro DLC. You also can't do DSL over an ISDN BRI line. BTW, a little birdie told me that Sangoma is working on a BRI card. Yeah! ___ --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] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)
On Wednesday 12 April 2006 18:40, Jim Rice wrote: Had I have documented the process and included config files and log files and tcpdump traces, wouldn't I have received the TMI lecture instead? Depends on how verbose you were. The [macid].cfg file is very, very short though (1-3 lines IIRC) and while a tcpdump -s0 -w blah host [ftp server] and port 21 could be verbose, stripping out all but the relevant data would have given something like RETR 00045601d3.cfg 200 OK RETR bootrom.ld 200 OK STOR 00045601d3-boot.log 200 OK RETR sip.ld 200 OK RETR phone221.cfg 200 OK RETR phone1.cfg 200 OK RETR sip.cfg 200 OK STOR 00045601d3-app.log 200 OK (or something similar) Not too wordy... it's when you go to paste the raw binary or include 50k lines of text to your message that the TMI stuff comes around... at least with me, I can't speak with others on the list. I did not ask for what you have tried. Only if you had seen the error. I've seen similar errors, and I was sharing my experience. OK, now I throw up my hands. Your answer then, is no. That *specific* error, no I have not seen. -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] OT: MWI on Treo 600/650
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. dbc. ___ --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 BRI in the USA
Hi Joe, In your mail you wrote that I've heard a few stories that reported partial success with an Eicon Diva Server card, but always with the caveat that it doesn't work quite right or something along those lines. I can ensure you that this is not the case. We are implementing a Diva Server card in our call centre with Asterisk - so it works perfectly on both BRI and PRI lines. You need to follow the instructions here though. http://www.eicon.com/support/helpweb/slnxen/asterisk.asp Happy to hear it... it's just depressing how poorly BRI is supported in general here in the States though. I know a number of folks who do (or did) BRI for telephony going back a number of years (mid '90's at least) and I've yet to hear of one who succeeded in hooking one up to Asterisk without issue, though there are a few, as I mentioned, who reported partial success with the Eicon. BRI is becoming more poorly supported as time goes on by CPE mfrs, and while there was at one time a loose race to implement features such as Caller-ID delivery to POTS ports on ISDN CPE, now there doesn't appear to be much even happening in the way of development of new CPE. This pretty much matches the way it is being supported by the telcos, who do not want to offer any of the powerful capabilities that could be possible over BRI. By offering only packages that implement basic functionality, and requiring that you buy services a-la-carte if you want to do anything odd like (shudder) get an extra DN or ten, and pricing a-la-carte services sky-high, they've guaranteed that it isn't competitive with PRI - but also guaranteed that it isn't all that useful to anyone, either. It's insane that the only solution that seems to work for people is a $1K-range card. It's good that it works perfectly for you; I've yet to meet anyone else who has made that claim, as the reports I've seen are all of partial success. For a long time, I thought I was going to have to go get a Cisco gateway with a VIC-2BRI or something like that, but that scared me too. :-) So I still like my (very pricey) solution with the Adtran. It's nice and it works so far (caveat, haven't got it working with * yet). More interestingly, it can act as either network or user, meaning I can hook up existing devices such as the USR I-Modem to it, and direct only a single B channel over there... now I've just got to find out what happens when I configure up its PRI and hook it up to *. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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] OT: MWI on Treo 600/650
On Thursday 13 April 2006 09:02, David Cook wrote: My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. I've been working on this off and on for AGES. There are some SMS portal sites that claim to be able to do this as well, but I have not managed to find one. -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] AAstra 9133i register double account.. ??
If the phones are registering twice, and you are 100% sure they are only configured via the WebUI, then you must have the settings in two places. There are SIP configs for Global SIP, and also for every available line. You may have one set of settings in Global SIP, and a different one for one of the lines, and thus have two registrations. To be sure that you don't have any latent config items, you may want to factory default the phone, and start from scratch. The .cfg files are simple text config files that can be downloaded from a TFTP/FTP/HTTP server. This is all explained in the Administrator's Guide available from Aastra's website: http://www.aastra.com/support/enterpriseip/show_manuals.asp ___ --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] IP logging
Andrew, This is written to the asterisk database. Use database show from the CLI. That will show the IP addresses of the people that are registered in the /SIP/Registry/(username). William Piper From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Nowrot Sent: Thursday, April 13, 2006 7:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP logging Hi, I need to have the information about the current IP address of the user. I want to know IP address from which user is registered to Asterisk server. Is it possible with Asterisk to log this information to the database or file? Does anyone can give me some info about this issue? Thanks in advance Cheers Andrew ___ --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] AgentCalled event
Hi, I'm writing a Java client/server application that talks to the Asterisk manager interface via the asterisk-java stuff. The idea being it will give you an app to run on your desktop that monitors your phone essentially. Once I've got something vaguely working it will be released under the GPL and hopefully people will contribute to it etc... As part of this, I'm currently trying to understand the various Asterisk manager events, for normal calls I can now successfully keep track of them, understand when they finished, if they were answered or not, handle transfers etc etc. The problem I've got is with queues. In my system for example I used AgentCallbackLogin. I can handle the renaming that happens when a call is connected to an Agent, the problem I've got is determining what queue a call to an agent is from (I have the same agents in multiple queues). The reason I'm having problems, is that the AgentCalled event doesn't have a queue name in it, is this deliberate, and if so why, as the AgentConnect and AgentComplete events both have queue names in. I've come up with a workaround by looking at the context on the AgentCalled event, but this isn't reliable enough to use in general, as it is specific to my system. The other issue is that I'm wondering if there is a better way of determining if a new call is an agent call other than matching the callerid details from the AgentCalled event with the incoming call, and if they match and its within a certain amount of time of the AgentCalled event assume they're related. I guess what I'm looking for is a uniqueid in the AgentCalled event that I can simply compare? Thanks in advance, Alex Brett [EMAIL PROTECTED] http://www.loho.co.uk/ ___ --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] Music on hold problem
