Re: [Asterisk-Users] hint on Zap channels
[EMAIL PROTECTED] ha scritto: is it possible to use the cid of a isdn-phone as well to identify multiple devices behind one line ? I did not understand the question, what you mean? 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
RE: [Asterisk-Users] Weird IAX trunking/7960/ILBC quality issue
I know it's bad form to reply to one's own messages, but I should have added that both boxes in question are running 1.2. I was under the impression that many of the IAX jitter buffer issues had been resolved in 1.2? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth 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 mysql cdr garbage
hi, Just wanted to know if anyone else is experiencing 'garbage' mysql call detail records on asterisk v1.2? So, where the à is, tehre should be a number.. Example: 1. 2005-12-16 10:07:08 Local/[EMAIL PROTECTED] à 400Tech: à 210 ANSWERED 00:00 2. 2005-12-16 10:02:43 SIP/206-69... à 400Tech: à 210 ANSWERED 04:25 If I restart asterisk, this is solved for a few hours.. Cheers, Kristof. ___ --Bandwidth 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] Fax detected, but no fax extension
--- JP Carballo [EMAIL PROTECTED] a écrit : [EMAIL PROTECTED] wrote: OK, Is Asterisk able to switch incoming calls according to fax or voice to the right extension . Which function detect incoming signal ? If you have faxdetect enabled in zapata.conf, (the default is off), asterisk listens for a fax tone when a call comes in. I enable faxdetect in zapata.conf Now if you Answer() the call before you Dial(), it will switch to the fax extension. What do you think of NVFaxDetect() in extension.conf to listen the tones ? -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.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
[Asterisk-Users] Meetme option Ax
Dear All, I am a bit confused with the Meetme option A and x. My intention is to close the conference room when the user calls [newConf] hangup, I have the following line in extensions.conf: [newConf] exten = s,1,Answer exten = s,2,MeetMe(,eAx) [enterConf] exten = s,1,Answer exten = s,2,MeetMe(,) UserA calls newConf and is assigned to room 100, userB calls enterConf and entered room 100 when prompted. Then if userA hangup, the conference room 100 is still active if userB still on hold. What did I miss? What should I do to make this work? Thanks for any help available. Regards, TC Chan ___ --Bandwidth 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] hint on Zap channels
an isdn-line has two usable 64k channels and you can connect multiple phones to an isdn-line each phone is using it's own msn/cid for calls towards the isdn-phones you can tell asterisk to use an specified channel eg. exten-123,1,Dial(Zap/1/123) exten-124,1,Dial(Zap/2/124) this way hints for Zap channels work for incoming calls but usually you use a group/span in your dialplan so it's possible to use both channels for any extension/msn but for outgoing calls both isdn-devices use any free channel of the isdn-line [EMAIL PROTECTED] wrote on 16.12.2005 09:13:33: [EMAIL PROTECTED] ha scritto: is it possible to use the cid of a isdn-phone as well to identify multiple devices behind one line ? I did not understand the question, what you mean? 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 ___ --Bandwidth 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] Configuration of two Asterisk server
Hi I am have two Asterisk server at two different location one is having static ip 203.101.42.14 and other is having static ip 10.42.16.1 how can i integrate both so that i can use the others dial plan. Regards Mantu ___ --Bandwidth 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] Alternative source for Asterisk-IM
Thank's Takayuki Uehara for your information about asterisk-im Takayuki Uehara [EMAIL PROTECTED] Enviado Por: [EMAIL PROTECTED] 16/12/05 01:51 Favor responder a Asterisk Users Mailing List - Non-Commercial Discussion Para:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com cc: Assunto:[Asterisk-Users] Alternative source for Asterisk-IM - I tried to download the Aserisk-IM software from the URL below but the server returns 404 not found response. http://www.jivesoftware.org/wildfire/plugins/asterisk-im.jar Does anybody know any alternative source for downloading Asterisk-IM? Thanks in advance, Ooey -- Takayuki Ooey Uehara [EMAIL PROTECTED] 090-1426-4482, Skype ID: tuehara ___ --Bandwidth 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] E1 Echo (was: Small explanation of txgain rx gain statement please)
On 12/15/05, Colin Anderson [EMAIL PROTECTED] wrote: Does anyone have any experience in this area? Any ideas? How heavy handed would it be to increase the tap length to 256? I have not seen anyone suggest that this might be a good idea. On my PRI, 256 made things bad, super echo-y. Moving back to 128 works good 99% of the time, for me. I tried this last night, and have to agree that 256 does seem to be somehow broken. Thanks for the datapoint. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Raltime database schemas
On 12/16/05, Douglas Garstang [EMAIL PROTECTED] wrote: Where can I find the realtime database schemas documented? Apparently almost any static .conf file can be mapped in realtime. What about meetme.conf, rtp.conf and so on? Where is the table format of these documented? You can try the Wiki at voip-info.org to see about documentation for applications which have had realtime integration. There is currently a patch on the bug tracker to introduce RealTime into meetme. I don't believe it's currently part of the mainstream Asterisk as of yet. BJ -- 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
[Asterisk-Users] HW Echo Cancellers
Hi, To solve echo problems, I'm considering 2 alternatives. 1 Sangoma A104d - I can't find support for asterisk 1.2.1 2 Desktop echo canceller - http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html - I want to know where to buy and price. Any suggestion is appreciated. Thanks. Jason. p.s. : asterisk cli command reload can change rx_gain and tx_gain? __ 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
[Asterisk-Users] Does hardware like this exist...?
Hi all! I am looking for a device that I can stick in a USB-port on my Asterisk server and that allows me to connect one/more (cordless) PSTN-phones in such a way that they'll work with SIP/Asterisk. I know there are USB-phones, but what I'm looking for is 'the USB-phone without the phone', if you know what I mean... ;-) Regards, Evert ___ --Bandwidth 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] Romania/Rumania setup
Hi guys, is there somebody that have experience setting up an Asterisk box with Sangoma card, in Romania? I've installed Asterisk-1.2.1 with UniCall-0.0.3pre8. Wanrouter-status says connected, ztcfg says: [EMAIL PROTECTED] asterisk]# cat /proc/zaptel/1 Span 1: WPE1/0 wanpipe1 card 0 AMI/ 1 WPE1/0/1 CAS 2 WPE1/0/2 CAS 3 WPE1/0/3 CAS 4 WPE1/0/4 CAS 5 WPE1/0/5 CAS 6 WPE1/0/6 CAS 7 WPE1/0/7 CAS 8 WPE1/0/8 CAS 9 WPE1/0/9 CAS 10 WPE1/0/10 CAS 11 WPE1/0/11 CAS 12 WPE1/0/12 CAS 13 WPE1/0/13 CAS 14 WPE1/0/14 CAS 15 WPE1/0/15 CAS 16 WPE1/0/16 HDLCFCS 17 WPE1/0/17 CAS 18 WPE1/0/18 CAS 19 WPE1/0/19 CAS 20 WPE1/0/20 CAS 21 WPE1/0/21 CAS 22 WPE1/0/22 CAS 23 WPE1/0/23 CAS 24 WPE1/0/24 CAS 25 WPE1/0/25 CAS 26 WPE1/0/26 CAS 27 WPE1/0/27 CAS 28 WPE1/0/28 CAS 29 WPE1/0/29 CAS 30 WPE1/0/30 CAS 31 WPE1/0/31 CAS I've also, as suggested, to cut-off the channel 16, but nothing is changed. Dialing a number, this is what I receive: -- -- Accepting AUTHENTICATED call from XXX.XXX.XXX.XXX: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (gsm|ulaw|alaw), priority = mine -- Executing SetCallerID(IAX2/USER-3, ) in new stack -- Executing Dial(IAX2/USER-3, UniCall/g1/X) in new stack Dec 16 12:04:04 WARNING[8205]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(1) Dec 16 12:04:04 WARNING[8205]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Make call Dec 16 12:04:04 WARNING[8205]: chan_unicall.c:1077 unicall_call: Make call failed - Blocked -- Couldn't call g1/X Dec 16 12:04:04 WARNING[8205]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel gains Dec 16 12:04:04 WARNING[8205]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel switching -- Hungup 'UniCall/1-1' == Everyone is busy/congested at this time (0:0/0/0) -- Timeout on IAX2/USER-3 == CDR updated on IAX2/USER-3 -- Executing Hangup(IAX2/USER, ) in new stack == Spawn extension (trunk, t, 1) exited non-zero on 'IAX2/USER-3' -- Hungup 'IAX2/USER-3' --my unicall.conf: -- [channels] context=trunk usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=yes amaflags=billing protocolclass=mfcr2 protocolvariant=ro,20,9 protocolend=cpe loglevel=255 group = 1 channel = 1-15 ;skip time slot 16 channel = 17-31 --Any ideas, suggenstions? If you need more infos, just ask. Thanks a lot! -- .:FaberK:. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to change the Dial command H option to ## ?
Quoting Matt Riddell [EMAIL PROTECTED]: Hi Matt, I have read up on features.conf but the documentation is rather sparse. Can you show a more detailed example of the method involved? Obelix wrote: I want to use '##' to terminate a call instead of the '*' used by the Dial command's H option. Is there a way to change the key or use another option to achieve the same effect? Application map in features.conf assigning ## to Hangup() ? Maybe :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth 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] Will ooh323 ever move from addons?
Will the ooh323c module ever be moved to asterisk as a standard module or will it always remain an addon? It is in addons for licensing reasons; the underlying H.323 stack from Objective Systems is dual-licensed like Asterisk is; users who want to use chan_ooh323 in a commercial environment (like Asterisk Business Edition) must obtain a commercial license for the H.323 stack as well. This is a similar situation to the MySQL connector modules, which is the same reason they are in asterisk-addons. With our move to Subversion it might be possible to merge these back together and when we make commercial versions of Asterisk available we would just exclude them; I'll have to talk to Mark and the others about that. Thanks Rich ___ --Bandwidth 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] E1 Echo
I am beginning to wonder whether what echo IS heard is being caused by packetisation delays in the network - The default tap length is 128, or I believe 16ms. If something in the PSTN causes a delay more than that length (no idea what might cause that) then echo would still be heard. We have found that a relatively innocent change by the local incumbent operator has forced us to modify our pstn gateways to change from 128 taps to 256 taps. What type of a change did they make? ___ --Bandwidth 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] ztdummy / timer problem with kernel 2.6.14
I got a problem with musiconhold / meetme which I cannot get to work correctly. I just install a fresh installation off Fedora core 4 and did a yum update on the system before I downloaded asterisk 1.2.1. First I compiled zaptel with make clean, make linux26, make install, make config, and then I configured libpri with make clean, make, make install. I restarted the system and sees that zaptel is bringing up ztdummy. Doing a lsmod I can see this: ztdummy 7816 0 zaptel193540 12 ztdummy crc_ccitt 6209 1 zaptel So the modules have been added into the system correctly. I then compiled and install asterisk with, make clean, make, make mpg123, make install, make samples. And then I added test user in sip.conf, added a new exten = 605,1,MusicOnHold I edited the zaptel.conf to remove the ; in front of musiconhold = default, then I started asterisk with safe_asterisk, logged in with my sip client. Now when I dial exten 605, I see this in asterisk console: -- Executing Answer(SIP/590-6f28, ) in new stack -- Executing MusicOnHold(SIP/590-6f28, ) in new stack -- Started music on hold, class 'default', on channel 'SIP/590-6f28' I can also see that mpg123 has started, ps aux |grep -i mpg123 Now the problem is that the music is playing for 1 sec, stops up for a couple sec then starts again, stops up and starting etc, doing a moh reload it play for about 2 sec and stops, starts, stops etc. I cannot understand what I have done wrong during the installation. When I do a zttest I get this results: Results after 49 passes --- Best: 99.963379 -- Worst: 99.926758 -- Average: 99.948681 I have also tried on the same server to use Slackware 10.2, reinstall, and created a kernel 2.6.14, remove the usb drivers to make sure that zaptel not using that, did many different things with this but still the same problem (also results with zttest). I have also tried with madplay, shoutcast streams, but still the same problem, and to top this off I also had the same problem with many different distro on different virtual hosts on a Vmware enterprise server (I know that vmware is the problem with vmware tools and rtc), but anyway this very strange/irritating for me that this doesn't work. Do I have need to by a digium card to get the timer problem work correctly with asterisk? Is there something that I have missed during the installation or is it my hardware that is not good enough? The hardware is an IBM 306 with 3.6 xeon, 1GB ram and raid 1, 2x 80 GB sata disk. Is there anyone that can help me, has the same problem and a solution for me to try out!! Regards, Fredrik Jensen ___ --Bandwidth 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] cdr mysql problem
Dear folks, I've just compiled asterisk-addon1.2.1 after installing MySQL and MySQl-devel packages. and adjust my cdr_mysql.conf to use the defined database using username and password. But as soon as starting asterisk i get error messages informing me of error, error message is as follows : cdr_addon_mysql.c : Failed to connect mysql database cdr on localhost and res_config_mysql.c : Failed to connect database server on . Im realy lost and dont know whats wrong. I've checked the connection to MySql in command line using the same user and host and its been connected without any problem. Anyone has any idea whats wrong here. Regards. --- M. Shokuie Nia. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/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] Does hardware like this exist...?
