Re: [asterisk-users] Meetme
FaberK wrote: On Mon, Jul 7, 2008 at 2:48 PM, Philipp Ott [EMAIL PROTECTED] wrote: Hi! FaberK schrieb: My question is, is it possible to cut off that request topress one? I think you want to get rid of the number-pressing. The only option to omit this seems to be option E - select an empty pinless conference. Well we need the PIN feature so I have to find another solution. I'm looking into the code and it seems to me, that this request is part of the app_voicemail: Why don't you do pin authentication prior using say the Authenticate application? Hi Matt, the problem is inside the code. The part interested comes from the app_voicemail.c. I'm trying to remove the code interested, but until now, no luck. Thanks -- .:FaberK:. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme
Hi folks, we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:press one to accept the recording... My question is, is it possible to cut off that request topress one? Thanks to all -- .:FaberK:. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme
Hi, but if I edit the sound file, remain that I have to press the 1 button to go ahead. Thanks to all. On Mon, Jul 7, 2008 at 1:57 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: FaberK schrieb: we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:press one to accept the recording... My question is, is it possible to cut off that request topress one? Audacity. Edit the sound file. Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme
On Mon, Jul 7, 2008 at 2:48 PM, Philipp Ott [EMAIL PROTECTED] wrote:
Hi!
FaberK schrieb:
My question is, is it possible to cut off that request topress one?
I think you want to get rid of the number-pressing. The only option to
omit this seems to be option E - select an empty pinless conference.
Well we need the PIN feature so I have to find another solution.
I'm looking into the code and it seems to me, that this request is
part of the app_voicemail:
--
default:
/* If the caller is an ouside caller, and the
review option is enabled,
allow them to review the message, but let
the owner of the box review
their OGM's */
if (outsidecaller !ast_test_flag(vmu, VM_REVIEW))
return cmd;
if (message_exists) { /* I THINK IS THIS */
cmd = ast_play_and_wait(chan, vm-review);
}
else {
cmd = ast_play_and_wait(chan, vm-torerecord);
if (!cmd)
cmd = ast_waitfordigit(chan, 600);
}
if (!cmd outsidecaller
ast_test_flag(vmu, VM_OPERATOR)) {
cmd = ast_play_and_wait(chan, vm-reachoper);
if (!cmd)
cmd = ast_waitfordigit(chan, 600);
}
#if 0
if (!cmd)
cmd = ast_play_and_wait(chan, vm-tocancelmsg);
#endif
if (!cmd)
cmd = ast_waitfordigit(chan, 6000);
if (!cmd) {
attempts++;
}
if (attempts max_attempts) {
cmd = 't';
}
}
}
if (outsidecaller)
ast_play_and_wait(chan, vm-goodbye);
if (cmd == 't')
cmd = 0;
return cmd;
}
--
Thanks
--
.:FaberK:.
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[asterisk-users] Receptionist SNOM-360
Hi to all, I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls) and 15 SIP extensions. The receptionist has a SNOM-360. How many SIP accounts would you configure on that phone? Only one would be enough? One SIP account, has a limit on concurrent calls? I saw that the SNOM-360 can handle up to eleven SIP accounts. Thanks -- .:FaberK:. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and radius
Hi folks, I'm trying to install asterisk with radius cdr support. I got freeradius up and running, so following radius instructions inside asterisk source package, I've installed radiusclient-ng and relative headers. But when I start configure(asterisk 1.4.18.1) I got: checking for rc_read_config in -lradiusclient-ng... no If I type: ./configure --with-radius=/usr/share/radiusclient-ng the answer is the same, more: checking for rc_read_config in -lradiusclient-ng... no configure: *** configure: *** The Radius Client installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** without explicitly specifying --with-radius But the installation of radiusclient, didn't give me any problems. Any hints? Thanks -- .:FaberK:. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-gui
Hi Steve, you are totally right, but my question is because a saw that gui into SVN and not yet released, but at the same time used into AsteriskNOW. Was just a question... ;o) Thanks 2007/10/12, Steve Totaro [EMAIL PROTECTED]: FaberK wrote: Hi to all, I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? Thanks to all -- .:FaberK:. I may be totally wrong but at Astricon, during the What's New at Digium (SwitchVox purchase) I asked the question of what would happen to AsteriskNow to one of the Adtran/Digium guys. There was not a real direct answer, I will try to quote as best I can from memory. He simply said It will remain opensource. I take that to mean that they will not be developing it anymore and it is up to the community to further the project. Why would Digium continue to develop a GUI for free that would compete with SwitchVox (or whatever they change the name to). Maybe I am wrong. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-gui
Hi to all, I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? Thanks to all -- .:FaberK:. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggestion for installation
Hi to all, till now I've used SER as sip registrar and Asterisk as its gateway(PSTN) and for billing. Now, I've received a request to setup a solution, for 5000 + o - users(this is what they expext to have). I was thinking to use only Asterisk with Freeradius, no SER. Any suggestion/experience? Thanks to all -- .:FaberK:. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] checking credit by phone
Hi to all, I've tried to use the ASTCC credit check a long time ago and it worked pefectly, but now... no more Any suggestions for some new software? Thanks to all. ___ --Bandwidth 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] CISCO 2600 - VWIC 1MFT-E1
http://pastebin.ca/271763 Hi to all, To Fran: As I understand your configuration , dial-peer voice 697617664 voip, only forward the pattern 697617664( destination-pattern 697617664) to XXX.XXX.XXX. 115:5060 ( session target ipv4:XXX.XXX.XXX.115:5060) that I think is your Asterisk box. you are right, XXX.XXX.XXX.115:5060 is my * box where I've created a friend called 697617664 An incoming call in your E1 must much a destination pattern, your only destination pattern is 697617664. Usually an E1 has several DID associated it in a consecutive range, 91 5344XXX for example. here too, you are right, but I'm trying to receive at leat 1 call to 697617664, then for all the others will be not a problem. But first i need to let it works...!!! otherwise, for outgoing calls you must configure a pots dial peer ,you can put a randon name to the dial peer. You can configure asterisk , without user registration with the sip.confinsecure option when I copied dial-peer voice 10 pots destination-pattern 0T should be .T it tells cisco 26xx router what patterns can be reached throught E1 I´ll take a look into the cisco web site for sip user authentication, I have a configuration done, but with FXS interfaces and worsk fine. For outgoing calls, at this moment I'm not interested. On the new configuration, I've also changed the codecs, leaving the g711 only. Unfortunately always the same: calling my number, the call reach the 2600(infact I hear the tone), but is not forwarded to the sip-server. To Pavel: thanks for your suggestion regarding MGCP, but the fact is that I got all sip, and never worked with mgcp. Thanks to all Best Regards F. ___ --Bandwidth 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] CISCO 2600 - VWIC 1MFT-E1
Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .: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] CISCO 2600 - VWIC 1MFT-E1
http://pastebin.ca/270840 This is the newone with some changements. Unfortunately, always the same problem. Fran, if I add the dial-peer voice 10 pots, I receive the advise that the number does not exist. Also, I do not find the way to add authentication username asterisk-uername password XX. The story continues... Thanks F. ___ --Bandwidth 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] cisco 2600
Hi,does anybody used cisco 2600 as * gateway with E1?Thanks-- .: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 configure NOKIA N70 with Asterisk?