Daniel Korndorfer wrote: i'm having problems with the MOH module. In a queue sometimes it just stop playing, does anyone have some idea what could be wrong? No verbose data... Occasionally I am seeing music on hold stop playing for parked calls, so this isn't unique for queues. Maybe it is as Daniel said something to do with MOH. I am running 1.2.6. Don Pobanz ___ --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] Question on parkinglot
Hi I'm a little confused here... trying to setup a parking lot... lot is setup but how do I send calls to the parkinglot? If I allow #700 transfer, it seems I can only transfer on inbound calls... if I use a T in my dialplan I can only transfer on outbound calls... additionally pressing # to use an auto attendant elsewhere causes asterisk to try to transfer... any thoughts? Is there another way to blind transfer to the 70 extension yet be told what parking space your call is in? ___ --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] Codec GSM Makefile Patch for IA64
Currently, compiling asterisk on an Itanium fails with the GSM codec. All I could find on Google was a hack to basically remove GSM from the build which is not an option for some. This patch will allow it to compile and seems to work perfectly. Thanks, Steve Totaro http://www.asteriskhelpdesk.com IA64.codecs.gsm.Makefile.patch Description: IA64.codecs.gsm.Makefile.patch ___ --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] AgentCalled event
On 4/13/06, Alex Brett [EMAIL PROTECTED] wrote: Hi, I'm writing a Java client/server application that talks to the Asterisk manager interface via the asterisk-java stuff. The idea being it will give you an app to run on your desktop that monitors your phone essentially. Once I've got something vaguely working it will be released under the GPL and hopefully people will contribute to it etc... As part of this, I'm currently trying to understand the various Asterisk manager events, for normal calls I can now successfully keep track of them, understand when they finished, if they were answered or not, handle transfers etc etc. The problem I've got is with queues. In my system for example I used AgentCallbackLogin. I can handle the renaming that happens when a call is connected to an Agent, the problem I've got is determining what queue a call to an agent is from (I have the same agents in multiple queues). The reason I'm having problems, is that the AgentCalled event doesn't have a queue name in it, is this deliberate, and if so why, as the AgentConnect and AgentComplete events both have queue names in. I've come up with a workaround by looking at the context on the AgentCalled event, but this isn't reliable enough to use in general, as it is specific to my system. The other issue is that I'm wondering if there is a better way of determining if a new call is an agent call other than matching the callerid details from the AgentCalled event with the incoming call, and if they match and its within a certain amount of time of the AgentCalled event assume they're related. I guess what I'm looking for is a uniqueid in the AgentCalled event that I can simply compare? It is intentional because AgentCalled is fired off from within chan_agent and you can call an agent's channel from outside of a queue. AgentConnect and AgentComplete are called from app_queue. I agree, it is somewhat confusing. We should probably consider a change to the terminology. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] IP logging
Hi,Thanks for so fast replyOk I know about this but actually I am thinking about logging the IP address of a user in realtime. Each time the user changes his location and register Asterisk will log the time and IP address. Is it possible? Best wishes ___ --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] freepbx dialing prefix
Just an update found a few bug tickets regarding it and a change to page.trunk.php which allows the w. Apparently it will be fixed by version 2.1 Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, April 12, 2006 9:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] freepbx dialing prefix Submit a bug report to the FreePBX team? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland Sent: Wednesday, April 12, 2006 8:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] freepbx dialing prefix I need to put a w in the dialing prefix, but it says it isnt valid. If I manually modify the extension file, it then affects all calls made over any trunk. Any ideas? Sean -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/309 - Release Date: 4/11/2006 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/309 - Release Date: 4/11/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/310 - Release Date: 4/12/2006 ___ --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 nat bug
hi, can you someone explain this bug? (or point me to number from bugs.digium.com) 2006-03-28 19:07 + [r15699] Olle Johansson [EMAIL PROTECTED] * channels/chan_sip.c: Fix breakage of NAT support for peers with qualify=yes. Thanks Damin for access to your system, sorry folks. thanks --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --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] Music on hold problem
Don Pobanz wrote: Daniel Korndorfer wrote: i'm having problems with the MOH module. In a queue sometimes it just stop playing, does anyone have some idea what could be wrong? No verbose data... Occasionally I am seeing music on hold stop playing for parked calls, so this isn't unique for queues. Maybe it is as Daniel said something to do with MOH. I am running 1.2.6. I have an extension set up that just dumps the caller into MusicOnHold, which I use for testing new MOH music. I've noticed a similar thing with this, that sometimes it finished playing one track, and just doesn't start the next. Alex -- Alex Brett [EMAIL PROTECTED] http://www.loho.co.uk/ ___ --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)
Currently, compiling asterisk on an Itanium fails with the GSM codec.
All I could find on Google was a hack to basically remove GSM from the
build which is not an option for some. This patch will allow it to
compile and seems to work perfectly.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
--- Makefile2006-03-12 12:57:37.0 -0500
+++ ../../../../asterisk-1.2.6/codecs/gsm/Makefile 2006-04-12
15:11:19.0 -0400
@@ -45,6 +45,7 @@
ifneq ($(shell uname -m),ppc64)
ifneq ($(shell uname -m),alpha)
ifneq ($(shell uname -m),armv4l)
+ifneq ($(shell uname -m),ia64)
ifneq (${PROC},sparc64)
ifneq (${PROC},arm)
ifneq (${PROC},ppc)
@@ -62,6 +63,7 @@
endif
endif
endif
+endif
#The problem with sparc is the best stuff is in newer versions of gcc
(post 3.0) only.
#This works for even old (2.96) versions of gcc and provides a small
boost either way.