On 12/16/05, Evert Meulie [EMAIL PROTECTED] wrote: Hi all! I am looking for a device that I can stick in a USB-port on my Asterisk server and that allows me to connect one/more (cordless) PSTN-phones in such a way that they'll work with SIP/Asterisk. I know there are USB-phones, but what I'm looking for is 'the USB-phone without the phone', if you know what I mean... ;-) You're looking for a USB FXS port. Yes, they do exist. You can take a look at the Astribank-8 from Xorcom (www.xorcom.com). I really don't know how well they work as I haven't any personal experience with their equipment, but they were exhibiting this solution at the last Astricon a few months back. -- 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] cdr mysql problem
Mohammad Shokuie wrote: error messages informing me of error, error message is as follows : cdr_addon_mysql.c : Failed to connect mysql database cdr on localhost and res_config_mysql.c : Failed to connect database server on . Im realy lost and dont know whats wrong. I've checked the connection to MySql in command line using the same user and host and its been connected without any problem. I had this problem after upgrading mysql. I had to move back to version *4.0.20. If you do a nmap on the mysql system, you'll probably not see it listening on port 3306. 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] Connecting Meridian M8x24-DS to Asterisk - No DTMFtones
Sorry, this is slightly off topic, but I wonder if somebody has some hints on getting our Meridian system to output DTMF tones to our Asterisk box. Simply put, when buttons as pressed, nothing happens. The Asterisk box has a 4 port Digium FXO card. This is what we've got: Meridian M8X24-DS Meridian M12X0 Thanks for any tips! Phil Hey, thanks for the response. I have another system (Toshiba) which works fine with it, so I'm doubting that it's a issue specifically with the Asterisk server or it being able to properly decode the tones. Have you put a butt set on the line and listen to see if there is DTMF and it is just not being recognized? I've done similar tests where instead of the asterisk server, I used a phone instead. When buttons are hit, no tones are emitted from the Meridian, but a little acknoledge beep on the Meridian phone is heard as buttons are pressed. It seems to me like a programming or some other issue on the Meridian. But, I don't know where to start to correct it. The tech is completely MIA after my call to set up an appointment. I might have to call another service group. I'm fairly competent at managing and building out Toshiba DK systems (I maintain two DK40's), but this Meridian was sort of inherited from another source. Pure guess and I'm not a Meridian experienced person either... The problem sounds like the Meridian is configured to not transmit dtmf after answer supervision. When the Meridian connects to the asterisk fxo port, the call is considered answered regardless of whether anyone on the asterisk box actually answered the call. I'd be looking for a config option on the Meridian associated with that. (Personal theory: The Meridian probably supports electronic phone sets where dtmf tones are not actually transmitted from the phone set to the pbx. Rather, a digital signal is sent to the pbx, and the pbx generates the dtmf tone when instructed to do so. I'd have to guess that might be considered a feature to keep Meridian users from accidently hitting a button on the phone set and blasting dtmf tones to the user of the analog extension (eg, asterisk). So, it makes sense they would have an option to turn dtmf on/off on a per analog line basis. Also, the Meridian is expected to react to electronic phone set buttons for special features without generating dtmf tones, which further suggests there might be an option associated with the analog line interface of the Meridian. That might even be translated (in Meridian terms) to that analog interface being defined as an extension verses an ananlog trunk. That's all just a theory with no factual experience on my part.) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr mysql problem
I've just compiled asterisk-addon1.2.1 after installing MySQL and MySQl-devel packages. and adjust my cdr_mysql.conf to use the defined database using username and password. But as soon as starting asterisk i get error messages informing me of error, error message is as follows : cdr_addon_mysql.c : Failed to connect mysql database cdr on localhost and res_config_mysql.c : Failed to connect database server on . Im realy lost and dont know whats wrong. I've checked the connection to MySql in command line using the same user and host and its been connected without any problem. Anyone has any idea whats wrong here. Regards. --- At least the second is just the usual error message when you don't use MySQL for realtime configuartion. Shouldn't affect your cdr_mysql. Is the name of the _database_ really cdr? By default it is asteriskcdrdb. Chris ___ --Bandwidth 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: AoC (Advice of Charge)
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Does Asterisk support Advice of Charge? I was told that my Telco sends me billing signalization that way, and I wonder can I use it? I have found out that this is part of EURO ISDN standard. q.956 - Advice Of Charge. Does anybody know how to implement this with Asterisk? I would like to store those informations (that I recive from my telco by q.956 standard) in MySQL, csv or any other format. -- Tomislav Parcina [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] getting started
Wanted some advice for the docs that you'd recommend someone new to Asterisk to read. I have a good knowledge of Unix and networking, so that part shouldn't be a problem. Try... http://www.asteriskdocs.org/modules/news/ The authors of Asterisk: The Future of Telephony are pleased to announce their book in PDF form, available immediately, for free. The book can be downloaded from www.asteriskdocs.org. Thanks to O'Reilly Media for supporting us and allowing us to publish the book under the Creative Commons license. Or, purchase the book from O'Reilly. I'd recommend it as an excellent starting point. ___ --Bandwidth 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] Central Registration mechanism
Hello, I would like to know if there is any mechanism whereby one can have several Asterisk servers catering to different physical locations but only one central Asterisk server responsible for client registration. From what I have read on the mailing lists, the one way to do this is by using Realtime asterisk with a MySQL database. The other is to use SER as a frontend. Is there a simpler method? - Noel. ___ --Bandwidth 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] E1 Echo
Rich Adamson wrote: We have found that a relatively innocent change by the local incumbent operator has forced us to modify our pstn gateways to change from 128 taps to 256 taps. What type of a change did they make? Although it's a bit unclear how things evolved exactly (since no-one ever tells us), a number of interconnection points throughout the country were consolidated, significantly increasing the chance that delay exceeded 128 taps. ___ --Bandwidth 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] cdr mysql problem
i am using asterisk 1.2.1 with mysql 5 without any issues, please check your configuration again, make sure you have hostname=localhost too and the dbname, user, password are correct [global] hostname=localhost dbname=databasename user=user password=password port=3306 sock=/var/lib/mysql/mysql.sock Diyanat From: Mohammad Shokuie [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cdr mysql problem Return-Path: [EMAIL PROTECTED] Dear folks, I've just compiled asterisk-addon1.2.1 after installing MySQL and MySQl-devel packages. and adjust my cdr_mysql.conf to use the defined database using username and password. But as soon as starting asterisk i get error messages informing me of error, error message is as follows : cdr_addon_mysql.c : Failed to connect mysql database cdr on localhost and res_config_mysql.c : Failed to connect database server on . Im realy lost and dont know whats wrong. I've checked the connection to MySql in command line using the same user and host and its been connected without any problem. Anyone has any idea whats wrong here. Regards. --- M. Shokuie Nia. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/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 ___ --Bandwidth 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] cdr mysql problem
error messages informing me of error, error message is as follows : cdr_addon_mysql.c : Failed to connect mysql database cdr on localhost and res_config_mysql.c : Failed to connect database server on . Im realy lost and dont know whats wrong. I've checked the connection to MySql in command line using the same user and host and its been connected without any problem. I had this problem after upgrading mysql. I had to move back to version *4.0.20. If you do a nmap on the mysql system, you'll probably not see it listening on port 3306. Or, simply do 'netstat -an | more'. Should see something like: tcp0 0 0.0.0.0:3306 If that is not seen, then mysql is not listening on a tcp port. ___ --Bandwidth 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: Meetme option Ax
In article [EMAIL PROTECTED], tcchan [EMAIL PROTECTED] wrote: Dear All, I am a bit confused with the Meetme option A and x. My intention is to close the conference room when the user calls [newConf] hangup, I have the following line in extensions.conf: [newConf] exten = s,1,Answer exten = s,2,MeetMe(,eAx) [enterConf] exten = s,1,Answer exten = s,2,MeetMe(,) UserA calls newConf and is assigned to room 100, userB calls enterConf and entered room 100 when prompted. Then if userA hangup, the conference room 100 is still active if userB still on hold. What did I miss? What should I do to make this work? I think you need to specify option x also on the second MeetMe (but not option A). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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] incoming dtmf handling by ATA devices ?