Hi,thanks Jean-Yves, but I've already found that page (googling), but I asked because following those instruction I couldn't find the SIP settings.Maybe are not present on my N70?Well I'll investigate*## on my mobile says: V 2.0539.1.219-10-05RM-84Any hints?Thanks2006/8/1, Jean-Yves Avenard [EMAIL PROTECTED]: HiOn 8/1/06, FaberK [EMAIL PROTECTED] wrote: Hi folks, I got an N70. Any lynks for the voip/sip configuration? Thanks .:FaberK:.they aren't hard to find !this one works for me:http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html One note of warning :the Nokia will not work if behing NAT ... I've tried everything butI've never managed to get it to work unless the Nokia had a public IPaddress or was on the same subnet as the asterisk server. Be interested to know if you can find a way around thisJY___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .: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] Re: How to configure NOKIA N70 with Asterisk?
Hi,the problem is that I have not the sip choice into my N70 menu.Today I've made an update of the system, now I have:V 5..0609.2.0.1but still no sip.I think is because my mobile has been customized by my telephone company, H3G. I'll investigate.Thanks01 Aug 2006 20:54:53 +0200, Benny Amorsen [EMAIL PROTECTED]: FK == FaberK[EMAIL PROTECTED] writes:FK Hi, thanks Jean-Yves, but I've already found that page (googling),FK but I asked because following those instruction I couldn't find FK the SIP settings. Maybe are not present on my N70? Well I'llFK investigate *## on my mobile says: V 2.0539.1.2 19-10-05FK RM-84 Any hints?Tools-settings-connections-sip. So far the only problems I've had are the ones which are already wellknown:No NAT traversalSwitching between making calls on WLAN and GSM/UMTS isn't automatic,and it's not just an easy button push either /Benny___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- .: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 configure NOKIA N70 with Asterisk?
Hi folks,I got an N70.Any lynks for the voip/sip configuration?Thanks.: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] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)]
Hi,it cannot find mysql.sock. Is it your mysql running?2006/5/8, ali asma [EMAIL PROTECTED]: Hello, I have an error when installing AMP, when I do ./install_amp --debug, it show me : Connecting to database..FAILED [DEBUG] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)] ** mysql://user:[EMAIL PROTECTED]/asteriskamp Try running ./install_amp --username=user --password=pass (using your own user and pass) [FATAL] Cannot connect to database I tried with ./install_amp --username=user --password=pass, but the same error persist Coud some one help me Thunks, Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Cliquez ici. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- .: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] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)]
so, check if mysql.sock is in /tmp/mysql.sock insted of /var/lib/mysql/mysql.sock.If so, change the value into AMP.2006/5/8, ali asma [EMAIL PROTECTED] :yes, I have mysql 4.1.18 runs Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Cliquez ici. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- .: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] answer delay
Hi Folks,using this:exten = x,1,Playback(audio,noanswer)exten = x,2,Answerexten = x,3,BackGround(out)exten = x,103,HangupI'm not billed and remain connected, but the file 'audio' is not played...well I do not ear it. But after, it pass correctly to answer and I can ear the 'out' audio file.Any idea/suggestion???Thanks!2006/3/21, FaberK [EMAIL PROTECTED] :Hi,I've tryed it using my mobile and I've been charged. Maybe, my mobile operator(Vodafone) does not support it?Thanks again.p.s.: hi John, I love to learn(books, google, lists, ecc...) and cooperation and I can say that everytime I learn something new. :o) 2006/3/21, CC Jay [EMAIL PROTECTED]: Not sure about 1.2.4 but with 1.0.9, you'll need to add noanswer to playback since playback will try to answer the line, i.e.,exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this, noanswer) exten = 5551234,n,Answeretc. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. -- .: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] answer delay
Hi Peter,but if I Answer, I'll be billed, isn't it?What I need is to play an announce of the service cost, so that if the guest do not want to go ahead for the cost, can hungup without pay.I'll try your solution. Thanks.2006/4/25, Peter J Dean [EMAIL PROTECTED]: You need to have an established and open channel before the audio canbe played.exten = x,1,Answerexten = x,n,Playback(audio,noanswer)exten = x,n,BackGround(out)exten = x,n,Hangup On 24/04/2006, at 10:25 PM, FaberK wrote: Hi Folks, using this: exten = x,1,Playback(audio,noanswer) exten = x,2,Answer exten = x,3,BackGround(out) exten = x,103,Hangup I'm not billed and remain connected, but the file 'audio' is not played...well I do not ear it. But after, it pass correctly to answer and I can ear the 'out' audio file. Any idea/suggestion??? Thanks! 2006/3/21, FaberK [EMAIL PROTECTED] : Hi, I've tryed it using my mobile and I've been charged. Maybe, my mobile operator(Vodafone) does not support it? Thanks again. p.s.: hi John, I love to learn(books, google, lists, ecc...) and cooperation and I can say that everytime I learn something new. :o) 2006/3/21, CC Jay [EMAIL PROTECTED]: Not sure about 1.2.4 but with 1.0.9, you'll need to add noanswer to playback since playback will try to answer the line, i.e., exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this, noanswer) exten = 5551234,n,Answer etc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. -- .: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___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .: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] answer delay
I've just tryed and I've been billed.No other ideas???I'm still searching.2006/4/25, FaberK [EMAIL PROTECTED]: Hi Peter,but if I Answer, I'll be billed, isn't it?What I need is to play an announce of the service cost, so that if the guest do not want to go ahead for the cost, can hungup without pay. I'll try your solution. Thanks.2006/4/25, Peter J Dean [EMAIL PROTECTED] : You need to have an established and open channel before the audio canbe played.exten = x,1,Answerexten = x,n,Playback(audio,noanswer)exten = x,n,BackGround(out) exten = x,n,Hangup On 24/04/2006, at 10:25 PM, FaberK wrote: Hi Folks, using this: exten = x,1,Playback(audio,noanswer) exten = x,2,Answer exten = x,3,BackGround(out) exten = x,103,Hangup I'm not billed and remain connected, but the file 'audio' is not played...well I do not ear it. But after, it pass correctly to answer and I can ear the 'out' audio file. Any idea/suggestion??? Thanks! 2006/3/21, FaberK [EMAIL PROTECTED] : Hi, I've tryed it using my mobile and I've been charged. Maybe, my mobile operator(Vodafone) does not support it? Thanks again. p.s.: hi John, I love to learn(books, google, lists, ecc...) and cooperation and I can say that everytime I learn something new. :o) 2006/3/21, CC Jay [EMAIL PROTECTED]: Not sure about 1.2.4 but with 1.0.9, you'll need to add noanswer to playback since playback will try to answer the line, i.e., exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this, noanswer) exten = 5551234,n,Answer etc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. -- .: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___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. -- .: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] Playback(something,noanswer) on Zap?