@@ -233,6 +235,7 @@
ifneq ($(shell uname -m),ppc)
ifneq ($(shell uname -m),ppc64)
ifneq ($(shell uname -m),alpha)
+ifneq ($(shell uname -m),ia64)
ifneq ($(shell uname -m),armv4l)
ifneq ($(shell uname -m),sparc64)
ifneq (${PROC},arm)
@@ -247,6 +250,7 @@
endif
endif
endif
+endif
TOAST_SOURCES = $(SRC)/toast.c \
$(SRC)/toast_lin.c \
@@ -297,6 +301,7 @@
ifneq ($(shell uname -m), ppc)
ifneq ($(shell uname -m), ppc64)
ifneq ($(shell uname -m), alpha)
+ifneq ($(shell uname -m), ia64)
ifneq ($(shell uname -m), sparc64)
ifneq ($(shell uname -m), armv4l)
ifneq ($(shell uname -m), parisc)
@@ -309,6 +314,7 @@
endif
endif
endif
+endif
TOAST_OBJECTS =$(SRC)/toast.o \
$(SRC)/toast_lin.o \
___
--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] app_meetme.so
Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian ___ --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] free tollfree termination
I seem to be having the same problems. Is anyone from trxtel reading this? I guess you get what you pay for :) - Waldo On Apr 13, 2006, at 6:36 AM, Gustavo Hernandez wrote: Hi ! Anybody know if 1800 free termination services from trxtel are in troubles? I can´t reach it, and don´t know why. Thanks a lot gus At 06:49 a.m. 26/03/2006, you wrote: Hi there, Thanks for the tip ! I am happily using this service now. One question though : I cannot get DTMF to work. Is there anything I can do in my asterisk setup to fix this ? Thanks, Lukas trixter aka Bret McDanel wrote: http://www.trxtel.com/index.php?page=Tollfree_Termination This is a free service, I am not selling anything with this service. I just thought that individuals that read this list may enjoy getting tollfree access free this way (yet another way) given that it lets you send your caller id and some of the other free providers dont let you do that. Starting a test service now, for individuals free north american tollfree termination. For carriers that do large quantities of minutes (a not really defined term as yet, more a negotiated value) we will share revenue with you for sending calls to us. If you set up IP PBX systems for customers, add a route in and make residuals off those customers. Run a ITSP? Get revenue for each minute that a customer dials a north american toll free. If anyone has any problems using the service I would appreciate hearing about it, the service will remain free even after the test period, however to get compensation requires an account so that it can be uniquely tracked. Granted tollfree traffic isnt usually the bulk of a provider, but at least now you can provide it free to your customers without losing on costs like bandwidth :) ___ --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] fxotune error
Hi, I have an asterisk 1.2.1 on a debian box with a tdm400p card and a monoBRI card. I tryed to use fxotune: turned off asterisk leaving modules active as seen from lsmod: zaptel224132 1 wctdm crc_ccitt 2432 1 zaptel I launched fxotune: *pbxtest:/etc/asterisk# /usr/src/zaptel-1.2.1/fxotune -i 4* but fxotune displays these messages: *Tuning module 1Skipping non-TDM / non-FXO Failure! Could not fill input buffer Tuning module 2Failure!* What does they mean?? TIA Giorgio Incantalupo ___ --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: Fwd: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording
At 01:19 PM 04/11/2006, you wrote: The last point also brings up a question. Does anyone know how gracefully Asterisk handles attempting to write leg files to a full disk? For some number of days my * box was running with the disk set to read only and I only discovered it when I noticed some very odd error messages. Never bothered the system and nothing about the way the phones worked ever gave us any indication of the problem. A dying disk caused the system to mount the drive read only. I'm not intentionally logging anything, but the system does write occasional entries to the log files and the inability to do that didn't seem to cause problems. Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.385 / Virus Database: 268.4.1/310 - Release Date: 04/12/2006 ___ --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] app_meetme.so
Un-comment ztdummy and build re-zaptel and then re-build asterisk Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Sebastian Milioto [mailto:[EMAIL PROTECTED] Sent: Thursday, April 13, 2006 10:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] app_meetme.so Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian ___ --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] Music on hold problem
Are you guys using native music on hold, or MP3 music on hold? On Thu, 13 Apr 2006 10:09:26 -0400, Alex Brett [EMAIL PROTECTED] wrote: Don Pobanz wrote: Daniel Korndorfer wrote: i'm having problems with the MOH module. In a queue sometimes it just stop playing, does anyone have some idea what could be wrong? No verbose data... Occasionally I am seeing music on hold stop playing for parked calls, so this isn't unique for queues. Maybe it is as Daniel said something to do with MOH. I am running 1.2.6. I have an extension set up that just dumps the caller into MusicOnHold, which I use for testing new MOH music. I've noticed a similar thing with this, that sometimes it finished playing one track, and just doesn't start the next. Alex -- Justin Tunney ___ --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] app_meetme.so
Mmmm.. sorry... I'm rookie. Coudl you be a litlle more specific please? Where I have to uncomment ztdummy? On 4/13/06, Steve Totaro [EMAIL PROTECTED] wrote: Un-comment ztdummy and build re-zaptel and then re-build asterisk Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Sebastian Milioto [mailto:[EMAIL PROTECTED] Sent: Thursday, April 13, 2006 10:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] app_meetme.so Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian ___ --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] app_meetme.so
Google is your friend. http://www.google.com/search?sourceid=navclientie=UTF-8rls=GGLD,GGLD:2 004-48,GGLD:enq=uncomment+ztdummy Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Sebastian Milioto [mailto:[EMAIL PROTECTED] Sent: Thursday, April 13, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] app_meetme.so Mmmm.. sorry... I'm rookie. Coudl you be a litlle more specific please? Where I have to uncomment ztdummy? On 4/13/06, Steve Totaro [EMAIL PROTECTED] wrote: Un-comment ztdummy and build re-zaptel and then re-build asterisk Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Sebastian Milioto [mailto:[EMAIL PROTECTED] Sent: Thursday, April 13, 2006 10:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] app_meetme.so Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian ___ --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] sip nat bug
marek cervenka wrote: can you someone explain this bug? (or point me to number from bugs.digium.com) 2006-03-28 19:07 + [r15699] Olle Johansson [EMAIL PROTECTED] * channels/chan_sip.c: Fix breakage of NAT support for peers with qualify=yes. Thanks Damin for access to your system, sorry folks. There is nothing to explain. A change was committed that broke NAT support, and the next day it was corrected. There was no bug report in Mantis. ___ --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] OT: MWI on Treo 600/650
On 4/13/06, David Cook [EMAIL PROTECTED] wrote: My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. dbc. Have a look at http://www.bayhamsystems.com/asterisk.html there are some mention of MWI etc on there. 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 ___ --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] TE410P upgrade to TE411P = (solution to) no more fax carrier detection !