sorry if the answer is well known but i couldn't find a relevant pointer. I am trying to figure out if/how it is possible to connect a dtmf-controlled device (e.g. answering machine) to an ATA, and how to configure asterisk to achieve this. A bit of expermients with a HandyTone 286 shows that my ATA only produces audible tones on the phone when using inband dtmf and ulaw codec. Other options (rfc2833, info) do not produce any audible sound, though the SIP or RTP message do get delivered. Am i missing something ? cheers luigi ___ --Bandwidth 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] Experience sharing on Planet VIP-450 + Asterisk
Hi all, Previously I have asked about stopping iLBC in Asterisk, and I would like to use G.711 u-law only. Actually I have tried entirely remove anything file related to "ilbc" in /usr/lib/asterisk/modules, but it still didn't work. The error message about the improper RTP packet length still there, and I still can't make DTMF detection work. What's next? Well... thanks to the buggy firmware and imcompatable standard with Asterisk... First of all, I can't deny that Planet VIP-450 does a good job in packetizing voice stream, the voice quality is really good and delay is really small. Also the hardware itself is quite robust, it seldom halt.. (the machine has been up for a few days). Also it is quite feature-rich, I can say. BUT I think there is quite a number of BUGS in the firmware! In order to see which kind of DTMF Relay it is using, I have done a packet analysing. When I try to pass SIP INFO type DTMF band to VIP-450, it replies "501 Unimplemented". Also when I try to pass DTMF from my POTS phone via the FXO port, only RTP payload can be seen in the packet captures. I DID suspect that it is RFC2833, because as far as I know RFC2833 did have the DTMF textx inside the RTP packet somewhere (seems header). But asterisk just simply did not regconize them (of coz I have set DTMFmode=rfc2833)! It is pretty strange that the user manual states "VIP handles DTMF Relay per SIP specification". So VIP-450 actually is using what kind of SIP specification? How about using its Inband DTMF relay? This will certainly generate strange warning just like my case : improper ilbc frame size and tell me to use u-law to do DTMF even if I AM using G.711 u-law. It is seems that the DTMF tone generated by VIP-450 generate is kinda strange... So the final solution is, SIMPLY SWITCH OFF THE DTMF RELAY IN VIP-450. Please try to type "show coding" in console mode and you will see a lot of coding (codec) profiles. Most of them are with DTMF relay. Just switch off them by "set coding profile id dtmf_relay off" (please check with the manual). If you want to stop certain codec, just simply make that coding profile unusable in voice. For example, "set coding profile id voice off". If youonly turn on the profile withu-law, the SIP header it issues will just consist of 0x4 (ulaw) codec, not 0x105. In mypoint of view, Planet isexpectingthis deviceisconnected to another VIP-450, not really for Asterisk or anything else, even not fora soft phone. Certainly this is not enough for everyone, at leastI can't do any IVR and something what a PBX should have (just like what I can do in Asterisk). I hope my experience will help anyone who is using VIP-450 with Asterisk, just like me. I have done Googling for 3 days but I can search for nothing related to this issue. Sorry for my poor written English. Cheers, Jason Chan, Hong Kong No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.1/204 - Release Date: 15/12/2005 ___ --Bandwidth 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] E1 Echo
We have found that a relatively innocent change by the local incumbent operator has forced us to modify our pstn gateways to change from 128 taps to 256 taps. What type of a change did they make? Although it's a bit unclear how things evolved exactly (since no-one ever tells us), a number of interconnection points throughout the country were consolidated, significantly increasing the chance that delay exceeded 128 taps. Strange... I would never had expected consolidation to have that kind of impact. It almost sounds like they have something in the E1 data stream that buffers (and delays) content, maybe decoding and re-encoding in some fashion. ___ --Bandwidth 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] E1 Echo
On Friday 16 December 2005 08:12, Florian Overkamp wrote: Although it's a bit unclear how things evolved exactly (since no-one ever tells us), a number of interconnection points throughout the country were consolidated, significantly increasing the chance that delay exceeded 128 taps. I need to do some investigation of bringing the tap count WELL above that... I'd like to see what kind of performance we can get with 128 MILLISECOND tail... 128 taps is only 16ms... and 16ms of echo cancel is damn near useless, as it's fast enough that you'd likely not even hear the echo as anything more than a sidetone anyway. I imagine it's deathly hard on the CPU though. :-) -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] E1 Echo
Andrew Kohlsmith wrote: On Friday 16 December 2005 08:12, Florian Overkamp wrote: Although it's a bit unclear how things evolved exactly (since no-one ever tells us), a number of interconnection points throughout the country were consolidated, significantly increasing the chance that delay exceeded 128 taps. I need to do some investigation of bringing the tap count WELL above that... I'd like to see what kind of performance we can get with 128 MILLISECOND tail... 128 taps is only 16ms... and 16ms of echo cancel is damn near useless, as it's fast enough that you'd likely not even hear the echo as anything more than a sidetone anyway. I imagine it's deathly hard on the CPU though. :-) Actually, the problem is different. If you receive an echo on the PSTN gateway that has a 16ms echo, the problem would not be noticeable there, but if you then add a VoIP connection the delay added would make the echo audible. Florian ___ --Bandwidth 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] E1 Echo
Rich Adamson wrote: Strange... I would never had expected consolidation to have that kind of impact. It almost sounds like they have something in the E1 data stream that buffers (and delays) content, maybe decoding and re-encoding in some fashion. Well, the problem is the difference between keeping under 16ms and sliding _just_ over limit to 18ms would make the effect audible almost immediately. We used the sangoma echospike tools to measure the delay and adjusted our taps accordingly. Florian ___ --Bandwidth 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: Does hardware like this exist...?
That looks like an interesting way to add some FXS ports to a cramped 1U server. Would using USB on the Asterisk server cause an interrupt or CPU usage issue? Especially if it already has two T1 cards? I know that I am currently looking into a new server just due to having run out of slots. Due to an issue passing faxes to a legacy PBX, I had to move the fax machines to new Telco lines and set the DID numbers for them in asterisk to dial back out to the new phone numbers that the faxes were moved to. Just because I couldn't put an FXO/FXS card in the server. I guess the USB device would only be a solution if it handles the faxes like ZAP and doesn't pocketsize them. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- BJ Weschke [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On 12/16/05, Evert Meulie [EMAIL PROTECTED] wrote: Hi all! I am looking for a device that I can stick in a USB-port on my Asterisk server and that allows me to connect one/more (cordless) PSTN-phones in such a way that they'll work with SIP/Asterisk. I know there are USB-phones, but what I'm looking for is 'the USB-phone without the phone', if you know what I mean... ;-) You're looking for a USB FXS port. Yes, they do exist. You can take a look at the Astribank-8 from Xorcom (www.xorcom.com). I really don't know how well they work as I haven't any personal experience with their equipment, but they were exhibiting this solution at the last Astricon a few months back. -- 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 ___ --Bandwidth 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] incoming dtmf handling by ATA devices ?
sorry if the answer is well known but i couldn't find a relevant pointer. I am trying to figure out if/how it is possible to connect a dtmf-controlled device (e.g. answering machine) to an ATA, and how to configure asterisk to achieve this. A bit of expermients with a HandyTone 286 shows that my ATA only produces audible tones on the phone when using inband dtmf and ulaw codec. Other options (rfc2833, info) do not produce any audible sound, though the SIP or RTP message do get delivered. Am i missing something ? Not sure this will help much, but just tested the following: C7960 - asterisk(a) - iax2/gsm - asterisk(b) - spa3k The sip definition for asterisk(b) to the spa3k is rfc2833 and g711u. When a call is completed between the C7960 and the spa3k, pressing any key on the C7960 results in dtmf being heard on the analog phone attached to the spa3k. An ethereal inspection of the sip packets flowing into the spa3k does not indicate the presence of rfc2833-formated packets. Therefore it would appear that either asterisk(a) or asterisk(b) is actually generating the dtmf tones inband. The dtmf tones are always approx 100 ms in duration. You might take a look at an ethereal trace of the sip packets delivered to the ata to see what might be happening. ___ --Bandwidth 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: Connecting Meridian M8x24-DS to Asterisk - NoDTMFtones
I agree. I am sure it is a programming issue with DTMF on Stations vs. Trunks. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Rich Adamson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Sorry, this is slightly off topic, but I wonder if somebody has some hints on getting our Meridian system to output DTMF tones to our Asterisk box. Simply put, when buttons as pressed, nothing happens. The Asterisk box has a 4 port Digium FXO card. This is what we've got: Meridian M8X24-DS Meridian M12X0 Thanks for any tips! Phil Hey, thanks for the response. I have another system (Toshiba) which works fine with it, so I'm doubting that it's a issue specifically with the Asterisk server or it being able to properly decode the tones. Have you put a butt set on the line and listen to see if there is DTMF and it is just not being recognized? I've done similar tests where instead of the asterisk server, I used a phone instead. When buttons are hit, no tones are emitted from the Meridian, but a little acknoledge beep on the Meridian phone is heard as buttons are pressed. It seems to me like a programming or some other issue on the Meridian. But, I don't know where to start to correct it. The tech is completely MIA after my call to set up an appointment. I might have to call another service group. I'm fairly competent at managing and building out Toshiba DK systems (I maintain two DK40's), but this Meridian was sort of inherited from another source. Pure guess and I'm not a Meridian experienced person either... The problem sounds like the Meridian is configured to not transmit dtmf after answer supervision. When the Meridian connects to the asterisk fxo port, the call is considered answered regardless of whether anyone on the asterisk box actually answered the call. I'd be looking for a config option on the Meridian associated with that. (Personal theory: The Meridian probably supports electronic phone sets where dtmf tones are not actually transmitted from the phone set to the pbx. Rather, a digital signal is sent to the pbx, and the pbx generates the dtmf tone when instructed to do so. I'd have to guess that might be considered a feature to keep Meridian users from accidently hitting a button on the phone set and blasting dtmf tones to the user of the analog extension (eg, asterisk). So, it makes sense they would have an option to turn dtmf on/off on a per analog line basis. Also, the Meridian is expected to react to electronic phone set buttons for special features without generating dtmf tones, which further suggests there might be an option associated with the analog line interface of the Meridian. That might even be translated (in Meridian terms) to that analog interface being defined as an extension verses an ananlog trunk. That's all just a theory with no factual experience on my part.) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Experience sharing on Planet VIP-450 + Asterisk
Hi all, Previously I have asked about stopping iLBC in Asterisk, and I would like to use G.711 u-law only. Actually I have tried entirely remove anything file related to ilbc in /usr/lib/asterisk/modules, but it still didn't work. The error message about the improper RTP packet length still there, and I still can't make DTMF detection work. What's next? Well... thanks to the buggy firmware and imcompatable standard with Asterisk... First of all, I can't deny that Planet VIP-450 does a good job in packetizing voice stream, the voice quality is really good and delay is really small. Also the hardware itself is quite robust, it seldom halt.. (the machine has been up for a few days). Also it is quite feature-rich, I can say. BUT I think there is quite a number of BUGS in the firmware! In order to see which kind of DTMF Relay it is using, I have done a packet analysing. When I try to pass SIP INFO type DTMF band to VIP-450, it replies 501 Unimplemented. Also when I try to pass DTMF from my POTS phone via the FXO port, only RTP payload can be seen in the packet captures. I DID suspect that it is RFC2833, because as far as I know RFC2833 did have the DTMF textx inside the RTP packet somewhere (seems header). But asterisk just simply did not regconize them (of coz I have set DTMFmode=rfc2833)! It is pretty strange that the user manual states VIP handles DTMF Relay per SIP specification. So VIP-450 actually is using what kind of SIP specification? How about using its Inband DTMF relay? This will certainly generate strange warning just like my case : improper ilbc frame size and tell me to use u-law to do DTMF even if I AM using G.711 u-law. It is seems that the DTMF tone generated by VIP-450 generate is kinda strange... So the final solution is, SIMPLY SWITCH OFF THE DTMF RELAY IN VIP-450. Please try to type show coding in console mode and you will see a lot of coding (codec) profiles. Most of them are with DTMF relay. Just switch off them by set coding profile id dtmf_relay off (please check with the manual). If you want to stop certain codec, just simply make that coding profile unusable in voice. For example, set coding profile id voice off. If you only turn on the profile with u-law, the SIP header it issues will just consist of 0x4 (ulaw) codec, not 0x105. In my point of view, Planet is expecting this device is connected to another VIP-450, not really for Asterisk or anything else, even not for a soft phone. Certainly this is not enough for everyone, at least I can't do any IVR and something what a PBX should have (just like what I can do in Asterisk). I hope my experience will help anyone who is using VIP-450 with Asterisk, just like me. I have done Googling for 3 days but I can search for nothing related to this issue. Sorry for my poor written English. Cheers, Jason Chan, Hong Kong You should post this stuff and future findings on the wiki. Thanks, 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
RE: [Asterisk-Users] Connecting Meridian M8x24-DS to Asterisk - NoDTMFtones
Sorry, this is slightly off topic, but I wonder if somebody has some hints on getting our Meridian system to output DTMF tones to our Asterisk box. Simply put, when buttons as pressed, nothing happens. The Asterisk box has a 4 port Digium FXO card. This is what we've got: Meridian M8X24-DS Meridian M12X0 Thanks for any tips! Phil Hey, thanks for the response. I have another system (Toshiba) which works fine with it, so I'm doubting that it's a issue specifically with the Asterisk server or it being able to properly decode the tones. Have you put a butt set on the line and listen to see if there is DTMF and it is just not being recognized? I've done similar tests where instead of the asterisk server, I used a phone instead. When buttons are hit, no tones are emitted from the Meridian, but a little acknoledge beep on the Meridian phone is heard as buttons are pressed. It seems to me like a programming or some other issue on the Meridian. But, I don't know where to start to correct it. The tech is completely MIA after my call to set up an appointment. I might have to call another service group. I'm fairly competent at managing and building out Toshiba DK systems (I maintain two DK40's), but this Meridian was sort of inherited from another source. Pure guess and I'm not a Meridian experienced person either... The problem sounds like the Meridian is configured to not transmit dtmf after answer supervision. When the Meridian connects to the asterisk fxo port, the call is considered answered regardless of whether anyone on the asterisk box actually answered the call. I'd be looking for a config option on the Meridian associated with that. (Personal theory: The Meridian probably supports electronic phone sets where dtmf tones are not actually transmitted from the phone set to the pbx. Rather, a digital signal is sent to the pbx, and the pbx generates the dtmf tone when instructed to do so. I'd have to guess that might be considered a feature to keep Meridian users from accidently hitting a button on the phone set and blasting dtmf tones to the user of the analog extension (eg, asterisk). So, it makes sense they would have an option to turn dtmf on/off on a per analog line basis. Also, the Meridian is expected to react to electronic phone set buttons for special features without generating dtmf tones, which further suggests there might be an option associated with the analog line interface of the Meridian. That might even be translated (in Meridian terms) to that analog interface being defined as an extension verses an ananlog trunk. That's all just a theory with no factual experience on my part.) On one system I worked on you had an option to program a memory block to use actual DTMF. It was optional. The customer had to buy a display phone and a tech's manual to get it to work since programming was done through the phone and all they had were low end display-less phones. Don't remember what kind of system it was but once that memory block was set and the system rebooted, it worked like a champ. Thanks, 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] ISDN/CAPI outgoing calls - weirdness with ringing
All,
I have the following set up:
Fedora Core 4 box (yum updated to current)
Asterisk 1.2.1 + Chan_Capi-cm-0.6.1
AVM C4 card
2 x ISDN2e lines bonded with switchboard number, fax number and 10 x DDI
numbers from British Telecom
14 x Cisco 7960 phones with SIP 7.5
The ISDN lines work in P2P mode and calls are presented with the last 4
digits only - I land them in a context and branch out from there -
everything to do with incoming calls works just fine!