Hi Dmitry,may I ask you if is possible to see the pub2ext.agi code?I'm looking for a solution like your, with no luck since long time(you can see from ml-archive).Thanks a lot!
2006/4/24, Dmitry Ivanov [EMAIL PROTECTED]:
On Thursday 20 April 2006 19:22, Kevin P. Fleming wrote: A better solution is to set the PRI hangup cause before dropping the incoming call; if you set the hangup cause to 'number not assigned' then your telco's switch will play its normal intercept message to
the caller.Thank you! This works!context from-e1 {_X. = {AGI(pub2ext.agi);PRI_CAUSE=1;Hangup();};};Now caller hears voice from his/her telco (not from my telco) saying
that number is not available. This is even better.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list
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Re: [Asterisk-Users] answer delay
I know that, but as soon as I answer, guest is billed.Sorry to trouble you.Thanks2006/4/25, Peter J Dean [EMAIL PROTECTED] :You are stilling going to need to answer the call before you can playany message or music or other. Here is a start for you.So lets look at it logically;- Advise the caller of the charges and conditions.- The caller must has a option to acknowledge they understand andaccept the charges - (Optionally) The caller has can repeat the charges and conditionsmessage.This no leads us towards a simple IVR system. Below is an example,and you will need to clean it up.You can use the ResetCDR() command to reset the time used, and continue from there.So, extending on the below (untested, but you get the general idea);[context_askcustomerbeforechargingthem]exten = x,1,Answerexten = x,n,Playback(audio) exten = x,n,Playback(press_one_to_accept_the_charges_and_conditions)exten = x,n,Playback(just_simply_hangup_if_you_do_not_wish_to_proceed)exten = x,n,WaitExten(5) exten = x,n,Playback(out)exten = x,n,Hangupexten = _1,1,Playback(thankyou_for_selecting_our_service_or_whatever)exten = _1,n,ResetCDR()exten = _1,n,Goto or do whatever next from here On 25/04/2006, at 8:30 AM, FaberK wrote: Hi Peter, but if I Answer, I'll be billed, isn't it? What I need is to play an announce of the service cost, so that if the guest do not want to go ahead for the cost, can hungup without pay. I'll try your solution. Thanks. 2006/4/25, Peter J Dean [EMAIL PROTECTED]: You need to have an established and open channel before the audio can be played. exten = x,1,Answer exten = x,n,Playback(audio,noanswer) exten = x,n,BackGround(out) exten = x,n,Hangup On 24/04/2006, at 10:25 PM, FaberK wrote: Hi Folks, using this: exten = x,1,Playback(audio,noanswer) exten = x,2,Answer exten = x,3,BackGround(out) exten = x,103,Hangup I'm not billed and remain connected, but the file 'audio' is not played...well I do not ear it. But after, it pass correctly to answer and I can ear the 'out' audio file. Any idea/suggestion??? Thanks! 2006/3/21, FaberK [EMAIL PROTECTED] : Hi, I've tryed it using my mobile and I've been charged. Maybe, my mobile operator(Vodafone) does not support it? Thanks again. p.s .: hi John, I love to learn(books, google, lists, ecc...) and cooperation and I can say that everytime I learn something new. :o) 2006/3/21, CC Jay [EMAIL PROTECTED]: Not sure about 1.2.4 but with 1.0.9, you'll need to add noanswer to playback since playback will try to answer the line, i.e., exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this, noanswer) exten = 5551234,n,Answer etc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. -- .: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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .: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 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- .: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
[Asterisk-Users] H323 problems
Hi all,I'm experiencing problems using H323, with 60 calls, * crashes...!My server is a PIV with 2 Gb, on CentOS 3.6, Asterisk 1.2.5, openh323_v1_17_1, asterisk-addons-1.2.2.Errors messages, are:chan_h323.c:1483 cleanup_connection: Avoiding H.323 destory deadlock on ip$xxx.xxx.xxx.xxx/28508channel.c: Avoided initial deadlock for '0x9e291d0', 10 retries!res_features.c: Bridge failed on channels Zap/105-1 and H323/xxx.xxx.xxx.xxx-821 Also, normally I use IAX2 and each * processes use 12Mb, now with H323 it grows to 26Mb.So, could be an insufficient hardware resources problem?Any suggestions?Thanks in advance-- .: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
[Asterisk-Users] compiling chan_h323
Hi to all,I'm trying to compiling H323 under * 1.2.5, but no luck.I got a loto of problems.I've installes Pwlib 1.9 as required into the source but no luck with Openh323 1.7.1.I've got gcc 3.2.3I've googled with no results. Any hints?Thanks-- .: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] answer delay
Hi,I've tryed it using my mobile and I've been charged.Maybe, my mobile operator(Vodafone) does not support it?Thanks again.p.s.: hi John, I love to learn(books, google, lists, ecc...) and cooperation and I can say that everytime I learn something new. :o) 2006/3/21, CC Jay [EMAIL PROTECTED]: Not sure about 1.2.4 but with 1.0.9, you'll need to add noanswer to playback since playback will try to answer the line, i.e.,exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this, noanswer) exten = 5551234,n,Answeretc. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .: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
[Asterisk-Users] answer delay
Hi guys, maybe youìve got the answer...! When a caller(not internal, but from PSTN) call *, I need to let him hear a message, before * answer and the bill start running. If is not clear, just let me know. caller-telco(telco bill to the caller as soon as * answer)-asterisk Thanks in advance. -- .: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] answer delay