George Pajari wrote: For the moment, if you need FAX tone detection, you will need to use 'vpmdtmfsupport=0' in your module configuration for loading the wct4xxp module; this will not disable the echo canceler, just stop using it for tone detection. Any idea if/when this will be addressed? We had been going with the VPM modules to get better DTMF detection and echo can but if we have to move DTMF detection back to Asterisk in order to get CNG detection then we're concerned. (We haven't added fax detection to our systems yet but it is planned.) It is in our engineering department's hands to address; I can't give you any expected timeframe for a fix, though... just that it is pretty high priority. ___ --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] playback soundfile in memory
Akpome Akpoguma wrote: I want to playback sound file loaded in memory not from a file...is this possible? Akpome, If the sound file is being played more than once, there is a good chance that this is already happening. At one point, our production system had 100 calls in queue. Each call had pre-queue announcements from the dialplan, native MOH, and in-queue announcements. I ran iostat on the system, and there was no disk read activity at all. I believe this can be accounted for by Linux's file caching. If you run iostat on your system and see read activity, give this a try: 1) Set up a RAM disk 2) Build an init-script that copies your sound files from the hard disk to the RAM disk on boot 3) Configure Asterisk to play the files from the RAM disk Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Display Confideltial or unknown on called id display
Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display Confidential or unknown as we sometimes see ? Andre ___ --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: OT: MWI on Treo 600/650
I've been working on this off and on for AGES. There are some SMS portal sites that claim to be able to do this as well, but I have not managed to find one. I had found a company called bahamasystems which has an asterisk interface but it's a service and it's expensive. Another poster pointed me at nowsms.com. Looks a little more attractive (except for the Windows gateway part). However, I have not been able to find out the actual codes. Just the 111# stuff to get a return receipt, etc. ___ --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] Music on hold problem
Justin Tunney wrote: Are you guys using native music on hold, or MP3 music on hold? I believe I am using MP3. My musiconhold.conf file looks like this [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 -- Don Pobanz ___ --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] NAT/STUN Server
Hi, I am trying to register SIP clients which are behind NAT on different network. In order to achieve this goal I think I need STUN Server . I downloaded STUN Server from http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz But I don't know how to install/configure it. And please advice me that STUN server is good idea for this scenario? Thanks in advance Wazb ___ --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] Display Confideltial or unknown on called id display
Andre Courchesne - Consultant wrote: Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display Confidential or unknown as we sometimes see ? In the US, you can't. Its the responsibility of the terminating central office to do a database lookup to obtain the callerid name. ___ --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] callerid name inboune from PRI
It is in fact required for some implementations of callerid name. It comes on a facility message that arrives after the call is setup. It also can come in a display IE in the call setup. It really depends on which way they are sending it. Matthew Fredrickson On Apr 11, 2006, at 12:49 PM, C F wrote: No, I'm taking receiving CallerID name and *not* sending. and no on a PRI wait should not be required for callerID to come in. On 4/11/06, Jerry Jones [EMAIL PROTECTED] wrote: I CAN VERIFY via aa dozen PRI from XO that yes indeed provide incoming callerID on PRI. It arrives shortly after the setup message. Hence the wait(1) required to display it. Now if you are referring to sending caller name across PSTN - that does NOT work since the terminating switch will do a CNAM lookup. On Apr 10, 2006, at 10:55 PM, C F wrote: On 4/10/06, Andres [EMAIL PROTECTED] wrote: Steven wrote: I switched PRI vendors recently, and one of my questions was do you provide caller ID name in addition to number? ATT Local did not, But XO communications said they did. You heard wrong. We have multiple PRIs from XO and they DO NOT send caller name. We have discussed the issue with them on several ocassions. The sales people will say whatever they want, but the tech people who actually work in the switches know that caller name is not supported. I believe he didn't hear wrong, a lot of providers are now providing CallerID with name over PRI. The tech people know it *is* supported (all it is is a stupid simple lookup on the SS7 side of the equipment) but wasn't done until now because the competition didn't offer it, but now that the competition is offering it, everyone else is as well. Before I call to complain, is there an setting to turn this on in asterisk? I want to make sure that I have my side covered before I call XO. My current zaptel.conf is: context=from-pstn switchtype=national pridialplan=unknown prilocaldialplan=unknown priindication = outofband signalling=pri_cpe usecallerid=yes hidecallerid=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no group=0 callgroup=1 pickupgroup=1 accountcode=I musiconhold=default channel = 1-23 -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Display Confideltial or unknown on called iddisplay
Hi, Have you try to set hidecallerid=yes in zapata.conf? Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Courchesne - Consultant Sent: Friday, April 14, 2006 12:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Display Confideltial or unknown on called iddisplay Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display Confidential or unknown as we sometimes see ? Andre ___ --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] Display Confideltial or unknown on called iddisplay
Prepend *67 if your carrier allows it Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Andre Courchesne - Consultant [mailto:[EMAIL PROTECTED] Sent: Thursday, April 13, 2006 12:02 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Display Confideltial or unknown on called iddisplay Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display Confidential or unknown as we sometimes see ? Andre ___ --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] Display Confideltial or unknown on calledid display
Maybe hidecallerid=yes in Zapata.conf Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Thursday, April 13, 2006 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Display Confideltial or unknown on calledid display Andre Courchesne - Consultant wrote: Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display Confidential or unknown as we sometimes see ? In the US, you can't. Its the responsibility of the terminating central office to do a database lookup to obtain the callerid name. ___ --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] Display Confideltial or unknown on called id display
Rich Adamson wrote: In the US, you can't. Yes, you can. You just set the 'presentation' bits to show that the number is not known or is restricted. However, you can't control the actual words that show up on the recipient's device instead of the CNAM... ___ --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] Display Confideltial or unknown on called id display
If you mean to have a private caller ID, I would think that the phone company would need to update you record in the database. On 4/13/06, Andre Courchesne - Consultant [EMAIL PROTECTED] wrote: Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display Confidential or unknown as we sometimes see ? Andre ___ --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] - [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
Re: [Asterisk-Users] Music on hold problem
Don Pobanz wrote: I believe I am using MP3. My musiconhold.conf file looks like this [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 Daniel and Don, Try switching to native MOH. You'll eliminate the decoding of the MP3s and the host of problems that come along with using mpg123. The MOH is handled by the same thread that's handling the call, so you should see an overall performance benefit. Memory usage may go up a little, but I don't think many Asterisk boxes are memory bound. Here's a guide to converting your MOH files: http://www.oinko.net/astrecipes/index.php?n=152 Since you're already running Asterisk 1.2, I don't see any reason not to take advantage of this feature. Please give it a shot and report your results back to the list. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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/STUN Server
Just point your ATA to stun.fwdnet.net; it is a free service by free world dialup. Sure beats creating your own stun server. If you do need to create your own stun server, I suggest http://www.voip-info.org/wiki/view/Vovida.org+STUN+server -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wasif Sent: Thursday, April 13, 2006 11:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] NAT/STUN Server Hi, I am trying to register SIP clients which are behind NAT on different network. In order to achieve this goal I think I need STUN Server . I downloaded STUN Server from http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz But I don't know how to install/configure it. And please advice me that STUN server is good idea for this scenario? Thanks in advance Wazb ___ --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 __ NOD32 1.1476 (20060407) 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
[Asterisk-Users] Segfault on Inbound call?
I'm running asterisk 1.2.6 with Sangoma A200 4 port FXO card. Outbound calls are working fine. However, when I have an inbound call.. asterisk will segfault.. and then start again ... then it will take 1 call fine I'm running asterisk with a -U and -G of asterisk. Any thoughts? ___ --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] Display Confideltial or unknown on callediddisplay
Here is what you would do for a sip call. I'm sure it isn't much different
for a PRI call.
[cid-block]
exten = _*67.,1,Ringing
exten = _*67.,2,goto(dial-cid-block,${EXTEN:3},1)
[dial-cid-block]
exten = _XXX,1,Macro(cid-block,1${CALLERIDNUM:-10:3}${EXTEN})
exten = _1NXXNXX,1,Macro(cid-block,${EXTEN:1})
exten = _NXXNXX,1,Macro(cid-block,1${EXTEN})
exten = _011.,1,goto(restricted,${EXTEN},1)
[macro-cid-block]
exten = s,1,SetCallerID(*)
exten = s,2,Dial(SIP/[EMAIL PROTECTED])
exten = s,3,hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Thursday, April 13, 2006 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Display Confideltial or unknown on
callediddisplay
Prepend *67 if your carrier allows it
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: Andre Courchesne - Consultant [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 13, 2006 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Display Confideltial or unknown on
called
iddisplay
Hi,
When making a call from an Asterisk box over a PRI connection, I am
able to set the Caller ID phone number to what ever I want. This works
find.