I have a problem with outgoing calls that are routed over the BT network and
the way in which 'ringing' is presented... depending on the called party
number (hence phone provider) I get different results. For example:
a) if I dial another BT number I get a fraction of a second's ring followed
by silence until the called party answers. The Cisco phone displays:
Proceeding (in 100)
very briefly and is almost immediately over-written by:
Session Progress (in 183)
until the called party answers - at no point is Ringing Destination (in 180)
displayed
b) if I dial an Orange or O2 mobile number I get a second or two's worrth of
silence [while the Orange network locates the mobile] then the mobile rings
in the normal way and the Cisco phone plays out US style ringing. When the
number is dialled the phone displays:
Proceeding (in 100)
when the mobile starts to ring the Cisco phone displays:
Ringng Destination (in 180)
c) if I dial a Bulldog phone number then I get three messages:
Proceeding (in 100) - for a second or so
Session Progress (in 183) - for a couple of seconds
Ringng Destination (in 180) - while the called party's phone rings
d) and the really weird one - if I dial *some* international numbers I get
both UK (BT) ringing tone overlaid with Asterisk/VoIP (US) ringing tone
I have two ways of dialling out:
1. with an explicit 9 for an outside line -- get dialtone from BT and then
dial rest of the digits - like a legacy PBX
2. dialing just based on the fact that the extension starts with a zero so
its an outside call via BT
I have tried all combinations of early B3 connect 'always', 'on success' and
'never' and it doesn't appear to change things... the relevant part of
extensions.conf is below for completness.
Before I dive in to the next level down:
- is this a known issue?
- is there a solutiuon/workaround/patch/fix
- do I need to get down and dirty with CAPI and SIP debug?
Mike
;
; external-routes: this is where we get to dial out
;
[external-routes]
;
; outgoing via main ISDN line using explicit 9 for an outside line
; and ISDN eqarly B3 connect (overlap sending) to drop us to the
; BT provided dialtone and work like a normal/legacy phone system -
; we force the caller ID to our exchange number so that DDI's dont
; leak out
;
exten = 9,1,NoOp(ISDN: Pickup outside line (early B3 connect) for:
${CALLERIDNUM})
exten = 9,2,SetCallerId(${THORCOM_MAIN})
exten = 9,3,Dial(CAPI/g1//b)
exten = 9,4,Hangup
;
; implicit trunked call - here we could/should do an ENUM look
; up to see if we can place the call via IP and fall back to BT
; if not... just for now this isn't implemented and we always call
; out via BT!!
;
exten = _0.,1,Dial(CAPI/g1/${EXTEN}/b); early B3
connect always
;exten = _0.,1,Dial(CAPI/g1/${EXTEN}/B) ; early B3
connect on success
;exten = _0.,1,Dial(CAPI/g1/${EXTEN}) ; no special
options
exten = _0.,2,Hangup
___
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Re: [Asterisk-Users] E1 Echo
On Friday 16 December 2005 09:02, Florian Overkamp wrote: Well, the problem is the difference between keeping under 16ms and sliding _just_ over limit to 18ms would make the effect audible almost immediately. We used the sangoma echospike tools to measure the delay and adjusted our taps accordingly. Sangoma echospike tools? Please elaborate! -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] A rather big setup.
Tom Rymes wrote: Also, if these tenants are not related, then why not run more than one Asterisk server and avoid interconnecting them? Sure, you'll have multiple systems to maintain, but they will be smaller, less complex systems. Also, since each company is unrelated, there is little benefit to having them all on the same server (no need to dial between offices, etc) I agree. You have an exact multi-tenant setup. I would take each of the E1 connections and plug them into a server, one for each floor. If one of the servers runs out of connectins on its E1, then it can use the outbound lines from an * for another floor. You do not want the entire building to have an outage. You happen to have 8 E1 and 8 floors, right? If not, then you can have any floors that don't have a direct E1 connection dial out through the * servers that do have an E1. Also, *do not* run fax over SIP if at all possible. You will need channel banks to handle the faxes. Setup a hylafax server so each tenant can receive faxes over email, and send them from their computers, or just strike up a deal with efax for the building. 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
Re: [Asterisk-Users] Re: AoC (Advice of Charge)
hello tomislav, Tomislav Parcina wrote: Does Asterisk support Advice of Charge? I was told that my Telco sends me billing signalization that way, and I wonder can I use it? I have found out that this is part of EURO ISDN standard. q.956 - Advice Of Charge. Does anybody know how to implement this with Asterisk? I would like to store those informations (that I recive from my telco by q.956 standard) in MySQL, csv or any other format. i have _partially_ implemented AOC into the libpri and chan_zap part of asterisk (the IEs for AOC units are decoded and encoded and you will see the AOC info on the console if you have increased verbosity to 5). unfortunately it was beyond my scope to propagate this information to the bridged channel, as the info from the telco provider is transmitted during the call termination phase and asterisk destroys the bridge to early (right after it receives the first notice that the call has to be terminated) and so there is no possibility to pass the AOC to the bridged channel nor to write anything to the CDR. what has to be done is to rewrite the call termination process so it does not terminate each of it's bridged legs seperately in a state machine but together. regards frank ___ --Bandwidth 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: ztdummy / timer problem with kernel 2.6.14
In article [EMAIL PROTECTED] srv02.netpower.lan, [EMAIL PROTECTED] says... I just install a fresh installation off Fedora core 4 and did a yum update on the system before I downloaded asterisk 1.2.1. First I compiled zaptel with make clean, make linux26, You don't need make linux26 anymore, you can use make. strange/irritating for me that this doesn't work. Do I have need to by a digium card to get the timer problem work correctly with asterisk? I don't know how to solve your problem, but with Kernel 2.6 (it has timing mehanism) you don't need Digium hardver to make MOH work. Hope this helps. -- Tomislav Parcina [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] HW Echo Cancellers
I have used the orion...you can buy right from them. However, I was not impressed with their sales teamI have one on a beta test, and they threatened to call a collection agency in when I refused paybent before the beta expired. I had some weird DTMF issues with the Orion, otherwise ok. -D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Friday, December 16, 2005 5:52 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] HW Echo Cancellers Hi, To solve echo problems, I'm considering 2 alternatives. 1 Sangoma A104d - I can't find support for asterisk 1.2.1 2 Desktop echo canceller - http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html - I want to know where to buy and price. Any suggestion is appreciated. Thanks. Jason. p.s. : asterisk cli command reload can change rx_gain and tx_gain? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 Echo
Well, the problem is the difference between keeping under 16ms and sliding _just_ over limit to 18ms would make the effect audible almost immediately. We used the sangoma echospike tools to measure the delay and adjusted our taps accordingly. Sangoma echospike tools? Please elaborate! See sangoma's -users posting from Dec 13th, which I quote: I just wanted to let you know that we do provide a tool to debug echo. We send a unit impulse and record the Finite Impulse Response (FIR) so it can be plotted and analyzed. The code that does this is the release at ftp.sangoma.com/linux/custom/2.3.4. Instructions on using it are in the wiki in http://sangoma.editme.com/wanpipe-linux-asterisk-debugging. Although the code is wanpipe, all the interaction is at the zaptel level, so I am pretty sure it will work on Digium or other cards as well. Just being able to see what the echo looks like on a troublesome line gives quite a lot of info. You can see if the echo is delayed, or markedly non-linear. I haven't tried it as yet but plan to do so. Rich ___ --Bandwidth 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] Mediatrix 1204 help please.
OK we need some help in setting up a good wiki-info page for setting up the Mediatrix 1204 to work with asterisk. If anyone has set these unit's up and have them working please post your settings here so we can create a page on the wiki. These unit's are being sold to be used via sip format with asterisk and there is no real information on getting them working. At present there one of the worst I have run into to get correctly working. These are very expensive and some of us can't afford to send them back for a restocking fee. If someone working with Mediatrix has a white paper on getting these unit's working please let us know the link for it. It would be very helpful for many 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] Mediatrix 1204 help please.
Ariel, There are some notes in the list archives about getting them going. -Nate -Original Message- From: Ariel Batista [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 9:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Mediatrix 1204 help please. OK we need some help in setting up a good wiki-info page for setting up the Mediatrix 1204 to work with asterisk. If anyone has set these unit's up and have them working please post your settings here so we can create a page on the wiki. These unit's are being sold to be used via sip format with asterisk and there is no real information on getting them working. At present there one of the worst I have run into to get correctly working. These are very expensive and some of us can't afford to send them back for a restocking fee. If someone working with Mediatrix has a white paper on getting these unit's working please let us know the link for it. It would be very helpful for many 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] CallerID/Extension Matching with Realtime Extensions
Douglas Garstang wrote: which matches when a user with callerid 5551212 dials 8000. This doesn't work with realtime extensions. or does it? Does someone know how it's done? The following doesn't work. Asterisk can't find the number. The docs for Realtime extensions clearly state that CallerID matching is not supported. ___ --Bandwidth 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] hint on Zap channels
[EMAIL PROTECTED] wrote: an isdn-line has two usable 64k channels and you can connect multiple phones to an isdn-line each phone is using it's own msn/cid Since Asterisk is not aware of these being individual devices, there is no way that hints could reliably work for them. ___ --Bandwidth 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] E1 Echo
Hi Rich, Rich Adamson wrote: Sangoma echospike tools? Please elaborate! See sangoma's -users posting from Dec 13th, which I quote: I just wanted to let you know that we do provide a tool to debug echo. We send a unit impulse and record the Finite Impulse Response (FIR) so it can be plotted and analyzed. The code that does this is the release at ftp.sangoma.com/linux/custom/2.3.4. Instructions on using it are in the wiki in http://sangoma.editme.com/wanpipe-linux-asterisk-debugging. Although the code is wanpipe, all the interaction is at the zaptel level, so I am pretty sure it will work on Digium or other cards as well. Just being able to see what the echo looks like on a troublesome line gives quite a lot of info. You can see if the echo is delayed, or markedly non-linear. I haven't tried it as yet but plan to do so. Correct, this is what we used. Florian ___ --Bandwidth 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] hint on Zap channels
Kevin, I'm not sure this would work here, but maybe it would. There was a bug posted about being able to use hint against local channels, would that not help him? http://bugs.digium.com/view.php?id=5779nbn=4 After looking at it again, I realize it might only work for parked channels, so I'm not sure. On 12/16/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: an isdn-line has two usable 64k channels and you can connect multiple phones to an isdn-line each phone is using it's own msn/cid Since Asterisk is not aware of these being individual devices, there is no way that hints could reliably work for them. ___ --Bandwidth 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] hint on Zap channels
[EMAIL PROTECTED] wrote on 16.12.2005 16:18:49: [EMAIL PROTECTED] wrote: an isdn-line has two usable 64k channels and you can connect multiple phones to an isdn-line each phone is using it's own msn/cid Since Asterisk is not aware of these being individual devices, there is no way that hints could reliably work for them. thanks for the answer - I expected this, although I hoped something different ___ --Bandwidth 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: Does hardware like this exist...?
