[Asterisk-Users] RE: X100P help required
hello all, I m joined this asterisk group few months back. Actually i have installed asterisk on my PC and using X100P PSTN interface card to make this PC work as a single line VoIP gateway. Now i have one problem which is as follows: As i told i m using X100P card. I am getting good quality of speech from it whenever i m making call or receiving call through it. Now I have installed Soft X-Lite ip phone on 3 pc's. Now whenever i make call from analog phone to any IP phone, we can talk. After finishing talk I hang up the analog phone. Now when just after 2-3 mins when i want to make call to any IP phone, from my analog phone i m getting "engaged tone". I have to restart my asterisk server, to come out of that "engaged tone" . Now evertime i cant restart asterisk server. I usually uses following commands to start asterisk server each time: first "safe_asterisk" then "asterisk -r". So i m confused now, why this is happening? 1) Am i starting asterisk server in wrong way? or 2) Is there any problem with X100P card? Where i m wrong, i m not getting? hope all of u will help me to solve this problem. Thanks Tejas Relax. Yahoo! Mail virus scanning helps detect nasty viruses!___ --Bandwidth 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: where can i find all .C files
hi all, i m using debian to run my asterisk gateway.I want to make some customization in voicemail application.For that i need to modify voicmail's .C(source file) file. can any body tell me where exactly all .C files resides in the system.. thanks tejas __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.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] RE:how many calls Asterisk gateway can handle
hi all, I am newbie to asterisk. I have installed asterisk based VoIP gateway in my LAB. Now i want to how many simultaneous calls (internal and external) can this gateway can handle? hereby i m sending my system details: 1) asterisk gateway is running on P-IV 2.6GHz machine. 2) i have installed one X100P FXO card on my PC for PSTN connection. 3) i have installed 4-soft X-LITE phones on 4 different PCs. 4) I am using SIP protocol. 5) codec is G.711u. 1) Now can anyone tell me how many simultaneous calls can my asterisk gateway handle? 2) How is it to possible to simulate the performance of asterisk VoIP gateway? 3) Is any tool available so that it can generate many calls and i can check gateway performance? 4) I am thinking of SIPp tool? 5) can any body have idea that what changes i have to make in sip.c onf and extension.conf to register those calls generated by SIPp? I mean how asterisk server make enrty of those calls? Pls help me out.. thanks tejas Yahoo! Photos Ring in the New Year with Photo Calendars. Add photos, events, holidays, whatever.___ --Bandwidth 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: how many call an Asterisk gateway can handle
hi all, I have implemented asterisk VoIP gateway in my lab using X100P card.Now i m running this gateway on pentium-IV, 2.6 GHz PCit has severals othere applications running on it..I have installed 5 Soft SIP phones in the LabNow my question is Using above configuration how many simultaneous (internal and external) calls my gateway can handle? Is there any free tools available through which i can analyze and measure the performance of my gateway? pls help me... thanks tejas Yahoo! Photos Ring in the New Year with Photo Calendars. Add photos, events, holidays, whatever.___ --Bandwidth 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: simulator for asterisk gateway
hi all, I have implemented VoIP gateway using X100P card. i have downloaded 3 X-Lite phones on 3 different PCs. Because of X100P, i can make call to analog phone also. Now i want to simulate my VoIP gateway. how much bandwidth it can consume, jitter, delay in network etc can anybodytell me, which free downloadble (linux OS based)simulator i can use for this?thankstejas Yahoo! for Good - Make a difference this year. ___ --Bandwidth 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:problem with X100P card
hi all, I wanted to knw whether it is possible to make call to analog phone (outbound call) using X100P card. I have only single piece of card. I m receiving call from analog phone properly,but cant make outbound call. If any one have a dialplan structure pls tell me. Thanks, Tejas Yahoo! for Good - Make a difference this year. ___ --Bandwidth 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:probelem in working of X100P
hi all, I m a newbie to asterisk. I have just installed X1OOP card on my PC.In that PC i have installed asterisk. I have configured three files zaptal.conf,zapta.conf,extensions.conf for cheking X100P card's working ; I did following modifications in Zaptel.conf : fxsks=1 loadzone=us defaultzone=us zapata.conf: language=en context=incoming signalling=fxs_ks usecallerid=yes echocancle=yes transfer=yes echocanclewhenbridged=yes channel = 1 extension.conf : [general] static=yes writeprotect=yes [incoming] exten = s,1,Answer() exten =s,2,Echo() Now in O'reilley Asterisk book i read that whenever i will make call to my PSTN no. with the another analog phone Asterisk will pick the phone and run the Echo() application, so that i can hear my own ech o which will show that card in working in both directions. Now whenever i m making call frm another analog phone, i m just hearing a ring and phone is not picked up by asterisk application. On server i m getting these types of things : - - starting simple switch on 'Zap/1-1' NOTICE[2444] : chan_zap.c : 5624 ss_thread : got event 2 (RING/ANSWERED)... NOTICE[2444] : chan_zap.c : 5624 ss_thread : got event 2 (RING/ANSWERED)... == Starting ZAP/1-1 at analog,s,1 failed so falling back to extension 's' == Starting ZAP/1-1 at analog,s,a still failed so falling back to extension 'default' WARNING[2444] : pbx.c : 1893 ast_pbx_run : Channel 'Zap/1-1' sent into invalid extension 's' in context 'default' but no invalid handeler so where i m getting wrong.pls send me suggestion how to test whether my card in working in both directions or not? Thanks, tejas Yahoo! Photos Ring in the New Year with Photo Calendars. Add photos, events, holidays, whatever.___ --Bandwidth 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: how to make contribution in asterisk