How to I make the called party callerid display Confidential or
unknown as we sometimes see ?
Andre
___
--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
__ NOD32 1.1476 (20060407) 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
[Asterisk-Users] DTMF Not working for only one number
Anyone have any ideas why DTMF would not work on only one number? Looking through the logs, anytime a button is pressed, this is what shows up: 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Echo cancellation already on We don't have any problems with any other numbers, and the DTMF works on phones that are on our meridian system (which we currently pass through) so I know it's somewhere in our T1 lines or gateway... don't really know where to go from there. I did turn vpmdtmfsupport off and that didn't help at all. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] ast_sched_runq ran 281 scheduled tasks all at once
We get these messages too, but they don't seem to cause any problems. Are you connecting 2 * (with different versions) via IAX2? Are these messages only appear on the lower version one? I asked a similar question on the list, and the suggestion was to upgrade them to the same version. Hope this helps. Let us know how it goes. Andy On 4/13/06, Gareth Blades [EMAIL PROTECTED] wrote: Just noticed that I occasionally get these messages:- Apr 12 09:27:03 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 281 scheduled tasks all at once Apr 13 09:13:18 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 1987 scheduled tasks all at once Apr 13 12:47:56 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 1804 scheduled tasks all at once Are they anything to be concerned about? ___ --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] Display Confideltial or unknown on called id display
Kevin P. Fleming wrote: Rich Adamson wrote: In the US, you can't. Yes, you can. You just set the 'presentation' bits to show that the number is not known or is restricted. However, you can't control the actual words that show up on the recipient's device instead of the CNAM... Ops, I read it verbatim. Wanted to set the words Confidential or Unknown. ___ --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] Sipura 2100
I am wondering if anyone has sample XML config for the Sipura 2100 ATA. We have been autoprovisioning our 2002s with success and the 2100's take the same XML that we have come up with, but I am not sure of the syntax for specific things that I need these boxes to do, such as turning T.38 on. If anyone is willing to share their xml for autoprovisioning Sipura 2100's, it would be much appreciated. Best, -- Darrell S. Long BestWeb Corporation ___ --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] placing call with agi
I'm trying to set up a system so that I can record a conversation over SIP. Monitor and the like don't work so well for me, because I need to pipe the conversation to other programs in realtime, rather than record to a file, so I've been trying to use EAGI instead. (if anyone has any other suggestions about this, it would be greatly appreciated!) At this point, I'm a little stumped about getting the AGI script to dial out though - http://www.voip-info.org/wiki-Asterisk+AGI explains that you can't just dial out, since the script then disconnects from asterisk. I've been attempting the auto-dial call file method that that page links to, but I don't really understand how it's supposed to work. How can I connect the new call to the existing call? It seems that this method is just for starting a call from scratch, but I've already called an extension on the asterisk server which ran the eagi script in the first place. Can the new outgoing connection be attached to that call? I'm not sure if my description will make sense to anyone else, but please let me know if there's any way I can clarify things! Thanks! -- Jon-o Addleman - http://redowl.dyndns.org ___ --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 register question
I am trying to link an asterisk box up to a SIP server on the same subnet. The SIP server does not have a password (and is locked down by IP number 'allow'). How do I specify this on the register line? Based on the documentation, the line looks like this: register = user[:secret[:[EMAIL PROTECTED]:port][/extension] It looks like [EMAIL PROTECTED] is the minimum required. Is there anyway to specify a username of null, or something? Thanks in advance! 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
[Asterisk-Users] Asterisk 1.2.7 Page()
I just upgraded to Asterisk 1.2.7 from 1.2.5. Page() is behaving differently. I'm getting an error - Incomplete destination '' supplied. -- Executing Page(SIP/2944093-5999, SIP/3254107SIP/3254105|) in new stack Apr 13 11:06:11 WARNING[9294]: app_page.c:193 page_exec: Incomplete destination '' supplied. -- Playing 'beep' (language 'en') -- Created MeetMe conference 1023 for conference '1592290043d' -- Hungup 'Zap/pseudo-1092580194' == Spawn extension (oneeighty_start, 1000, 1) exited non-zero on 'SIP/2944093-5999' Also... the recipients, 3254107 and 3254105 are not being bridged into the meetme conference. Anyone? 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] call center running Asterisk -sound quality-critical!
Wai, Please explain how the in and out channels are mixed first before they are written to the disk using monitor with no mixing onto the scsi drive. I'd love to implement this on our system to cut in half the I/O associated with Monitor(). Also, what bug does MixMonitor() have? It is my understanding that MixMonitor() is based on ChanSpy() and we seem to be having an issue with ChanSpy() where the legs of a call fall out of synch. My hunch is that it has to do with a caller being muted or placed on hold. Do these issues seem related? Just bumping this in case you missed it the first time. It's easy to do with a list as busy as this one. I'm sure everyone using Monitor() would love to hear how you're doing this. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Set language in Asterisk auto-dial out
Hello,
I use .call files in /var/spool/asterisk/outgoing to initiate calls
automatically. And I'd like to setup the language used for the call in
this file but I haven't found any way of doing this. I tried something
like Set: language=fr, Set: ${LANGUAGE}=fr, ... but nothing worked.
Is that possible?
--
Benoît Mérouze
Ingénieur Dévéloppement d'Application Réseau
[EMAIL PROTECTED]
___
--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] Help Cas Circuit
Hi I need help in setting up the CAS circuit using Asterisk and Digium Dual Port T1 card. I tried it but without any luck. Any one have experienced the problem with Feature Group D on Asterisk 1.2.6 Thanks Ali ___ --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 1.2.7 Page()
Douglas Garstang wrote: I just upgraded to Asterisk 1.2.7 from 1.2.5. Page() is behaving differently. I'm getting an error - Incomplete destination '' supplied. This was a bug introduced in 1.2.7. I have just fixed it in Subversion, so you can update to the latest branch-1.2 code from there if you wish. We will get a corrected tarball release out shortly. Sorry for the inconvenience. ___ --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] AAstra 9133i register double account.. ??
Check your 'GENERAL' section and your 'LINE' sections.. you probably only changed in one place. On 4/13/06, William Harrison [EMAIL PROTECTED] wrote: If the phones are registering twice, and you are 100% sure they are only configured via the WebUI, then you must have the settings in two places. There are SIP configs for Global SIP, and also for every available line. You may have one set of settings in Global SIP, and a different one for one of the lines, and thus have two registrations. To be sure that you don't have any latent config items, you may want to factory default the phone, and start from scratch. The .cfg files are simple text config files that can be downloaded from a TFTP/FTP/HTTP server. This is all explained in the Administrator's Guide available from Aastra's website: http://www.aastra.com/support/enterpriseip/show_manuals.asp ___ --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] Ztmonitor shows RX is always on.