That unit looks VERY promising! Thanks! :-) Would anyone happen to know an approx. price for a unit like this? Regards, Evert BJ Weschke wrote: On 12/16/05, Evert Meulie [EMAIL PROTECTED] wrote: Hi all! I am looking for a device that I can stick in a USB-port on my Asterisk server and that allows me to connect one/more (cordless) PSTN-phones in such a way that they'll work with SIP/Asterisk. I know there are USB-phones, but what I'm looking for is 'the USB-phone without the phone', if you know what I mean... ;-) You're looking for a USB FXS port. Yes, they do exist. You can take a look at the Astribank-8 from Xorcom (www.xorcom.com). I really don't know how well they work as I haven't any personal experience with their equipment, but they were exhibiting this solution at the last Astricon a few months back. -- 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 ___ --Bandwidth 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] hint on Zap channels
C F wrote: Kevin, I'm not sure this would work here, but maybe it would. There was a bug posted about being able to use hint against local channels, would that not help him? http://bugs.digium.com/view.php?id=5779nbn=4 No, the issue is that multiple ISDN devices are not distinct channels as far as Asterisk is concerned; they are all 'Zap/1' with different extensions behind that channel. This is the same question as asking 'if I have a PRI connected to my Panasonic PBX, can I use hints for all the extensions on that PBX'. It won't work in Asterisk, because it's not aware of the actual endpoints, only the channel that connects to them. ___ --Bandwidth 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: Does hardware like this exist...?
Evert Meulie wrote: That unit looks VERY promising! Thanks! :-) Would anyone happen to know an approx. price for a unit like this? Anyone? I bet the manufacturer of the unit would know a price for it, and it's probably even exact, not approximate :-) Since the manufacturer hasn't posted a price, it's likely that nobody knows yet... ___ --Bandwidth 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] Mediatrix 1204 help please.
OK we need some help in setting up a good wiki-info page for setting up the Mediatrix 1204 to work with asterisk. If anyone has set these unit's up and have them working please post your settings here so we can create a page on the wiki. These unit's are being sold to be used via sip format with asterisk and there is no real information on getting them working. At present there one of the worst I have run into to get correctly working. These are very expensive and some of us can't afford to send them back for a restocking fee. If someone working with Mediatrix has a white paper on getting these unit's working please let us know the link for it. It would be very helpful for many asterisk users. If you search the -users archives, you'll see where a couple of people have made them work. I believe there was at least one posting reflecting a working config. I did an eval on the 1204 in early 2004, but did not care for the way it interfaced with asterisk. The 1204 was really intended to interoperate with the 1104 as a toll bypass box. I was able to make it work and the audio was excellent with no echo whatsoever. Key items (in early 2004) included: - the 1204 does not have any sip register functions. One must configure it (and asterisk) to work with static IP addresses (instead of relying on the registration process). - calls from asterisk to (or through) the 1204 are treated as a group and the 1204 chooses the first available pstn port for all calls. If you want to direct a call to a specific port, one has to jump through hoops to force a CallerID (from asterisk) and then program 1204 to look for the callerid (which is then used to match a port number). Not cool. - programming the 1204 could only be done via snmp, and the snmp facility provided only ran on Windows. Each firmware upgrade to the 1204 required a new snmp implementation as the mib variables constantly changed. The snmp community string (eg, password) could not be changed from public, therefore exposing the 1204 to the internet would be a major security risk. (If you know snmp extemely well, you can use the mib definitions within a linux system to program the box, but you better be very good at snmp to do that.) - Support for the box is only offered through resellers, and their typical resellers are those firms reselling traditional pbx's. A fair number of those don't have a clue what voip is about and even fewer can spell asterisk. - All firmware upgrades are chargable regardless of what problem might be found. The upgrade charge was very high (something like $500 in 2004). Given the above (in 2004), the risk associated with using the 1204 was far to great and I returned the unit for full credit. (The eval was arranged through Mediaxtix sales rep even though the unit came from a reseller.) I've not touched or seen the 1204 since early 2004, so can't help any more then what is stated above. The product may have improved since then, but I don't have clue what might have changed. ___ --Bandwidth 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] Shutting down Asterisk when not in RTP Stream
Ya.. I also faced the same problem when running asterisk HA cluster. the workaround I did was to use a script to shut down network service first, then asterisk so that the BYE doesn't reach the client and then again start the network service (I needed to login remotely) Hope this helps --- Douglas Garstang [EMAIL PROTECTED] wrote: G! Asterisk sends a BYE to the phone when it gets shut down. What a pain. Eventhough it isn't in the RTP path, it must keep track of it's current call state, and when you shut it down, terminate all those calls. Reason I am trying this is that I've had asterisk core dump on me a few times, and I'd like to be able to restart it without losing calls in progress. Doug. -Original Message- From: Douglas Garstang Sent: Thursday, December 15, 2005 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream I'm very confused about something. I have two phones that have reinvited and have an RTP session open. I confirmed this by running ngrep on the Asterisk box. Asterisk still shows the calls on the console. *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.10.125 a00090201 45dfabad1bd 00103/0 ulaw No Tx: ACK 192.168.10.4 a00090101 ca3279d8-3e 00102/1 ulaw No Tx: ACK When I shut asterisk down, the call terminates. I don't understand that. If Asterisk isn't in the RTP path, how can shutting it down terminate an active call? Don't know if it's relevant, but the 192.168.10.4 is an OpenSER box. Thanks. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ 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] HW Echo Cancellers
On 12/16/05, Darren Wright [EMAIL PROTECTED] wrote: I have used the orion...you can buy right from them. However, I was not impressed with their sales teamI have one on a beta test, and they threatened to call a collection agency in when I refused paybent before the beta expired. Can you give an indication of price for their units? I've tried mailing a couple of times, but received no answer. I am just interested to know what price range we'd be looking at. Thanks, 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
Re: [Asterisk-Users] about g729
Hi Could you point me to where i can get this ram On 12/16/05, Martin Joseph [EMAIL PROTECTED] wrote: On Dec 8, 2005, at 3:27 AM, Andrea Riela wrote: snipWith g711 all works like a charm, but for audio quality, and bandwidth utilization, I'm trying now to work with g729 between CME and ISP. What about Asterisk? this is a pass-thru example, or maybe I've to pay a g729 license?Yes,you need to buy the codec for $10(us) per channel if you want to be able to translate g729.I purchased the unsupported OSX versionof the codec and it seems to work great and solved or improved manyquality issues I was seeing.Marty___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.2.0 queue.conf exit context
Anything else funky I need to do to get the exit context in
queues.conf working? I have the exit context defined, but when I'm
in the queue, I press 1 or 2 but it keeps me in the queue. No break
in musiconhold or anything. It's like the queue is ignoring/not
recognizing my keypress. Had this working a few CVS revs ago.
Keypress during other menus is fine so I don't think it's a DTMF
issue, but I don't know that much about it so I may be wrong.
; queues.conf
[supportq-emergency]
periodic-announce = supportq-emergency-periodic-announce
periodic-announce-frequency = 60
musiconhold = default
announce = supportq-emergency-agent-announce
strategy = ringall
context = supportq-emergency
timeout = 15
retry = 5
announce-frequency = 0
announce-holdtime = no
monitor-format = gsm
monitor-join = yes
joinempty = strict
member = Local/[EMAIL PROTECTED]
; extensions.conf
[supportq-emergency]
exten = 1,n,Wait(0.5)
exten = 1,n,Voicemail(u9000)
exten = 1,n,Hangup
exten = 2,n,Wait(0.5)
exten = 2,n,Playback(supportq-page-menu)
exten = 2,n,Playback(beep)
exten = 2,n,Read(CALLBACKNUM||20)
exten = 2,n,Playback(callbacknumber-recorded-as)
exten = 2,n,SayDigits(${CALLBACKNUM})
exten = 2,n,Playback(correct-1-reenter-2)
exten = 2,n,Read(1OR2||1)
exten = 2,n,GotoIf($[${1OR2} = 1 ] ?:1)
exten = 2,n,System(echo ${CALLERIDNAME} @ ${CALLBACKNUM} has paged
you via the Emergency Support Queue. | mail [EMAIL PROTECTED])
exten = 2,n,Playback(pagesent)
exten = 2,n,Hangup()
exten = i,1,Playback(pm-invalid-option)
exten = t,1,Goto(1,1)
___
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RE: [Asterisk-Users] HW Echo Cancellers
$1k for a single port T1 I've gone down the Tellabs route, and am infinitely more happy.thanks C F for the docs.. -D From: [EMAIL PROTECTED] on behalf of Steve Davies Sent: Fri 12/16/2005 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HW Echo Cancellers On 12/16/05, Darren Wright [EMAIL PROTECTED] wrote: I have used the orion...you can buy right from them. However, I was not impressed with their sales teamI have one on a beta test, and they threatened to call a collection agency in when I refused paybent before the beta expired. Can you give an indication of price for their units? I've tried mailing a couple of times, but received no answer. I am just interested to know what price range we'd be looking at. Thanks, 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 winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Having trouble calling out from Zap channel
I'm trying to dial from an SIP phone to another PBX through a PRI line The SIP phone gets the recording All circuits are busy now Here is the error at the CLI --- -- Executing SetVar(SIP/200-cbca, OUTNUM=9304752) in new stack -- Executing Cut(SIP/200-cbca, custom=OUT_1|:|1) in new stack -- Executing GotoIf(SIP/200-cbca, 0?16) in new stack -- Executing Dial(SIP/200-cbca, ZAP/g0/9304752) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(SIP/200-cbca, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/200-cbca, Dial failed due to CHANUNAVAIL) in new s the log file /var/log/asterisk/full has the error unable to create channel of type 'Zap' in it. The PRI span appears to be up and functioning. here is the output from pri show span 1 - asterisk1*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: Lucent 5E Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 - Here is the output from pri intense debug span 1 Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- Restarting T203 counter asterisk1*CLI [ 02 01 01 01 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- ACKing all packets from 0 to (but not including) 0 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (0) [ 02 01 01 01 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- Restarting T203 counter asterisk1*CLI [ 02 01 01 01 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- ACKing all packets from 0 to (but not including) 0 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter - I'm brand new to asterisk and using [EMAIL PROTECTED] Does anyone know what the next step in debuging this problem would be? -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2.0 queue.conf exit context
Jason Lixfeld wrote:
; extensions.conf
[supportq-emergency]
exten = 1,n,Wait(0.5)
exten = 1,n,Voicemail(u9000)
exten = 1,n,Hangup
exten = 2,n,Wait(0.5)
exten = 2,n,Playback(supportq-page-menu)
exten = 2,n,Playback(beep)
exten = 2,n,Read(CALLBACKNUM||20)
exten = 2,n,Playback(callbacknumber-recorded-as)
exten = 2,n,SayDigits(${CALLBACKNUM})
exten = 2,n,Playback(correct-1-reenter-2)
exten = 2,n,Read(1OR2||1)
exten = 2,n,GotoIf($[${1OR2} = 1 ] ?:1)
exten = 2,n,System(echo ${CALLERIDNAME} @ ${CALLBACKNUM} has paged
you via the Emergency Support Queue. | mail [EMAIL PROTECTED])
exten = 2,n,Playback(pagesent)
exten = 2,n,Hangup()
exten = i,1,Playback(pm-invalid-option)
exten = t,1,Goto(1,1)
Your '1' and '2' extensions don't have a priority 1 step to start from,
ergo, they don't exist as far as app_queue is concerned.
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Re: [Asterisk-Users] 1.2.0 queue.conf exit context
Thought it had to be something simple, thanks.
Still wrapping my head around the n priority.