hi all, I am a newbie in asterisk. I am doing my project on implementing "VoIP gateway".I installed asterisk 1.0.7 on Debian. This package was available in Debian-Sarge. For this implementation i choose asterisk.I just bought digitnetworks X100P PSTN card. I have some queries : 1)For this project purpose, Is this card suitable and enough? i m just going to download 3-4 soft IP phones. Since this card has only one FXO port, I think with this i can get PSTN call on my soft IP phones and also i can make call from any soft IP phones to analog phone. whether i m thiking in right direction or not? 2) After installation of this card i will go for simple dialplan structure to confirm how this VoIP gateway works.Since i m new to asterisk, By doing this i will get better idea abt asterisk. Am i doing right? 3) Since i m doing my project work, i hav e to show some implementation which should be my own. I heard about Asterisk Gateway Interface (AGI). So by using AGI what can i develop? since it uses PERL,PYTHON,PHP for development, which shd i go for. As all three are new for me. Which will be fast and easy to learn? 4)I think other option available for me is to do some modifications in the source code? How much time it will require to analyse and understand the asterisk code? I m not so much comfortable with C programming. So whether it will be be suitable to go for this modification? how much time will be reuired to understand the code? (probable time in days). Or i shd go for AGI? 5) Are there some other options available with which i can show that i have worked with asterisk and developed something new, so that i can showit as my project work? suggestions frm all asterisk users are most welcome... thanks Yahoo! for Good - Make a difference this year. ___ --Bandwidth 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: how to forward call within office
hi all, I am newbie to asterisk. I have installed asterisk sever on debian. I have also installed 4 X-Lite phones on four PCs. My all phones and asterisk server is working properly. Now i want to implement call forwariding facility on my server. I want to forward the call made by phone-1 to phone-2 , to another phone i.e. phone-3. Within office premises is it possible to forward calls? if yes, How can i implement this? what configurations files i have to change? I have one othere query. i want to knw, can i use "s" extension for my office pbx? I mean i dont have any PSTN connection to my asterisk server. I m using asterisk just as a internal gateway. so how can i use "s" extension? If i m not wrong "s" extension is used when call comes from PSTN line. Am i right? or m i making a wrong assumption ? Thanks tejas Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping ___ --Bandwidth 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:how to listen voicemail messages
hi all, I have made two voice mail boxes for 2 users on asterisk server (/var/spool/asterisk/voicemail/testmail/inside this 2 boxes for 2 users). i have made following settings in voicemail.conf : [testmail] vipul=,vipul patel, [EMAIL PROTECTED] tejas=,tejas shah,[EMAIL PROTECTED] i have made appprpriate settings in SIP.CONF and EXTENSIONS.CONF. now when any of the user is unavailable voicemail is getting active. It is also allowing to record messages on voicemail box. now my problem is.suppose for one user say TEJAS another user has send voicemail. then how that user TEJAS can listen voicemail message. Is there any command to run on asterisk server. how can i access my voicemail? thanks tejas Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping ___ --Bandwidth 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:Is it possible to install ZAPTEL after installation of Asterisk
Hi all, I have installed and configured asterisk on my debian machine. Right now i m making asterisk server for making connection between 2 X-Lite phones. I m working on different applications (voicemail, call queuing etc). I m plannning to take new hardware (digitnetwork's X100p FXO card) to make VoIP gateway. My question is if: - Is it possible to download and install ZAPTEL after working so much on asterisk? - If yes then for debian how and from where i can download it? - What changes i have to make in system after installation of ZAPTEL? - What other things (driver or other dependencies) i have to download? Suggestions from this group are most expected. Kindly help me to solve this problem. Thanks TEJAS Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Yahoo! Personals___ --Bandwidth 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: Help required on asterisk