Thanks a lot!!! Is exactly what I need to do. Send a message, before answer. Thanks to all! F.2006/3/20, Andrew Kohlsmith [EMAIL PROTECTED]: On Monday 20 March 2006 11:46, John Daragon wrote: Alas, most (if not all) telcos object to you transmitting voice over their circuits before they've started to charge you for the call.Incorrect.I do this all the time with a PRI.You can't do this with POTS. Simply don't Answer() until you're ready to bill.You can send audio but youcannot hear them until you answer the call.exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this)exten = 5551234,n,Answer exten = 5551234,n,Playback(now-the-meter-is-running)exten = 5551234,n,Record(and-we-can-hear-you.gsm)-A.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .: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] Problem with 827-4v and asterisk as a pstn GW
Hi Simone,I'm trying to use a 827-4V with SER+Asterisk.My problems are that calling out, from 827, is fine, but on the other way do not works.The PSTN caller ear no ringing and when the 827 answers, there is no outbound voice. Do you reach some results?Best Regards.2005/11/14, Simone Ricci [EMAIL PROTECTED]: Hi,I've a problem with a cisco 827-4v and asterisk (1.0.9) acting assip-to-pstn GW. The issue is that when a call comes in from the pstn,asterisk correctly contacts the router, which in turns send a 183 Session progress. Obviously, asterisk thinks that the telephone is notringing (because it expects a 180 Ringing) and we have no ringback onthe pstn side. Putting a ringing() in the dialplan is not an option. Anyone has suggestions?Cheers,Simone.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- .: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
[Asterisk-Users] uniqueid
Hi folks,I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls.I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong.In any case, somebody got same problem?Any suggestions?Thanks to all.-- .: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
[Asterisk-Users] Re: uniqueid
News!I've just replaced the cdr_addon_mysql.so with the old one, and it start to work properly!So I can suppose a bug into that module.I'll check the old cdr_addon_mysql.c and see difference of code, if any. Thanks.2006/3/5, FaberK [EMAIL PROTECTED]: Hi folks,I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls.I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong.In any case, somebody got same problem?Any suggestions?Thanks to all.-- .:FaberK:. -- .: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
[Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)
Hi guys,I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk.Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not.Then I saw that message appear on the Asterisk CLI, during the incoming call:NOTICE[3575]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: XXX.XXX.XXX.XXXNow I've checked into the router, and the VAD was already unset.Using normal IP-telephones, everything is perfect.Does anyone, got an idea or already got problems with that router?Thanks to all -- .: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
[Asterisk-Users] Dial and Congestion
Hi folks,very stupid question, how do I setup a Dial with multiple Zap choises?I've setup this, but maybe is wrong:exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(Zap/g2/${EXTEN})exten = _7653.,3,Dial(Zap/g4/${EXTEN})exten = _7653.,101,Congestionwhat I want to do is that as soon as lines are not available on g2, all others outgoing calls must go to g4.
Should be very easy, but it do not work.If the configuration is correct, then I must check the PRI.Thanks again-- .:FaberK:.
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Re: [Asterisk-Users] Dial and Congestion
Hi Massimiliano,thanks for your prompt reply. Unfortunately that solution seems to not work, but I not sure is your code, I'm starting to believe that this PRI, got some problems.My system is in production, so I have to wait for more tests.
In the meantime, I thank you so much.FaberK aka Fabrizio2006/2/22, Massimiliano Stucchi [EMAIL PROTECTED]:
On 220206, 12:54, FaberK wrote: Hi folks, very stupid question, how do I setup a Dial with multiple Zap choises?
I've setup this, but maybe is wrong: exten = _7653.,1,SetCallerID(${CALLERID(number)}) exten = _7653.,2,Dial(Zap/g2/${EXTEN}) exten = _7653.,3,Dial(Zap/g4/${EXTEN})
exten = _7653.,101,Congestion what I want to do is that as soon as lines are not available on g2, all others outgoing calls must go to g4.
Should be very easy, but it do not work. If the configuration is correct, then I must check the PRI.It should be like this:exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,ChanIsAvail(Zap/g2)exten = _7653.,3,Dial(Zap/g2/${EXTEN})exten = _7653.,103,Dial(Zap/g4/${EXTEN})exten = _7653.,204,playtones(congestion)Cheers--Massimiliano Stucchi
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[Asterisk-Users] codecs choice
Hi all,I have an * box dual Xeon, 4Gb ram, 2 A104.Normally I use gsm codec, but to allow using faxes, I let some users to use g711 as default codec.My question is:Is it possible to detect what a certain call is? So if is a phone call I'll use gsm, if is a fax I'll use g711.Thanks to all-- .: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
[Asterisk-Users] Asterisk + Ericsson PBX
Hi all,I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX.I need to use Asterisk as E1 line for the Ericsson PBX.How do I have to connect them?I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions?Thanks-- .: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] Asterisk + Ericsson PBX
Thanks!Now the E1 is up, but still problems.What I'm trying to do, is to let calls arrive to Asterisk from the net, and using the Sangoma pass them to the PBX.Is this possible?Thanks again. 2006/1/25, Mimmus [EMAIL PROTECTED]: You need a crossed E1 cable, that is different from an Ethernet E1 cable, with the following pin-outs: 1 - 42 - 54 - 15 - 2 Thantry without CRC in zaptel.conf. My Alcatel PBX didn't like it. Mimmus From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FaberKSent: Wednesday, January 25, 2006 1:26 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Asterisk + Ericsson PBX Hi all,I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX.I need to use Asterisk as E1 line for the Ericsson PBX.How do I have to connect them?I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions?Thanks-- .:FaberK:. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .: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] Merry Xmas to everybody!