On 16-Dec-05, at 11:40 AM, Kevin P. Fleming wrote:
Jason Lixfeld wrote:
; extensions.conf
[supportq-emergency]
exten = 1,n,Wait(0.5)
exten = 1,n,Voicemail(u9000)
exten = 1,n,Hangup
exten = 2,n,Wait(0.5)
exten = 2,n,Playback(supportq-page-menu)
exten = 2,n,Playback(beep)
exten = 2,n,Read(CALLBACKNUM||20)
exten = 2,n,Playback(callbacknumber-recorded-as)
exten = 2,n,SayDigits(${CALLBACKNUM})
exten = 2,n,Playback(correct-1-reenter-2)
exten = 2,n,Read(1OR2||1)
exten = 2,n,GotoIf($[${1OR2} = 1 ] ?:1)
exten = 2,n,System(echo ${CALLERIDNAME} @ ${CALLBACKNUM} has
paged you via the Emergency Support Queue. | mail [EMAIL PROTECTED])
exten = 2,n,Playback(pagesent)
exten = 2,n,Hangup()
exten = i,1,Playback(pm-invalid-option)
exten = t,1,Goto(1,1)
Your '1' and '2' extensions don't have a priority 1 step to start
from, ergo, they don't exist as far as app_queue is concerned.
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Re: [Asterisk-Users] HW Echo Cancellers
On 12/16/05, Darren Wright [EMAIL PROTECTED] wrote: $1k for a single port T1 I've gone down the Tellabs route, and am infinitely more happy.thanks C F for the docs.. Tellabs looks a little too up-scale for what I need :). $1k for a single port orion unit might be worth considering for really stubborn installs though. Does anyone else have suggestions for external E1 hardware echo canceller solutions? In my case, I would be interested in one or two port desktop offerings, but I'm interested in larger scale for research purposes. Thanks again. 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] Music On Hold
Help! No Music on Hold. Probably a novice mistake but I cant figure it out. Here are the details: CentOS 4.2 Asterisk 1.2.1 (Do I need to do something to get MOH to build?) Ztdummy loaded (conference works fine) musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 Sip device (x-lite also tried with an ATA) with canreinvite=no: sip.conf: [7211] username=7211 secret= host=dynamic type=friend context=standardphone disallow=all allow=gsm allow=ulaw allow=alaw allow=g723.1 allow=g729 canreinvite=no Extensions.conf: exten = 8702,1,Answer() exten = 8702,n,MusicOnHold(default) exten = 8702,n,Hangup() # asterisk -r Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.2.1 currently running on ccsip (pid = 4782) Verbosity is at least 3 == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-be01' -- Executing Answer(SIP/7211-cedb, ) in new stack -- Executing MusicOnHold(SIP/7211-cedb, default) in new stack -- Started music on hold, class 'default', on channel 'SIP/7211-cedb' -- Stopped music on hold on SIP/7211-cedb == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-cedb' The Stopped music on hold happens immediately like it cant find something. Should I give up and use madplay? -- Bud ___ --Bandwidth 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 won´t load module codec_g729a.so
Hello List, my Asterisk will not load the module codec_g729a.so asterisk3*CLI load codec_g729a.so Unable to load module codec_g729a.so What did i do wrong? I followed the README File from Digium step by step. cheers klaus begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth 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
Wich player do you use?
I use the one that is coming with Asterisk. Just cd to the Asterisk
Sources, make mpg123, cd mpg..., make make install
and im done. It worked fine all the time.
cheers
klaus
Bud Bach schrieb:
Help! No Music on Hold.
Probably a novice
mistake but I cant figure it out. Here are the details:
CentOS 4.2
Asterisk 1.2.1 (Do I need
to do something to get MOH to
build?)
Ztdummy loaded
(conference works fine)
musiconhold.conf:
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
Sip device (x-lite also
tried with an ATA) with canreinvite=no:
sip.conf:
[7211]
username=7211
secret=
host=dynamic
type=friend
context=standardphone
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=g723.1
allow=g729
canreinvite=no
Extensions.conf:
exten =
8702,1,Answer()
exten =
8702,n,MusicOnHold(default)
exten =
8702,n,Hangup()
# asterisk -r
Asterisk 1.2.1, Copyright
(C) 1999 - 2005 Digium.
Written by Mark Spencer
[EMAIL PROTECTED]
=
Connected to Asterisk
1.2.1 currently running on ccsip (pid
= 4782)
Verbosity is at least 3
== Spawn extension
(standardphone, 8702, 2) exited
non-zero on 'SIP/7211-be01'
-- Executing
Answer("SIP/7211-cedb",
"") in new stack
-- Executing
MusicOnHold("SIP/7211-cedb",
"default") in new stack
-- Started music on
hold, class
'default', on channel 'SIP/7211-cedb'
-- Stopped music on
hold on SIP/7211-cedb
== Spawn extension
(standardphone, 8702, 2) exited
non-zero on 'SIP/7211-cedb'
The Stopped music on
hold happens immediately
like it cant find something. Should I give up and use madplay?
-- Bud
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begin:vcard
fn:Klaus Peras
n:Peras;Klaus
org:HOB;Netzwerk Support
adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany
email;internet:[EMAIL PROTECTED]
tel;work:09103 / 715 - 329
url:http://www.hob.de
version:2.1
end:vcard
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RE: [Asterisk-Users] Music On Hold
Now Im really baffeled. I found some comments at the end of the musiconhold.config file about the native format. I copied the files in /var/lib/asterisk/mohmp3 to /var/lib/asterisk/moh-native (just cpt them). Then I uncommented the section in musiconhold.config: [native] mode=files directory=/var/lib/asterisk/moh-native and change the dialplan to: exten = 8702,n,MusicOnHold(native) And it works. Now, how do I make native the default? I tried to copy: mode=files directory=/var/lib/asterisk/moh-native from the native section to the default section and that didnt work -- Bud Dec 16 12:17:38 WARNING[6222]: interface.c:215 decodeMP3: Junk at the beginning of frame -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bud Bach Sent: Friday, December 16, 2005 10:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music On Hold Help! No Music on Hold. Probably a novice mistake but I cant figure it out. Here are the details: CentOS 4.2 Asterisk 1.2.1 (Do I need to do something to get MOH to build?) Ztdummy loaded (conference works fine) musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 Sip device (x-lite also tried with an ATA) with canreinvite=no: sip.conf: [7211] username=7211 secret= host=dynamic type=friend context=standardphone disallow=all allow=gsm allow=ulaw allow=alaw allow=g723.1 allow=g729 canreinvite=no Extensions.conf: exten = 8702,1,Answer() exten = 8702,n,MusicOnHold(default) exten = 8702,n,Hangup() # asterisk -r Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.2.1 currently running on ccsip (pid = 4782) Verbosity is at least 3 == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-be01' -- Executing Answer(SIP/7211-cedb, ) in new stack -- Executing MusicOnHold(SIP/7211-cedb, default) in new stack -- Started music on hold, class 'default', on channel 'SIP/7211-cedb' -- Stopped music on hold on SIP/7211-cedb == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-cedb' The Stopped music on hold happens immediately like it cant find something. Should I give up and use madplay? -- Bud ___ --Bandwidth 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
Thanks Klaus. I missed the make mpg123 step! -- Bud -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Peras Sent: Friday, December 16, 2005 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Music On Hold Wich player do you use? I use the one that is coming with Asterisk. Just cd to the Asterisk Sources, make mpg123, cd mpg..., make make install and i´m done. It worked fine all the time. cheers klaus Bud Bach schrieb: Help! No Music on Hold. Probably a novice mistake but I cant figure it out. Here are the details: CentOS 4.2 Asterisk 1.2.1 (Do I need to do something to get MOH to build?) Ztdummy loaded (conference works fine) musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 Sip device (x-lite also tried with an ATA) with canreinvite=no: sip.conf: [7211] username=7211 secret= host=dynamic type=friend context=standardphone disallow=all allow=gsm allow=ulaw allow=alaw allow=g723.1 allow=g729 canreinvite=no Extensions.conf: exten = 8702,1,Answer() exten = 8702,n,MusicOnHold(default) exten = 8702,n,Hangup() # asterisk -r Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.2.1 currently running on ccsip (pid = 4782) Verbosity is at least 3 == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-be01' -- Executing Answer(SIP/7211-cedb, ) in new stack -- Executing MusicOnHold(SIP/7211-cedb, default) in new stack -- Started music on hold, class 'default', on channel 'SIP/7211-cedb' -- Stopped music on hold on SIP/7211-cedb == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-cedb' The Stopped music on hold happens immediately like it cant find something. Should I give up and use madplay? -- Bud ___--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] CID lookup from an Exchange Public folder
Has anybody done this? I looked at LDAP but you cant get to them that way, Im considering either a timed export, or some other way (can you access them via IMAP? Or by wget on the owa web structure?) Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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] Amtelco Infinity
Does anyone have any experience hooking * box up to an Infinity system? -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CID lookup from an Exchange Public folder
OWA sucks big time youll never get it to run right. If you use CDO in an ASP script, you can programmatically access basically every structure in Exchange. If you call a shell script via AGI with wget in it, that would call the ASP script which would filter items in the Exchange store via CDO, then return the item you want using SET VARIABLE syntax to Asterisk. Problem is there is massive overhead in CDO (its basically a MAPI client) so the latency would be so bad that it would never work in near real time and scaling would be a problem from the get-go. I do a caller ID lookup from our SQL server using ODBCSockets direct to Asterisk and it is under a second, with no scaling problems. This approach would work fine for you, as long as you had a _vbscript_ that you could run say once a day that would do the CDO, get the contact items, wipe all of the records from the SQL server table, and repopulate the table. This solves your latency problem. If you dont have SQL you can get the free SQL 2005 download from Microsoft. If you want, email me offlist and I will send you my ODBCSockets script. -Original Message- From: Steve Hanselman [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] CID lookup from an Exchange Public folder Has anybody done this? I looked at LDAP but you cant get to them that way, Im considering either a timed export, or some other way (can you access them via IMAP? Or by wget on the owa web structure?) Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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] Digium TE205 Card
Hello Everyone, I have a brand new Digium TE205 card, bought 2 days back, stillunopenedand for sale. Reason for selling is we need a quad span ISDN card now instead of dual span ISDN card. Selling it at USD700, this card retails at around USD900+ right now.Card was directlypurchased from Digium and their invoice will be supplied as proof. If you are interested, leave me an email at [EMAIL PROTECTED]or call me at 718 2336260x 120. With regards, San ___ --Bandwidth 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] Aastra 480i
I have an Aastra 480i that used to have firmware 1.0.0... on it. I got the new 1.3 firmware and had the phone fetch that from my TFTP server, but after running about 15s, it stops. No more downloading, no response to WebGUI, no response to buttons, nothing. I rebooted it (not a good idea, I know) and it complained that there was no application and tried to reload from tftp server again. Same thing happened. So I tried the firmware for the 480i CT IP phone, but that did the same thing. The little wheel spins on the display when it boots but it stops when the download stops. Anyone have any advice? I just want the firmware in there, I am happy to manually configure the phone with the UI... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth 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] hint on Zap channels