Details:Asterisk 1.0.9 Zaptel 1.0 Dell P3 1ghz with X100P Clone Location: IndiaThis is an interesting issue where when I open up ZTMonitor, it shows the RX as being on. It seems that Zaptel doesn't know to hang up the line so after a couple of hours when the telecom cuts the line, everythign stops working. Things I've tried include playing with the zaptel.conf, trying zaptel v1.2 (with Asterisk 1.0.9), and trying loopstart and kewlstart. Could I have a setting wrong in my extensions.conf or is this just a problem with the Indian phone lines? Regards, Min *Zaptel.conf*fxsks=1loadzone = usdefaultzone = uszapata.conf[channels]context=incomingsignalling=fxs_ks language=enrxwink=300 ; Atlas seems to use long (250ms) winksusedistinctiveringdetection=nobusydetect=yesbusycount=6callprogress=yesimmediate=nocallwaiting=nocallwaitingcallerid=no threewaycalling=notransfer=nocancallforward=nocallreturn=nouseincomingcalleridonzaptransfer=nocallerid=asreceivedusecallerid=norelaxdtmf=noechocancel=yesechocancelwhenbridged=yesechotraining=800 rxgain=0.0txgain=0.0group=0callgroup=1pickupgroup=1faxdetect=nochannel = 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
[Asterisk-Users] Asterisk 1.2.7.1 Released
The Asterisk Development Team has released version 1.2.7.1 of Asterisk. This release contains only two fixes, one of which is that the Page() application was entirely broken in version 1.2.7. If you have already upgraded to 1.2.7 and you do not use the Page() application in your dialplan, there is no need to upgrade to version 1.2.7.1. The release is available on the Digium FTP servers as PGP signed tarballs and also as PGP signed patch files, to ease upgrading from the previous versions. The keys used to sign these files can be verified by using the keyserver at pgp.mit.edu. Thanks for your support of Asterisk! ___ --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] Early Media Enable?
Hi, I've searched almost everywhere but have not come across a solution so I was hoping one of your fine folks can help me out. The problem is that a carrier is passing me early media on calls that sometimes have problems connecting. For example, calls to India mobile might play an early media message saying the phone is out of reach if mobile is out of area of coverage. Problem is that asterisk does not play this early media message and simply continues to ring indefinitely. Now I know asterisk will not open the audio streams till it gets acks from both sides but is there a way around it? To open one way audio right away? Any solution for this problem? Thanks for any help in advance. Regards, Mohammed Salim ___ --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] Static on ZAP channels
I have a TDM2400P with hardware echo cancel. We seem to have static on some calls but not others and the receive audio appears 'choppy'. Transmit side works fine and does not have any audio problems. I had to turn up the RX gain to 18 or the receive audio volume is too low. Can anyone shed some light? Thanks. TJ ___ --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] Static on ZAP channels
Tim Jackson wrote: I have a TDM2400P with hardware echo cancel. We seem to have static on some calls but not others and the receive audio appears 'choppy'. Transmit side works fine and does not have any audio problems. I had to turn up the RX gain to 18 or the receive audio volume is too low. Since this product is under warranty, you are far better off contacting Digium Support than asking the mailing list for help :-) ___ --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] Static on ZAP channels
On 4/13/06, Tim Jackson [EMAIL PROTECTED] wrote: I have a TDM2400P with hardware echo cancel. We seem to have static on some calls but not others and the receive audio appears 'choppy'. Transmit side works fine and does not have any audio problems. I had to turn up the RX gain to 18 or the receive audio volume is too low. Can anyone shed some light? Contact Digium Support. They should be able to assist you. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] call center running Asterisk -soundquality-critical!
I did not install soxmix in my linux box. If you having issues with mixmonitor, you can put both legs of the call into a conference and record the conference -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth Sent: Thursday, April 13, 2006 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -soundquality-critical! Wai, Please explain how the in and out channels are mixed first before they are written to the disk using monitor with no mixing onto the scsi drive. I'd love to implement this on our system to cut in half the I/O associated with Monitor(). Also, what bug does MixMonitor() have? It is my understanding that MixMonitor() is based on ChanSpy() and we seem to be having an issue with ChanSpy() where the legs of a call fall out of synch. My hunch is that it has to do with a caller being muted or placed on hold. Do these issues seem related? Just bumping this in case you missed it the first time. It's easy to do with a list as busy as this one. I'm sure everyone using Monitor() would love to hear how you're doing this. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Early Media Enable?
Early audio is played, as long as you do not have a r in your Dial statement. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mohammed Salim Sent: April 13, 2006 2:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Early Media Enable? Hi, I've searched almost everywhere but have not come across a solution so I was hoping one of your fine folks can help me out. The problem is that a carrier is passing me early media on calls that sometimes have problems connecting. For example, calls to India mobile might play an early media message saying the phone is out of reach if mobile is out of area of coverage. Problem is that asterisk does not play this early media message and simply continues to ring indefinitely. Now I know asterisk will not open the audio streams till it gets acks from both sides but is there a way around it? To open one way audio right away? Any solution for this problem? Thanks for any help in advance. Regards, Mohammed Salim ___ --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] CANADA 911 Update
ESWG Consensus 12-month Report on Nomadic VoIP Technical and Operating Impediments to 9-1-1/E9-1-1 Service Delivery in Canada DRAFT Executive Summary Emergency Services Working Group (ESWG) recommends on a consensus basis the Commission order the deployment of NENA Internet-2 (i2) compliant emergency services components, systems and upgrades to result in the operation within 18 months of enhanced 9-1-1 services for nomadic and fixed/non-native VoIP callers in Canada. ESWG also recommends that the Commission establish for planning purposes a milestone for the transition of all legacy analogue emergency services networks to IP-based emergency networks (so called next generation 9-1-1 networks) in Canada no sooner than 36 months after the deployment of i2. ESWG further recommends that the Commission order eight specific tasks with sequential milestones to assist with the orderly deployment of i2: 1. CISC should be ordered to deliver within 6 months a preferred PSAP funding model for VoIP E9-1-1 addressing regional/provincial variances and practices to produce a common national standard. 2. CISC should be ordered to deliver a comprehensive architecture for the implementation of VoIP E9-1-1 to deliver within 9 months specifying roles and responsibilities of all emergency services industry participants. 