Merry Christmas and Happy New Year to everybody Buon Natale e Felice Anno Nuovo >From Roma Italy FaberK 2005/12/23, Mark Phillips [EMAIL PROTECTED]: What's wrong with us that celebrate Kwanza?Mark, G7LTT/KC2ENIRandolph, NJhttp://www.g7ltt.comDmitry Ivanov wrote: On Friday 23 December 2005 10:22, Mauro Zanin wrote: Hi everybody,no issues this time. Only stopped to say: Merry Christmas and HappyNew Year. Yes, Merry Christmas, Happy New Year and Hanukkah :) Just received nice postcard from Digium :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- .: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
[Asterisk-Users] Re: Romania/Rumania setup
I try posting again... unfortunately, this time I'm not able to solve by myself!!! I've checked zttool and I've got no alarm just OK. No other news. Please HELP! Thanks!-- .: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
[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
[Asterisk-Users] Zap Channels
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
Thanks!
--
.:FaberK:.
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Re: [Asterisk-Users] Zap Channels
HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
group = 2
channel = 32-46,48-62
group = 3
channel = 63-77,79-93
transfer=yes
threewaycalling=yes
callwaitingcallerid=yes
callwaiting=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echotraining=yes
zaptel.conf
defaultzone=it
loadzone=it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,1,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,1,0,ccs,hdb3,crc4
bchan=94-109,111-124
dchan=110
--
2005/12/2, Xisco Mateu [EMAIL PROTECTED]:
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
Thanks!
--
.:FaberK:.
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--
.:FaberK:.
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Re: [Asterisk-Users] Zap Channels
PRITRUNK1 is defined into the extensions.conf globals:
--
[globals]
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
PRITRUNK3=Zap/g3
--
Well I know what's happening, from my asterisk CDRs, and also from the PRI-CDRs.
We use a Teles.
2005/12/2, Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, Dec 02, 2005 at 11:00:34AM +0100, FaberK wrote:
HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
group = 2
channel = 32-46,48-62
group = 3
channel = 63-77,79-93
transfer=yes
threewaycalling=yes
callwaitingcallerid=yes
callwaiting=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echotraining=yes
zaptel.conf
defaultzone=it
loadzone=it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,1,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,1,0,ccs,hdb3,crc4
bchan=94-109,111-124
dchan=110
--
2005/12/2, Xisco Mateu [EMAIL PROTECTED]:
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
What is PRITRUNK1? where is it defined?
How do you know something is wrong? Could you please paste the trace
from the logs/cli when verbosity is set to a high enough value? (e.g: 3)
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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.:FaberK:.
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[Asterisk-Users] stopped sounds
Hi guys, my scenario is this: PRI-Asterisk-Asterisk2-SER Now, until 2 days ago, everything was fine, I could call and receive, from yesterday a can call and receive, but no audio in both directions. In the errors I see the message in object stopped sounds I've tryied to modify the codecs, I've checked if the RTPPROXY was up, but no changes. Actual codecs are: ulaw alaw Here I copied deteiles of a call that I made incoming from PSTN to Asterisk and the debug are relatives to the both Asterisk machines with iax2debug on. Thanks for any suggestions. --- .:FaberK:. -- FIRST ASTERISK DEBUG -- -- Executing Dial(Zap/28-1, IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Accepting call from '06Y' to '06' on channel 0/28, span 1 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00014ms SCall: 1 DCall: 0 [192.168.1.186:4569] VERSION : 2 CALLED NUMBER : 06 CODEC_PREFS : (ulaw|alaw) CALLING NUMBER : 06 CALLING PRESNTN : 3 CALLING TYPEOFN : 33 CALLING TRANSIT : 0 LANGUAGE: it CALLED CONTEXT : incoming USERNAME: remote FORMAT : 4 CAPABILITY : 63500 ADSICPE : 2 DATE TIME : 2005-11-20 15:15:38 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 1 DCall: 1 [192.168.1.186:4569] AUTHMETHODS : 3 CHALLENGE : 241965875 USERNAME: remote Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00016ms SCall: 1 DCall: 1 [192.168.1.186:4569] MD5 RESULT : 9a4d5b94553eac72ee65abc9241132fd Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 6ms SCall: 1 DCall: 1 [192.168.1.186:4569] FORMAT : 4 -- Call accepted by 192.168.1.186 (format ulaw) -- Format for call is ulaw Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 6ms SCall: 1 DCall: 1 [192.168.1.186:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 00040ms SCall: 1 DCall: 1 [192.168.1.186:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00040ms SCall: 1 DCall: 1 [192.168.1.186:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: RINGING Timestamp: 00547ms SCall: 1 DCall: 1 [192.168.1.186:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00547ms SCall: 1 DCall: 1 [192.168.1.186:4569] -- IAX2/192.168.1.186:4569-1 is ringing Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: CONTROL Subclass: (255?) Timestamp: 02340ms SCall: 1 DCall: 1 [192.168.1.186:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02340ms SCall: 1 DCall: 1 [192.168.1.186:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass: ANSWER Timestamp: 02343ms SCall: 1 DCall: 1 [192.168.1.186:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 02343ms SCall: 1 DCall: 1 [192.168.1.186:4569] -- IAX2/192.168.1.186:4569-1 stopped sounds -- IAX2/192.168.1.186:4569-1 answered Zap/28-1 -- Channel 0/28, span 1 got hangup request Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: HANGUP Timestamp: 08500ms SCall: 1 DCall: 1 [192.168.1.186:4569] CAUSE CODE : 16 -- Hungup 'IAX2/192.168.1.186:4569-1' == Spawn extension (default, 020202, 1) exited non-zero on 'Zap/28-1' Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 08500ms SCall: 1 DCall: 1 [192.168.1.186:4569] -- Hungup 'Zap/28-1' -- FIRST ASTERISK DEBUG END-- -- SECOND ASTERISK DEBUG Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00014ms SCall: 1 DCall: 0 [192.168.1.188:4569] VERSION : 2 CALLED NUMBER : 06 CODEC_PREFS : (ulaw|alaw) CALLING NUMBER : 06 CALLING PRESNTN : 3 CALLING TYPEOFN : 33 CALLING TRANSIT : 0 LANGUAGE: it CALLED CONTEXT : incoming USERNAME: remote FORMAT : 4 CAPABILITY : 63500 ADSICPE : 2 DATE TIME : 2005-11-20 15:15:38 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 1 DCall: 1 [192.168.1.188:4569] AUTHMETHODS : 3 CHALLENGE : 241965875 USERNAME: remote Rx-Frame Retry