hi all, some days back i mailed to group abt my error on asterisk. Now also i m getting same error : cannot find extension context 'from-sip' . I tried DEFAULT context also. but at that time errot remains same: cannot fined extension context 'default'. I think problem is that it is not recognising any context. well i want some suggesion from group. It will be helpful for me if u will send ur suggestions. Actually I want to implement VoIP gateway for my Project work. For that i choose Asterisk. I have certain questions :1) Now i m planning to choose Debian as an operating system. So what do u thinkhows asterisk support for debian. Is asterisk work well on debian? 2) Also if i will use digitnetworks X100P card for PSTN interface would i able toconnect with analog phone. Is only this hardware is sufficient for making gateway? 3) And last but not leastsince i m a newbie to asterisk n Debian how much time(approximately in days) it will take to develop VoIP gateway with asterisk ? sorry i m giving u all a lot of pain, but sir ur suggestions will be very much valuable for me. Thanks, waiting for ur reply Tejas Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Try Yahoo! Personals___ --Bandwidth 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: how to remove asterisk 1.2 from Red Hat 9
hi all,Can anyone tell me how i can remove (uninstall) asterisk 1.2 from Red Hat 9. Also pls tell me which version of asterisk is most suitable for making VoIP gateway on Red Hat 9.Thankstejas Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Yahoo! Personals___ --Bandwidth 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:how to solve error : cannot find extension context 'from-sip'
hi, I am a newbie to asterisk. I am tryining to connect two sip based soft X-Lite phones to an asterisk server. i made following settings in sip.conf: [general] port=5060 bindaddr=0.0.0.0 allow=all context=bogon-calls [2000] type=friend username=2000 secret=tejas host=dynamic context=from-sip [2001] type=friend username=2001 secret=tejas host=dynamic context=from-sip and made following configuration in extension.conf : exten = 2000,1,Dial(SIP/2000,20) exten = 2001,1,Dial(SIP/2001,20) also i made proper settings in both SIP phones. Now the problem is when i am making call from any of phone at srever i am getting an error : "cannot find extension contex 'from-sip'. I analysed result in ethereal also. packets are comming to server. but server is saying : "Proxy authentication required" . N ow i m not getting where is the exact problem. can any one help me for this problem. Response to this problem is most welcome. my email-id is [EMAIL PROTECTED] thanks tejas Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Yahoo! Personals___ --Bandwidth 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] two sip phone communication using asterisk server
hi, I am a newbie to asterisk. I installed a asterisk server to make communication between 2 X-Lite's SIP based phones. I made following configuration in sip.conf :[general]port = 5060 ; Port to bind to (SIP is 5060)bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)allow=all ; Allow all codecscontext = bogon-calls ; Send SIP callers that we don't know about here[2000]type=friend ; This device takes and makes callsusername=2000 ; Username on devicesecret=9overthruster7
;
Password for devicehost=dynamic ; This host is not on the same IP addr every timecontext=from-sip ; Inbound calls from this host go heremailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it[2001] ; Duplicate of 2000, except with different auth datatype=friendusername=2001secret=11bbanzai9host=dynamiccontext=from-sipmailbox=101and following configuration in extension.conf :[general]static=yes ; These two lines prevent the command-line interfacewriteprotect=yes ; from overwriting the config file. Leave them here.[bogon-calls];; Take unknown callers that may have found; our system, and send them to a re-order tone.; The string "_." matches any dialed sequence, so all; calls will result in the Congestion tone application; being called. They'll get bored and hang up eventually.;exten = _.,1,Congestion [from-sip];; If the number dialed by the calling party was "2000", then; Dial the user "2000" via the SIP channel driver. Let the number; ring for 20 seconds, and if no answer, proceed to priority 2.; If the number gives a "busy" result, then jump to priority 102;exten = 2000,1,Dial(SIP/2000,20);; Priority 2 send the caller to voicemail, and gives the "u"navailable; message f
or user
2000, as recorded previously. The only way out; of voicemail in this instance is to hang up, so we have reached; the end of our priority list.;exten = 2000,2,Voicemail(u2000);; If the Dialed number in priority 1 above results in; a "busy" code, then Dial will jump to 101 + (current priority); which in our case will be 101+1=102. This +101 jump is built; into Asterisk and does not need to be defined.;exten = 2000,102,Voicemail(b2000)exten = 2000,103,Hangup;; Now, what if the number dialed was "2001"?;exten = 2001,1,Dial(SIP/2001,20)exten = 2001,2,Voicemail(u2001)exten = 2001,102,Voicemail(b2001)exten = 2001,103,Hangup;; Define a way so that users can dial a number to reach; voicemail. Call the VoicemailMain application with the; number of the caller already passed as a variable, so; all the user needs to do is type in the
password.;exten = 2999,1,VoicemailMain(${CALLERIDNUM})now my problem is when i m starting asterisk server both Sip phones are showing registration. when i make call from any of PC following error occurs on the screen of asterisk server :pbx.c:1731: can not find extension context 'from-sip'when i close asterisk server communication is taking place beween both phones. now i m stuck with this error. can anybody give me guidance on how to solve this problem?thankstejas
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