3. All 9-1-1 Service Providers ordered to provide MSAG for the purposes of LIS validity checking within 12 months subject to amended agreements. 4. All Broadband Internet Service Providers be ordered to provide LIS capability within 12 months at their own expense. 5. All 9-1-1 Service Providers be ordered to provide ALI/ANI capability consistent with NENA i2 implementation within 15 months at their own expense. 6. All local VoIP service providers be ordered to provide Call Servers and/or Proxy Gateway capability within 15 months at their own expense. 7. All 9-1-1 Service Providers be ordered to provide ESGW capability within 15 months at their own expense. 8. All VoIP 9-1-1 calls to be E9-1-1 delivered to the correct PSAP within 18 months (Full Production). ESWG also recommends the establishment of at least one pilot program / test region in Canada to evaluate and determine the best method and practices for transition from legacy to IP emergency services. Finally, ESWG requests Commission continue their practise of fostering advancement in emergency services by providing deadlines for the accomplishment of specific tasks through decisions and order the commencement of this deployment as quickly as is prudent. 1 Background 1.1 Decision CRTC 2005-21 Mandate This Emergency Services Working Group (ESWG) Consensus 12-month Report on Nomadic VoIP Technical and Operating Impediments to 9-1-1/E9-1-1 Service Delivery in Canada (the 12-month Report or the Report) is in response to the mandate given to CRTC Interconnection Steering Committee (CISC) by the Commission in Telecom Decision CRTC 2005-21 as follows: 72. The Commission remains of the view that, as these are technical and operational issues, the most effective approach to resolving them is through the CISC process, provided that CISC is guided by a fixed timeline. 73. Accordingly, the Commission requests CISC to submit to the Commission, within six months from the date of this Decision, a report identifying the technical and operational issues that impede 9-1-1/E9-1-1 service delivery when local VoIP service is offered on a fixed/non-native basis, and, within one year from the date of this Decision, a similar report with respect to local VoIP service offered on a nomadic basis. Each report should identify all viable solutions and recommend the preferred solution(s), with supporting rationale, and a proposed timeframe for implementation. [Emphasis added] 74. The Commission notes that certain parties suggested that CISC may benefit from participation in the NENA process in the United States. The Commission recognizes that the progress made by other national telecommunications regulators, with respect to the provisioning of emergency services with local VoIP services, may be of value to the Canadian industry and encourages CISC to monitor the reports and progress being made in other jurisdictions on this important issue. This 12-month Report follows up upon the issues identified in the ESWG 6-month Report on Fixed/Non-Native VoIP Technical and Operating Impediments to 9-1-1/E9-1-1 Service Delivery (the 6-month Report) as it was the conclusion of ESWG that the impediments in Canada were common between the Fixed/Non-native and Nomadic VoIP 9-1-1/E9-1-1 service delivery. In addition, this Report lays out the careful monitoring of the US-based National Emergency Number Association (NENA) process done by ESWG as well as the monitoring and contrast of the regulatory environment in the United States provided by the Federal Communications Commission (FCC) used to guide the development of the Report. 1.2 ESWG 6-month Report on
Re: [Asterisk-Users] transforming g729 files to wav files
Tofik Suleymanov wrote: Darrell Long wrote: The resulting file is not going to sound any better and its going to take up more space. What is the reason you need a WAV file? Perhaps there is a more efficient way to do what you are trying to do. Darrell S. Long BestWeb Corporation I understand issues about sound quality.Here is the situation: i am using g729-native sound files and g729 codecs everywhere.My voicemail is coming in g729 format also.Some time ago one of our customers asked for the voicemail to go to his e-mail and i want him to recieve just a .wav file. I've also tried to use: format=g729|wav in my voicemail.conf in order to have copies of voicemails in wav format but for unknown reason (after this change) i wasnt able to hear voicemail announcements when trying to access voicemail. http://redice.krisk.org P.S. - GX Transcoder has some audio quality problems. -- Kristian Kielhofner ___ --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 1.2.7 Page()
Hi Douglas. The Asterisk Development Team has released version 1.2.7.1 of Asterisk.This release contains only two fixes, one of which is that the Page() application was entirely broken in version 1.2.7. If you have alreadyupgraded to 1.2.7 and you do not use the Page() application in yourdialplan, there is no need to upgrade to version 1.2.7.1.The release is available on the Digium FTP servers as PGP signedtarballs and also as PGP signed patch files, to ease upgrading from theprevious versions. The keys used to sign these files can be verified by using the keyserver at pgp.mit.edu.Greatings Josue 2006/4/13, Kevin P. Fleming [EMAIL PROTECTED]: Douglas Garstang wrote: I just upgraded to Asterisk 1.2.7 from 1.2.5. Page() is behaving differently. I'm getting an error - Incomplete destination '' supplied.This was a bug introduced in 1.2.7. I have just fixed it in Subversion,so you can update to the latest branch-1.2 code from there if you wish. We will get a corrected tarball release out shortly. Sorry for theinconvenience.___--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] Asterisk 1.2.7 Page()
Thanks. I've upgraded. -Original Message-From: Josué Conti [mailto:[EMAIL PROTECTED]Sent: Thursday, April 13, 2006 1:48 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk 1.2.7 Page() Hi Douglas. The Asterisk Development Team has released version 1.2.7.1 of Asterisk.This release contains only two fixes, one of which is that the Page() application was entirely broken in version 1.2.7. If you have alreadyupgraded to 1.2.7 and you do not use the Page() application in yourdialplan, there is no need to upgrade to version 1.2.7.1.The release is available on the Digium FTP servers as PGP signedtarballs and also as PGP signed patch files, to ease upgrading from theprevious versions. The keys used to sign these files can be verified by using the keyserver at pgp.mit.edu.Greatings Josue 2006/4/13, Kevin P. Fleming [EMAIL PROTECTED]: Douglas Garstang wrote: I just upgraded to Asterisk 1.2.7 from 1.2.5. Page() is behaving differently. I'm getting an error - Incomplete destination '' supplied.This was a bug introduced in 1.2.7. I have just fixed it in Subversion,so you can update to the latest branch-1.2 code from there if you wish. We will get a corrected tarball release out shortly. Sorry for theinconvenience.___--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] DTMF sensitivity
List, I have recently downloaded installed Asterisk2Billing from http://www.asterisk2billing.org/ which is a great billing program for prepaid calling cards as well as SIP/IAX users. The problem is that our Asterisk server seems to have DTMF sensitivity too high. If you dial 123456789 it might pick up 112233456789. I believe that this is set by asterisk, not the a2billing program right? Is there a way to adjust the sensitivity of this? Thanks, William Piper ___ --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] call center running Asterisk-soundquality-critical!