[Asterisk-Users] Re: stopped sounds
more... during the call, the command iax2 show channels on the first Asterisk returns this: Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format IAX2/recserver2-1 192.168.1.186recserver 1/1 6/2 0ms -0001ms ms unknow Any ideas? Is very urgent!!! Thanks to all -- 2005/11/20, FaberK [EMAIL PROTECTED]: Hi guys, my scenario is this: PRI-Asterisk-Asterisk2-SER Now, until 2 days ago, everything was fine, I could call and receive, from yesterday a can call and receive, but no audio in both directions. In the errors I see the message in object stopped sounds I've tryied to modify the codecs, I've checked if the RTPPROXY was up, but no changes. Actual codecs are: ulaw alaw Here I copied deteiles of a call that I made incoming from PSTN to Asterisk and the debug are relatives to the both Asterisk machines with iax2debug on. Thanks for any suggestions. --- .:FaberK:. -- FIRST ASTERISK DEBUG -- -- Executing Dial(Zap/28-1, IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Accepting call from '06Y' to '06' on channel 0/28, span 1 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00014ms SCall: 1 DCall: 0 [192.168.1.186:4569] VERSION : 2 CALLED NUMBER : 06 CODEC_PREFS : (ulaw|alaw) CALLING NUMBER : 06 CALLING PRESNTN : 3 CALLING TYPEOFN : 33 CALLING TRANSIT : 0 LANGUAGE: it CALLED CONTEXT : incoming USERNAME: remote FORMAT : 4 CAPABILITY : 63500 ADSICPE : 2 DATE TIME : 2005-11-20 15:15:38 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 1 DCall: 1 [192.168.1.186:4569] AUTHMETHODS : 3 CHALLENGE : 241965875 USERNAME: remote Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00016ms SCall: 1 DCall: 1 [192.168.1.186:4569] MD5 RESULT : 9a4d5b94553eac72ee65abc9241132fd Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 6ms SCall: 1 DCall: 1 [192.168.1.186:4569] FORMAT : 4 -- Call accepted by 192.168.1.186 (format ulaw) -- Format for call is ulaw Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 6ms SCall: 1 DCall: 1 [192.168.1.186:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 00040ms SCall: 1 DCall: 1 [192.168.1.186:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00040ms SCall: 1 DCall: 1 [192.168.1.186:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: RINGING Timestamp: 00547ms SCall: 1 DCall: 1 [192.168.1.186:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00547ms SCall: 1 DCall: 1 [192.168.1.186:4569] -- IAX2/192.168.1.186:4569-1 is ringing Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: CONTROL Subclass: (255?) Timestamp: 02340ms SCall: 1 DCall: 1 [192.168.1.186:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02340ms SCall: 1 DCall: 1 [192.168.1.186:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass: ANSWER Timestamp: 02343ms SCall: 1 DCall: 1 [192.168.1.186:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 02343ms SCall: 1 DCall: 1 [192.168.1.186:4569] -- IAX2/192.168.1.186:4569-1 stopped sounds -- IAX2/192.168.1.186:4569-1 answered Zap/28-1 -- Channel 0/28, span 1 got hangup request Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: HANGUP Timestamp: 08500ms SCall: 1 DCall: 1 [192.168.1.186:4569] CAUSE CODE : 16 -- Hungup 'IAX2/192.168.1.186:4569-1' == Spawn extension (default, 020202, 1) exited non-zero on 'Zap/28-1' Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 08500ms SCall: 1 DCall: 1 [192.168.1.186:4569] -- Hungup 'Zap/28-1' -- FIRST ASTERISK DEBUG END-- -- SECOND ASTERISK DEBUG Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00014ms SCall: 1 DCall: 0 [192.168.1.188:4569] VERSION : 2 CALLED NUMBER : 06 CODEC_PREFS : (ulaw|alaw) CALLING
[Asterisk-Users] PRI to SIP
Hi guys,
this is the scenario:
PRI -Asterisk-SER
If I call from a Sip(SER) user everything is good, I can call
anywhere, but if I try to call from outside(PRI) everything is
wrong!!!
This is the CLI for an incoming call:
--
ast*CLI
-- Executing SetCallerID(Zap/14-1, outside) in new stack
-- Executing Set(Zap/14-1, CALLERID=outside) in new stack
-- Executing Dial(Zap/14-1,
SIP/[EMAIL PROTECTED]:5060|30|r) in new stack
-- Accepting call from 'outside' to 'mynumber201' on channel 0/14, span 1
-- Called [EMAIL PROTECTED]:5060
Nov 14 17:15:47 NOTICE[8459]: chan_sip.c:9492 handle_response_invite:
Failed to authenticate on INVITE to 'Unknown
sip:[EMAIL PROTECTED];tag=as6261e060'
-- SIP/sip.mydomain.com:5060-5eda is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(Zap/14-1, ) in new stack
-- Channel 0/14, span 1 got hangup request
== Spawn extension (default, 020201, 4) exited non-zero on 'Zap/14-1'
-- Hungup 'Zap/14-1'
ast*CLI
--
my extensions:
--
[general]
static=yes
writeprotect=no
[globals]
;TRUNK=Zap/g2
;TRUNKMSD=1
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
MYNUM=1234567
[default]
exten = _1234567XXX,1,SetCallerID(${CALLERID})
exten = _1234567XXX,2,Set(CALLERID=${CALLERID})
exten = _1234567XXX,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5060,30,r)
exten = _1234567XXX,103,Hangup
--
Where I'm wrong?
What's missing?
Thanks!
--
.:FaberK:.