On Fri, December 16, 2005 16:39, Kevin P. Fleming said: C F wrote: Kevin, I'm not sure this would work here, but maybe it would. There was a bug posted about being able to use hint against local channels, would that not help him? http://bugs.digium.com/view.php?id=5779nbn=4 No, the issue is that multiple ISDN devices are not distinct channels as far as Asterisk is concerned; they are all 'Zap/1' with different extensions behind that channel. This is the same question as asking 'if I have a PRI connected to my Panasonic PBX, can I use hints for all the extensions on that PBX'. It won't work in Asterisk, because it's not aware of the actual endpoints, only the channel that connects to them. I personally think this is a fault in (*). (Or rather Zaptel) Because there is such a thing as ISDN, I think it should be able to recognize separate channels for DIDs... Both internal and external ZAP channels should be able to recognise the different DID/CID/CLID as separate identifiable endpoints. That way you can chose a 'channel' and have (*) use the correct CID/CLID. When doing extensions it should dial it, when doing outbound the chosen channel could define which MSN/CLID to use, inbound the DID would define the channel. (Just like the way it does now for the channels/extensions, but for ISDN just dialing Zap/1 won't do the trick... You'll need to dial Zap/1/2020 to get the ISDN phone with MSN 2020) Just my EUR 0,02 -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --Bandwidth 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] Experience sharing on Planet VIP-450 + Asterisk
Jason Chan (jasonOfficial) wrote: What's next? Well... thanks to the buggy firmware and imcompatable standard with Asterisk... First of all, I can't deny that Planet VIP-450 does a good job in packetizing voice stream, the voice quality is really good and delay is really small. Also the hardware itself is quite robust, it seldom halt.. (the machine has been up for a few days). Also it is quite feature-rich, I can say. BUT I think there is quite a number of BUGS in the firmware! In order to see which kind of DTMF Relay it is using, I have done a packet analysing. When I try to pass SIP INFO type DTMF band to VIP-450, it replies "501 Unimplemented". Also when I try to pass DTMF from my POTS phone via the FXO port, only RTP payload can be seen in the packet captures. I DID suspect that it is RFC2833, because as far as I know RFC2833 did have the DTMF textx inside the RTP packet somewhere (seems header). But asterisk just simply did not regconize them (of coz I have set DTMFmode=rfc2833)! It is pretty strange that the user manual states "VIP handles DTMF Relay per SIP specification". So VIP-450 actually is using what kind of SIP specification? Sounds familiar.See below. How about using its Inband DTMF relay? This will certainly generate strange warning just like my case : improper ilbc frame size and tell me to use u-law to do DTMF even if I AM using G.711 u-law. It is seems that the DTMF tone generated by VIP-450 generate is kinda strange... So the final solution is, SIMPLY SWITCH OFF THE DTMF RELAY IN VIP-450. Please try to type "show coding" in console mode and you will see a lot of coding (codec) profiles. Most of them are with DTMF relay. Just switch off them by "set coding profile id dtmf_relay off" (please check with the manual). If you want to stop certain codec, just simply make that coding profile unusable in voice. For example, "set coding profile id voice off". If youonly turn on the profile withu-law, the SIP header it issues will just consist of 0x4 (ulaw) codec, not 0x105. This is what got my attention.Take a look at the commands that I use for the Yoda VG-400. If I'm not mistaken, they're exactly the same for the Planet.Same firmware libraries I presume. http://lists.digium.com/pipermail/asterisk-users/2005-August/120588.html If you notice, I also set dtmf_relay off.Too bad you didn't post any commands earlier, you would have saved a lot of time. Looks like we should compare notes. In mypoint of view, Planet isexpectingthis deviceisconnected to another VIP-450, not really for Asterisk or anything else, even not fora soft phone. Certainly this is not enough for everyone, at leastI can't do any IVR and something what a PBX should have (just like what I can do in Asterisk). I hope my experience will help anyone who is using VIP-450 with Asterisk, just like me. I have done Googling for 3 days but I can search for nothing related to this issue. Sorry for my poor written English. Cheers, Jason Chan, Hong Kong This is exactly what Yoda wants as well. I remember planning to buy a Planet unit during my love-hate relationship with Yoda and Asterisk but I'm now glad I didn't. It seems I would have had to tackle the same problems. I'm just happy the units I have work well. Thanks to Yoda's support. They're set on GnuGK rather than Asterisk so it was a first for me as well as them. Xie Xie -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth 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] .call files on PRI not waiting for answer in de sired context
If I generate a .call file to an external callee through my PRI, Asterisk will not wait to execute the priority in the target context, and instead will continue on as soon as the channel is dialled. I want it to wait for an answer, THEN continue. It detects the answer correctly. I have callprogress=yes in Zapata.conf. I have read the wiki with respect to this issue. Weird thing is, I swear this worked the way I wanted it to when I was running 1.0.9, now it's not under 1.2 beta 1. Am I crazy? ___ --Bandwidth 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] Merlin Legend mode codes
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I am getting ready to replace a 4 port vm system (audix) with asterisk and have been looking for the mode codes that the legend uses. Has anyone done this and would you mind sharing your extensions.conf for this? Thanks, Sean -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDoxcFy9wPyZpnL2URAhL0AJ4k1qaxwpzWP++iTModaB9xIyA5oQCePq7n G93wjWhufueqli0pbAT895E= =EyxP -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint on Zap channels
Francesco Peeters (Asterisk) wrote: I personally think this is a fault in (*). (Or rather Zaptel) You are certainly welcome to your opinion, but thinking that Asterisk should understand the concept of 'remote endpoints' as native devices is by no means a 'fault'. If nobody has wanted this enough before to be able to code it up and submit it, then it's just a lack of functionality. Because there is such a thing as ISDN, I think it should be able to recognize separate channels for DIDs... Both internal and external ZAP channels should be able to recognise the different DID/CID/CLID as separate identifiable endpoints. That way you can chose a 'channel' and have (*) use the correct CID/CLID. And how would Asterisk know when these endpoints communicate directly with each other to keep trace of device state? It would certainly be possible to do what you want, but it would need to be implemented by the Zaptel driver that is communicating with that ISDN interface, so it can present distinct 'channels' to chan_zap for each device on the ISDN bus. ___ --Bandwidth 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 files on PRI not waiting for answer in de sired context
Colin Anderson wrote: Weird thing is, I swear this worked the way I wanted it to when I was running 1.0.9, now it's not under 1.2 beta 1. Am I crazy? I'm not saying this has been fixed since that point, but why in the world are you running 1.2.0 beta 1 when 1.2.1 has been released? ___ --Bandwidth 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] hint on Zap channels
On Fri, December 16, 2005 20:44, Kevin P. Fleming said: Francesco Peeters (Asterisk) wrote: I personally think this is a fault in (*). (Or rather Zaptel) You are certainly welcome to your opinion, but thinking that Asterisk should understand the concept of 'remote endpoints' as native devices is by no means a 'fault'. If nobody has wanted this enough before to be able to code it up and submit it, then it's just a lack of functionality. OK, Maybe fault wasn't the right word here... Lacking is probably better... I'd love to look in to it and code it, but I simply haven't got the time to investigate and code it... Because there is such a thing as ISDN, I think it should be able to recognize separate channels for DIDs... Both internal and external ZAP channels should be able to recognise the different DID/CID/CLID as separate identifiable endpoints. That way you can chose a 'channel' and have (*) use the correct CID/CLID. And how would Asterisk know when these endpoints communicate directly with each other to keep trace of device state? Because it would either be the device in NT mode, and therefore initiate the connection, and be able to see the data flows. Or it would be TE mode, but still on the same bus (which means it'll still see the data) It would certainly be possible to do what you want, but it would need to be implemented by the Zaptel driver that is communicating with that ISDN interface, so it can present distinct 'channels' to chan_zap for each device on the ISDN bus. That's why I said 'Or rather Zaptel' in my original comment... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --Bandwidth 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 files on PRI not waiting for answer in desired context
Hey, baby steps. Truth is I've been too busy and I don't have a pressing need to upgrade. Everything's working fine (except this, of course!) -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] .call files on PRI not waiting for answer in de sired context Colin Anderson wrote: Weird thing is, I swear this worked the way I wanted it to when I was running 1.0.9, now it's not under 1.2 beta 1. Am I crazy? I'm not saying this has been fixed since that point, but why in the world are you running 1.2.0 beta 1 when 1.2.1 has been released? ___ --Bandwidth 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] Aastra 480i
On Fri, 2005-12-16 at 14:20 -0500, Michael George wrote: I have an Aastra 480i that used to have firmware 1.0.0... on it. I got the new 1.3 firmware and had the phone fetch that from my TFTP server, but after running about 15s, it stops. No more downloading, no response to WebGUI, no response to buttons, nothing. I rebooted it (not a good idea, I know) and it complained that there was no application and tried to reload from tftp server again. Same thing happened. So I tried the firmware for the 480i CT IP phone, but that did the same thing. The little wheel spins on the display when it boots but it stops when the download stops. I am having exactly the same problem with some 9133i phones. I bought 6 of them and three upgraded without problem and the other three will not finish the firmware download even after three hours. One of them did upgrade when I plugged it in two days later but the other two are still useless. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] .call files on PRI not waiting for answer in de sired context --ResponseTimeout the best answer?
Hmmm seems like every dialplan snippet I've seen so far relies on ResponseTimeout and looping back to s,1. Is this the only way I can get this to work kind-of the way I want? Any ideas welcome. -Original Message- From: Colin Anderson [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 12:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] .call files on PRI not waiting for answer in de sired context If I generate a .call file to an external callee through my PRI, Asterisk will not wait to execute the priority in the target context, and instead will continue on as soon as the channel is dialled. I want it to wait for an answer, THEN continue. It detects the answer correctly. I have callprogress=yes in Zapata.conf. I have read the wiki with respect to this issue. Weird thing is, I swear this worked the way I wanted it to when I was running 1.0.9, now it's not under 1.2 beta 1. Am I crazy? ___ --Bandwidth 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] FOP button limit?
Cant seem to get an answer anywhere else so hopefully somewhere here will have a clue. With Flash Operator panel, if you have too many extensions or trunks, the last ones wrap back to the beginning and cover the first ones. I have played with rectangle sizes with no luck. Anyone have a clue on how to fit more buttons on the screen? -Kerry ___ --Bandwidth 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] FOP button limit?