I just check the source code, Monitor uses ast_writestream and it eventurally goes down to au_write, g723_write, etc. They don't commit to the disk. So, in effect, if you have a lot of ram, the audio should stay in ram until it gets swap out or the file is closed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth Sent: Thursday, April 13, 2006 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -soundquality-critical! Wai, Please explain how the in and out channels are mixed first before they are written to the disk using monitor with no mixing onto the scsi drive. I'd love to implement this on our system to cut in half the I/O associated with Monitor(). Also, what bug does MixMonitor() have? It is my understanding that MixMonitor() is based on ChanSpy() and we seem to be having an issue with ChanSpy() where the legs of a call fall out of synch. My hunch is that it has to do with a caller being muted or placed on hold. Do these issues seem related? Just bumping this in case you missed it the first time. It's easy to do with a list as busy as this one. I'm sure everyone using Monitor() would love to hear how you're doing this. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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
AW: [Asterisk-Users] NAT/STUN Server
Hi, Good: Setting up an STUN is easy. Bad: I have only a link to an german tuto-site. (http://www.asteriskpbx.de/index.php?stun) You need at least 2 network-cards. Get the File: wget http://mesh.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz unpack: # tar zxf stund_0.96_Aug13.tgz # cd stund # make all If you got an error, it may be that the openssl c++ packages are missing. The install: openssl-devel and gcc-c++ If the certificate is missing for c++, you can import it here: http://ftp.upce.cz/centos/3.3/os/s390x/RPM-GPG-KEY-CentOS-3 Till -Ursprüngliche Nachricht- Von: Wasif [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 13. April 2006 17:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] NAT/STUN Server Hi, I am trying to register SIP clients which are behind NAT on different network. In order to achieve this goal I think I need STUN Server . I downloaded STUN Server from http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz But I don't know how to install/configure it. And please advice me that STUN server is good idea for this scenario? Thanks in advance Wazb ___ --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] Music on hold problem
Thank you Matt!!! Matt Roth wrote: Try switching to native MOH. You'll eliminate the decoding of the MP3s and the host of problems that come along with using mpg123. The MOH is handled by the same thread that's handling the call, so you should see an overall performance benefit. Memory usage may go up a little, but I don't think many Asterisk boxes are memory bound. I did not have mpg123 installed so I do not know what was being used for moh. Regardless my moh audio stopping problem was resolved by following the recipe you showed me at http://www.oinko.net/astrecipes/index.php?n=152 Thanks Matt Roth! Don Pobanz ___ --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] spa-942 support Page() / Intercom correctly?
Looking to possibly use the spa-942 in a business environment as a medium class sip phone. Customer absolutely wants support for Page() or Intercom. Does anyone know if this phone truly handles Page() with two-way audio correctly? ___ --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] Performance: Xeon or Opteron?
Has anybody used the sangoma fxo cards with asterisk? Anybody using multiple cards? Problems with irq and such (same as with digium ones)? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |John Novack |Sent: Wednesday, April 12, 2006 10:29 AM |To: [EMAIL PROTECTED] |Cc: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? | | | |Rich Adamson wrote: | | | While talking with one of the sangoma folks very recently, he was | rather emphatic the pci bus was designed to share |interrupts. I was | a little concerned as a test server had the wanpipe driver |sharing an | interrupt with libata and uhc1_hcd. His comment was that's the way | its suppose to work, sharing interrupts as needed. I've not had any | recognizable issues with the A200D card at all, and faxing |via a A200D | fxs port to a A200D fxo (pstn) port functions 100% reliably. | | What that would suggest is the TDM400 pci firmware (whether on card | logic or whatever) is the source of at least part of the |TDM400 shared | interrupt issue. I don't have any digium T1/E1 cards laying around, | but if memory serves correctly, the T1/E1 cards do not use the same | pci controller chip. That would suggest the T1/E1 cards are |less of an | issue then with the TDM400 card. | |That's good to know, but considering the response from Digium |on the TDM400 ( try another motherboard) when there didn't |seem to even be an int. sharing issue, the card just couldn't |be seen at all , and the support I received from Sangoma on a |recent FXS issue that was resolved within a few days, I would |tend to go with Sangoma for the T1 card, if and when I have the need. | |John Novack | |___ |--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/ShoreTel REFER support
Hello All, Here's the problem, we have happily set up several Asterisk servers to offer commercial service in the UK, our wholesale SIP termination partner (Magrathea - use SER/CiscoGW to provide us the service on a public IP address) - till now we have used Asterisk to connect clients on private IP's with Asterisk doing the required conversion for SIP/IAX between public and private IP's. The current issue is that we have recently agreed to support ShoreTel PBX's with their new SIP trunk feature, and in staging the first install we have found that certain features (blind transfer) require support for both SIP Refer and Refer Replace - which are not supported by the current VoIP provider SER config. (For some good reasons as they use public IP's) So the challenge is to quickly work out the possibility of either adding a SER setup in-between the ShoreTel PBX and the VoIP provider SER unit or preferably finding a way to use one of our current asterisk servers to provide support for this need. The intent of this setup is to both allow for NAT - E.g. use private IP's for the ShoreTel system and public Ip for the VoIP provider, as well as ensuring that the local Asterisk/SER server supports the required Refer and Refer replace commands to allow the ShoreTel PBX to be able to offer blind transfer support. ShoreTel uses the below call control steps during a transfer with the current architecture: . Blind transfer: A calls B. A puts B on hold. A sends a REFER to B transferring it to C. . Consult transfer: A calls B. A puts B on hold. A calls C. A puts C on hold. A sends REFER to C transferring it to B. ShoreTel architecture uses SIP REFER method for blind transfers and SIP REFER with Replaces header to do consult transfers. This means that since (For NAT reasons) our SIP.conf has two contexts - Sip trunk and ShoreTel trunk both have reinvite=no (also to maintain billing records) the SIP Refer functions are not working as planned or hoped.Or Refer is not supported? My problems are: a) My friend Google has little to offer in exactly which RFC's Asterisk supports (particularly as recently Google does not search correctly the list archives?) - Is the SIP Refer function supported? b) Very short timetable to deliver the working solution - 1 week-Particularly if we have to plunge into adding SER to the mix - Steep learning curve with SER? - as some (most?) of IpTels web site is down? Can any one offer guidance on whether my proposed solution will work and share any tips on problems I should be aware of? If any one is interested in taking this on as an Easter project for minor commercial reward - email me off list (magnus at mcomwifi dot net) If this is the wrong list for this type of thing - Apologies Thanks Magnus ___ --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] connecting Digium E1 pri card to panasonic TD-500
Hello, Maybe someone has connected Panasonic KX-TD500 to asterisk using the KX-TD5029 ? I would like to have a IVR-like setup with panasonic: Telco-BRIx2[PABX ] Telco-PRi--[ KX ]-(KX-TD5029)--Asterisk Telco_2nd-POTSx4---[TD500] Any help or your comments are welcome.. -- Krzysztof Drewicz Affordable 2/4 span E1/T1 PCI-cards. 100% Asterisk compatible. See http://4e1.pl ___ --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