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Re: [Asterisk-Users] PRI to SIP
Hi Jens,
this is my sip.conf
---
[general]
context=default
fromdomain=192.168.1.188
port=5060
bindaddr=0.0.0.0
localnet = 192.168.1.0/255.255.255.0
srvlookup=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
language=it
register = 1:[EMAIL PROTECTED]:5060/s
[1]
type=peer
username=1
fromuser=1
secret=1
Callerid=1
context=default
port=5060
dtmfmode=rfc2833
host=sip.mydomain.com
fromdomain=sip.mydomain.com
insecure=very
canreinvite = no
disallow=all
allow=alaw
allow=ulaw
---
and my extensions.conf
---
[general]
static=yes
writeprotect=no
[globals]
;TRUNK=Zap/g2
;TRUNKMSD=1
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
MYNUM=0123456
[default]
exten = _0123456XXX,1,Wait(0.75)
exten = _0123456XXX,2,SetCallerID(${CALLERID})
exten = _0123456XXX,3,Set(CALLERID=${CALLERID})
exten = _0123456XXX,4,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5060,30,tr)
exten = _0123456XXX,103,Hangup
exten = t,1,Hangup
exten = i,1,Answer
exten = i,2,Playback(pbx-invalid)
exten = i,3,Hangup
---
Maybe I do not see my error...
Thanks
2005/11/14, Jens Kübler [EMAIL PROTECTED]:
Am Montag 14 November 2005 17:22 schrieb FaberK:
Hi guys,
this is the scenario:
PRI -Asterisk-SER
If I call from a Sip(SER) user everything is good, I can call
anywhere, but if I try to call from outside(PRI) everything is
wrong!!!
This is the CLI for an incoming call:
--
ast*CLI
-- Executing SetCallerID(Zap/14-1, outside) in new stack
-- Executing Set(Zap/14-1, CALLERID=outside) in new stack
-- Executing Dial(Zap/14-1,
SIP/[EMAIL PROTECTED]:5060|30|r) in new stack
-- Accepting call from 'outside' to 'mynumber201' on channel 0/14, span
1 -- Called [EMAIL PROTECTED]:5060
Nov 14 17:15:47 NOTICE[8459]: chan_sip.c:9492 handle_response_invite:
Failed to authenticate on INVITE to 'Unknown
sip:[EMAIL PROTECTED];tag=as6261e060'
-- SIP/sip.mydomain.com:5060-5eda is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Here we go
You haven't disabled general authentication (if you wish to)
or haven't set a proper default context in sip.conf
or you aren't handling the default incoming sip context properly in
extensions.conf
Jens
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.:FaberK:.
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[Asterisk-Users] Sangoma 102 installation problem
Hi friends, during the installation, I receive that problem, but I've installed both Flex and, of course, C/C++ libraries. My OS is CentOS 3.6, completely updated. Any ideas??? Thanks - Compiling WANPIPE WanCfg Utility ... Failed! !!! WANPIPE WanCfg Compilation Failed !!! Possible solution: FLEX Package not installed Non-standard C/C++ library (eg: ulibc) Please contact Sangoma Tech. at 905 474-1990 - -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma 102 installation problem
Hi Florian, yes, I have Flex available: whereis flex flex: /usr/bin/flex /usr/share/man/man1/flex.1.gz other ideas? 2005/11/8, Florian Overkamp [EMAIL PROTECTED]: Hi, FaberK wrote: during the installation, I receive that problem, but I've installed both Flex and, of course, C/C++ libraries. My OS is CentOS 3.6, completely updated. Any ideas??? Thanks - Compiling WANPIPE WanCfg Utility ... Failed! !!! WANPIPE WanCfg Compilation Failed !!! Possible solution: FLEX Package not installed Non-standard C/C++ library (eg: ulibc) Please contact Sangoma Tech. at 905 474-1990 So, is FLEX available on your system ? (I don't know CentOS) Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] compiling problems
The problem is the 2.6. I know that there is compability also with that kernel, but in my small experience, I've got not these problems with 2.4. Now, I've got to migrate Asterisk into a Dual Xeon 3.0 4Gb RAM. What distro would you use? Until now, I've tested CentOS 3.4 Server with no problem, but not on this kind of server. With Fedora 3, too many problems, concerning the kernel 2.6. Suggestions? Thanks 2005/11/6, Tzafrir Cohen [EMAIL PROTECTED]: On Sat, Nov 05, 2005 at 07:29:18PM +0100, FaberK wrote: Fedora Core 3 kernel-0-2.6.9-1.667 and kernel-2.6.12-1.1380 (same results) Sangoma 102 Concerning udev, I've read that it uses hotplug and if I'm not wrong, I remember that zaptel got conflicts with hotplug. But maybe I'm confusing (terrible headache!) Thanks a lot! zaptel should not conflict with hotplug if the specific hardware driver module is well-written (e.g: declares PCI IDs it will identify). This will mean that hotplug will try using it automatically. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] compiling problems
Hi guys, in my 3rd asterisk installation, I have a problem with zaptel modules. I use the CVS. Instead of obtaining, for example, zaptel.o I got zaptel.ko. What is the reason? Like that, also all the other zaptel kernel modules got the same extension. Also, zaptel do not create /proc/zaptel/1 and relative channels into /dev/zap. Inte /dev/zap I only got: channel ctl pseudo timer Any ideas/suggestions? Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] compiling problems
Fedora Core 3 kernel-0-2.6.9-1.667 and kernel-2.6.12-1.1380 (same results) Sangoma 102 Concerning udev, I've read that it uses hotplug and if I'm not wrong, I remember that zaptel got conflicts with hotplug. But maybe I'm confusing (terrible headache!) Thanks a lot! 2005/11/5, Tzafrir Cohen [EMAIL PROTECTED]: On Sat, Nov 05, 2005 at 01:59:43PM +0100, FaberK wrote: Hi guys, in my 3rd asterisk installation, I have a problem with zaptel modules. I use the CVS. Instead of obtaining, for example, zaptel.o I got zaptel.ko. What is the reason? That you use kernel 2.6. Do you have kernel 2.4 or 2.6? What's the output of 'uname -r'? Like that, also all the other zaptel kernel modules got the same extension. Also, zaptel do not create /proc/zaptel/1 and relative channels into /dev/zap. Inte /dev/zap I only got: channel ctl pseudo timer Any ideas/suggestions? My guess is that you use udev and thus channels will be generated when needed. What linux distribution do you use? What zaptel hardware do you have? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 Asterisk boxes
Hi to all, I need to setup 2 Asterisk boxes like this: PRI --- Asterisk1 --- IAX2 --- Asterisk2 --- PRI any samples? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Asterisk boxes