Kerry Garrison wrote: Cant seem to get an answer anywhere else so hopefully somewhere here will have a clue. With Flash Operator panel, if you have too many extensions or trunks, the last ones wrap back to the beginning and cover the first ones. I have played with rectangle sizes with no luck. Anyone have a clue on how to fit more buttons on the screen? Bigger screen? :D -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Terry Gilsenan Information Systems Manager InterOil Corporation ph: +61-7-4046-4614 mb: +61-417-600-360 ===[Disclaimer]=== This electronic transmission, including any attachments, is confidential, may contain privileged information and should be read or retained only by the intended recipient. If you received this message in error, please delete it from your system and notify the sender immediately. Any review, dissemination or other use of this information by persons or entities other than the intended recipient is strictly prohibited. ===[End]=== ___ --Bandwidth 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 + H323 + 723
Hi I am using asterisk 1.2.1, does any one has any luck with asterisk and h323. I want to use the codecs 723 and 729 with it. I am having one way audio issues with oh323 with I receive a call to asterieks through 723 . is there a successful implementation ? regards kani ___ --Bandwidth 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-1.2.1 incomplete DID number on incoming T1 line
I am presently running asterisk 1.0.9 with AMP and a Sangoma A101 card with a T1 line and 12 channels. It has run solidly for four months. It receives a 4 digit DID number 9140. zaptel.conf span=1,0,0,esf,b8zs em=1-12 I have been testing Asterisk-1.2.1 on a duplicate set of hardware using the same Sangoma card and AMP-1.10.010. Internally the system works fine as well as Voicemail, but when receiving an outside call, the first call rings into the system as normal. The second, third, fourth, etc call have only the first DID digit 9 and then the system hangs up. The caller hears a busy signal. If I restart Asterisk then I can repeat the process of one good call and the rest busy signals. I have been in contact with Sangoma, but am not sure that it is a problem with their software. Any thoughts. Thanks. /var/log/asterisk/full [snip] First Call** Dec 15 05:44:16 VERBOSE[31698] logger.c: -- Starting simple switch on 'Zap/12-1' Dec 15 05:44:16 DEBUG[31680] acl.c: # Testing 192.168.10.135 with 192.168.0.0 Dec 15 05:44:16 DEBUG[31698] chan_zap.c: DTMF digit: 9 on Zap/12-1 Dec 15 05:44:16 DEBUG[31698] chan_zap.c: DTMF digit: 1 on Zap/12-1 Dec 15 05:44:17 DEBUG[31698] chan_zap.c: DTMF digit: 4 on Zap/12-1 Dec 15 05:44:17 DEBUG[31698] chan_zap.c: DTMF digit: 0 on Zap/12-1 Dec 15 05:44:17 DEBUG[31698] chan_zap.c: Enabled echo cancellation on channel 12 Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing SetVar(Zap/12-1, FROM_DID=9140) in new stack Dec 15 05:44:17 WARNING[31698] pbx.c: SetVar is deprecated, please use Set instead. Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing Goto(Zap/12-1, from-pstn|s|1) in new stack Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Goto (from-pstn,s,1) Dec 15 05:44:17 DEBUG[31698] pbx.c: Expression result is '0' Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing GotoIf(Zap/12-1, 0?from-pstn-reghours|s|1:) in new stack Dec 15 05:44:17 DEBUG[31698] pbx.c: Not taking any branch Dec 15 05:44:17 DEBUG[31698] pbx.c: Expression result is '0' Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing GotoIf(Zap/12-1, 0?from-pstn-afthours|s|1:) in new stack Dec 15 05:44:17 DEBUG[31698] pbx.c: Not taking any branch Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing GotoIfTime(Zap/12-1, 8:00-17:00|mon-fri|*|*?from-pstn-reghours|s|1:) in new stack Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing Goto(Zap/12-1, from-pstn-afthours|s|1) in new stack Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Goto (from-pstn-afthours,s,1) Dec 15 05:44:17 DEBUG[31698] pbx.c: Expression result is '1' Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing GotoIf(Zap/12-1, 1?from-pstn-afthours-nofax|s|1:2) in new stack Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Goto (from-pstn-afthours-nofax,s,1) Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing SetVar(Zap/12-1, intype=aa_2) in new stack Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing Cut(Zap/12-1, intype=intype|-|1) in new stack Dec 15 05:44:17 WARNING[31698] app_cut.c: The application Cut is deprecated. Please use the CUT() function instead. Dec 15 05:44:17 DEBUG[31698] pbx.c: Expression result is '0' Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing GotoIf(Zap/12-1, 0?4:5) in new stack Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Goto (from-pstn-afthours-nofax,s,5) Dec 15 05:44:17 DEBUG[31698] pbx.c: Expression result is '0' Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing GotoIf(Zap/12-1, 0?6:7) in new stack Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Goto (from-pstn-afthours-nofax,s,7) Dec 15 05:44:17 DEBUG[31698] pbx.c: Expression result is '0' Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing GotoIf(Zap/12-1, 0?8:11) in new stack Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Goto (from-pstn-afthours-nofax,s,11) Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing Answer(Zap/12-1, ) in new stack Dec 15 05:44:17 DEBUG[31698] chan_zap.c: Took Zap/12-1 off hook Dec 15 05:44:17 VERBOSE[31698] logger.c: -- Executing Wait(Zap/12-1, 1) in new stack Dec 15 05:44:18 VERBOSE[31698] logger.c: -- Executing Goto(Zap/12-1, aa_2|s|1) in new stack Dec 15 05:44:18 VERBOSE[31698] logger.c: -- Goto (aa_2,s,1) *Second Call Dec 15 05:44:45 VERBOSE[31710] logger.c: -- Starting simple switch on 'Zap/12-1' Dec 15 05:44:45 DEBUG[31680] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Dec 15 05:44:45 DEBUG[31710] chan_zap.c: DTMF digit: 9 on Zap/12-1 Dec 15 05:44:45 DEBUG[31710] chan_zap.c: Enabled echo cancellation on channel 12 Dec 15 05:44:45 VERBOSE[31710] logger.c: -- Executing SetVar(Zap/12-1, FROM_DID=9140) in new stack Dec 15 05:44:45 VERBOSE[31710] logger.c: -- Executing Goto(Zap/12-1, from-pstn|s|1) in new stack Dec 15 05:44:45 VERBOSE[31710] logger.c: -- Goto (from-pstn,s,1) Dec 15 05:44:45 DEBUG[31710] pbx.c: Expression result is '0' Dec 15
Re: [Asterisk-Users] asterisk + H323 + 723
Hi, I had the same troubles too. It does not recognise correctly g723 with oh323. With h323 i have dtmf rfc2833 issues but g723 and 729 are transported correctly via H323 capabilities. So, let make a try with h323 included in asterisk branch, not the oh323 Kanishka Somaratne wrote: Hi I am using asterisk 1.2.1, does any one has any luck with asterisk and h323. I want to use the codecs 723 and 729 with it. I am having one way audio issues with oh323 with I receive a call to asterieks through 723 . is there a successful implementation ? regards kani ___ --Bandwidth 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] Central Registration mechanism
You cannot use Realtime with multiple Asterisk systems sharing SIP information (which includes registration). Yes, hard to believe I know. Ask Kevin Fleming or see a previous discuss thread about this topic last week. Digium have said that it will be the better part of a year to fix this flaw. Doug. -Original Message- From: Noel Athaide [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 6:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Central Registration mechanism Hello, I would like to know if there is any mechanism whereby one can have several Asterisk servers catering to different physical locations but only one central Asterisk server responsible for client registration. From what I have read on the mailing lists, the one way to do this is by using Realtime asterisk with a MySQL database. The other is to use SER as a frontend. Is there a simpler method? - Noel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Jitterbuffer and trunking
Richard Scobie wrote: Is there a way to configure the IAX jitterbuffer to get the benefit of trunktimestamps, while not having any jitterbuffering (reducing delay)? My SVN asterisk systems use the following topologies: 1) PolycomSIP - *1 -IAX- *2 - H323 Gateway 2) PolycomSIP - *1 -IAX- *3 - Zap TDM400 Analog 3) H323 Gateway - *2 -IAX- *3 - Zap TDM400 Analog In all the above, the primary jitter path is the IAX one and the codec is Alaw all the way. In an effort to reduce path delay and multiple jitterbuffering I have configured the following: On the basis that the Polycom IP500 phones have a decent jitterbuffer built in, Asterisk 1 has jitterbuffer=no in iax.conf. Asterisk 2 has the same setting as the H323 GW has it's own jitterbuffer. Asterisk 3 has jitterbuffer=yes in iax.conf, to buffer the Zap interface and provide PLC. I notice that zapata.conf has an entry jitterbuffers=4 by default - is this a different one in which case should it be turned off or is it setting parameters for the IAX JB? There's a few points in here so far: 1) the new jitterbuffer and trunktimestamps are independent settings, and have independent effect. You get the same effect with trunktimestamps (correct pass-through of frame timestamps), whether you use the jb or not. 2) The IAX jitterbuffer is disabled _by default_ (unless you use forcejitterbuffer), when a call is bridged from an IAX channel to another VoIP channel. So, you don't need to forcibly disable the jb in your case, it should automatically be disabled: In your cases, it would only ever be enabled on box *3, when a call comes in from IAX, and goes to zap. 3) Yes, the setting in zapata.conf is for 4 very small buffers, which are different than than the IAX jb. Looking at README.jitterbuffer: If you don't use trunktimestamps, there's lots of ways the jitterbuffer can get confused because timestamps aren't necessarily sent through the trunk correctly. This presumably means that if I want to use IAX trunking effectively, I have to enable the IAX JB on all Asterisks. No, you don't need to enable the jitterbuffers anywhere except on the last machine that's receiving VoIP (in your cases above, *3). I would expect trunktimestamps would help you if you're using trunking for the IAX links between your boxes. -SteveK ___ --Bandwidth 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 480i
I had a lot of issues with 480i too and this is how I resolved it: 1) Make sure that the file on the tftp server is called firmware followed by the type they suggest (I do not remember the type name) 2) Once this is done, your phones should download the firmware and reboot properly --- Carlos Chavez [EMAIL PROTECTED] wrote: On Fri, 2005-12-16 at 14:20 -0500, Michael George wrote: I have an Aastra 480i that used to have firmware 1.0.0... on it. I got the new 1.3 firmware and had the phone fetch that from my TFTP server, but after running about 15s, it stops. No more downloading, no response to WebGUI, no response to buttons, nothing. I rebooted it (not a good idea, I know) and it complained that there was no application and tried to reload from tftp server again. Same thing happened. So I tried the firmware for the 480i CT IP phone, but that did the same thing. The little wheel spins on the display when it boots but it stops when the download stops. I am having exactly the same problem with some 9133i phones. I bought 6 of them and three upgraded without problem and the other three will not finish the firmware download even after three hours. One of them did upgrade when I plugged it in two days later but the other two are still useless. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? 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] Central Registration mechanism
I've seen this come across the list several times, and I'm curious if it's just a fluke that ours has been working perfectly between two different servers running with the same realtime database for the past 6-8 months. Aaron Douglas Garstang wrote: You cannot use Realtime with multiple Asterisk systems sharing SIP information (which includes registration). Yes, hard to believe I know. Ask Kevin Fleming or see a previous discuss thread about this topic last week. Digium have said that it will be the better part of a year to fix this flaw. Doug. -Original Message- From: Noel Athaide [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 6:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Central Registration mechanism Hello, I would like to know if there is any mechanism whereby one can have several Asterisk servers catering to different physical locations but only one central Asterisk server responsible for client registration. From what I have read on the mailing lists, the one way to do this is by using Realtime asterisk with a MySQL database. The other is to use SER as a frontend. Is there a simpler method? - Noel. ___ --Bandwidth 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] Central Registration mechanism
I wish it would have fluked for me... :( -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Central Registration mechanism I've seen this come across the list several times, and I'm curious if it's just a fluke that ours has been working perfectly between two different servers running with the same realtime database for the past 6-8 months. Aaron Douglas Garstang wrote: You cannot use Realtime with multiple Asterisk systems sharing SIP information (which includes registration). Yes, hard to believe I know. Ask Kevin Fleming or see a previous discuss thread about this topic last week. Digium have said that it will be the better part of a year to fix this flaw. Doug. -Original Message- From: Noel Athaide [mailto:[EMAIL PROTECTED] Sent: Friday, December 16, 2005 6:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Central Registration mechanism Hello, I would like to know if there is any mechanism whereby one can have several Asterisk servers catering to different physical locations but only one central Asterisk server responsible for client registration. From what I have read on the mailing lists, the one way to do this is by using Realtime asterisk with a MySQL database. The other is to use SER as a frontend. Is there a simpler method? - Noel. ___ --Bandwidth 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] FOP button limit?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan Sent: Friday, December 16, 2005 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FOP button limit? Kerry Garrison wrote: Cant seem to get an answer anywhere else so hopefully somewhere here will have a clue. With Flash Operator panel, if you have too many extensions or trunks, the last ones wrap back to the beginning and cover the first ones. I have played with rectangle sizes with no luck. Anyone have a clue on how to fit more buttons on the screen? Bigger screen? :D Nope, just gives you bigger buttons :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
Re: [Asterisk-Users] Aastra 480i
On Fri, 2005-12-16 at 12:48 -0800, Dave wrote: I had a lot of issues with 480i too and this is how I resolved it: 1) Make sure that the file on the tftp server is called firmware followed by the type they suggest (I do not remember the type name) 2) Once this is done, your phones should download the firmware and reboot properly Actually this is no longer true since firmware 1.2. Now the firmware has to be named like the model of the phone and the extension .st for it to download automatically. So for the 480i you need a file called 480i.st and for the 9133i you need 9133i.st. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FOP button limit?
Kerry Garrison wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan Sent: Friday, December 16, 2005 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FOP button limit? Kerry Garrison wrote: Cant seem to get an answer anywhere else so hopefully somewhere here will have a clue. With Flash Operator panel, if you have too many extensions or trunks, the last ones wrap back to the beginning and cover the first ones. I have played with rectangle sizes with no luck. Anyone have a clue on how to fit more buttons on the screen? Bigger screen? :D Nope, just gives you bigger buttons :o Ah! Krahp. Sorry :| ===[Disclaimer]=== This electronic transmission, including any attachments, is confidential, may contain privileged information and should be read or retained only by the intended recipient. If you received this message in error, please delete it from your system and notify the sender immediately. Any review, dissemination or other use of this information by persons or entities other than the intended recipient is strictly prohibited. ===[End]=== ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