Hi Paul, thanks for your reply There are several examples on the wiki to do this. However you did not say well, before post here, I've searched into wiki, but I didn't found it. But if you say so, I'll check again. if you wanted Asterisk1 to use PRI attached to Asterisk 2 and Visa Versa. Yes, I want it If you do, after creating the IAX link, set up a dial plan the uses the IAX link as well as its own pri. Are you trying to do this in AMP? Or manually coding it. I want to code it manually. Paul -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of FaberK Sent: Wednesday, November 02, 2005 7:14 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] 2 Asterisk boxes Hi to all, I need to setup 2 Asterisk boxes like this: PRI --- Asterisk1 --- IAX2 --- Asterisk2 --- PRI any samples? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: messagenet
Hi, this is what I continuously see into the logs: Oct 22 10:26:07 NOTICE[26614]: chan_sip.c:6924 handle_response: Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as77222f33' (tries '2') Oct 22 10:26:26 NOTICE[26614]: chan_sip.c:4055 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Thanks 2005/10/22, FaberK [EMAIL PROTECTED]: Hi, is there somebody using messagenet.it? From yesterday, I can only call out, but if somebody call me is always busy. I'm talking about the geo-number. If somebody is using this service, please let me know if you are experiencing something like this, too. Bye -- .:FaberK:. -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] messagenet
Hi, is there somebody using messagenet.it? From yesterday, I can only call out, but if somebody call me is always busy. I'm talking about the geo-number. If somebody is using this service, please let me know if you are experiencing something like this, too. Bye -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Planet Vip-150T
Hi All, I'm having problem with this phone. Problems are regarding voicemail message alert on the phone. --- handle_response: Host 'xxx.xxx.xxx.xxx' does not implement 'NOTIFY' --- Can somebody help? On the phone manual, is written that it can acept MWI, but... not mine!!! Thanks! -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail 2
Hi list, I'm trying, as usual, to set up voicemail. It works, but signaling to phones, doesn't. Into XLite logs, I have: -- Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 1/0 (0/0) -- but nothing appear on the XLite screen. So, I understand that I'm able to send the right signal, but something is still wrong. Ideas? Thanks in advance -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 2
I've got it too until yesterday!!!
Now no more envelope either.
This is from extensions.conf:
---
exten = 221,1,Dial(SIP/221,20,tr)
exten = 221,2,Voicemail(u${EXTEN})
exten = 221,102,Voicemail(b${EXTEN})
exten = 221,103,Hangup
---
this is from sip.conf:
---
[221]
type=friend
username=221
secret=221
callerid=221 221
fromuser=221
accountcode=221
context=local
host=dynamic
dtmfmode=rfc2833
nat=yes
qualify=yes
Port=5060
[EMAIL PROTECTED]
Disallow=all
Allow=gsm
Allow=ulaw
Allow=alaw
---
Some ideas?
2005/10/16, Linc Fessenden [EMAIL PROTECTED]:
FaberK wrote:
Hi list,
I'm trying, as usual, to set up voicemail.
It works, but signaling to phones, doesn't.
Into XLite logs, I have:
--
Messages-Waiting: yes
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 1/0 (0/0)
--
but nothing appear on the XLite screen.
So, I understand that I'm able to send the right signal, but something
is still wrong.
Ideas?
Thanks in advance
--
.:FaberK:.
I have something similar. I have the little mail envelope on the screen
of xten-xlite, but can't figure out how to clear it off.
--
-Linc Fessenden
In the Beginning there was nothing, which exploded - Yeah right...
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Re: [Asterisk-Users] Voicemail 2
First answer:
the envelope came back to me!
Yesterday I've added the line:
notifymimetype=text/plain
into sip.conf, so no more envelope just text into logs.
Is just an answer to you, Linc.
Nothing for me, yet.
2005/10/16, FaberK [EMAIL PROTECTED]:
I've got it too until yesterday!!!
Now no more envelope either.
This is from extensions.conf:
---
exten = 221,1,Dial(SIP/221,20,tr)
exten = 221,2,Voicemail(u${EXTEN})
exten = 221,102,Voicemail(b${EXTEN})
exten = 221,103,Hangup
---
this is from sip.conf:
---
[221]
type=friend
username=221
secret=221
callerid=221 221
fromuser=221
accountcode=221
context=local
host=dynamic
dtmfmode=rfc2833
nat=yes
qualify=yes
Port=5060
[EMAIL PROTECTED]
Disallow=all
Allow=gsm
Allow=ulaw
Allow=alaw
---
Some ideas?
2005/10/16, Linc Fessenden [EMAIL PROTECTED]:
FaberK wrote:
Hi list,
I'm trying, as usual, to set up voicemail.
It works, but signaling to phones, doesn't.
Into XLite logs, I have:
--
Messages-Waiting: yes
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 1/0 (0/0)
--
but nothing appear on the XLite screen.
So, I understand that I'm able to send the right signal, but something
is still wrong.
Ideas?
Thanks in advance
--
.:FaberK:.
I have something similar. I have the little mail envelope on the screen
of xten-xlite, but can't figure out how to clear it off.
--
-Linc Fessenden
In the Beginning there was nothing, which exploded - Yeah right...
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
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http://lists.digium.com/mailman/listinfo/asterisk-users
--
.:FaberK:.
--
.:FaberK:.
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Re: [Asterisk-Users] Voicemail 2
well, is just the context. You could call it as you prefer, mickeymouse??? ;o) Bye 2005/10/16, Linc Fessenden [EMAIL PROTECTED]: FaberK wrote: [EMAIL PROTECTED] --- Some ideas? Only thing I have that even looks different is [EMAIL PROTECTED] -- -Linc Fessenden In the Beginning there was nothing, which exploded - Yeah right... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 2
Yes, is defined in voicemail.conf too. The context is the glue of the system, this is what I understood, the way to follow, the arrow that show the direction to every application. But the problems, still remain. 2005/10/16, Jason Walker [EMAIL PROTECTED]: Correct - but is the context defined in voicemail.conf? As mickeymouse? Or whatever...? ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Saturday, October 15, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail 2 well, is just the context. You could call it as you prefer, mickeymouse??? ;o) Bye 2005/10/16, Linc Fessenden [EMAIL PROTECTED]: FaberK wrote: [EMAIL PROTECTED] --- Some ideas? Only thing I have that even looks different is [EMAIL PROTECTED] -- -Linc Fessenden In the Beginning there was nothing, which exploded - Yeah right... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PA168S/AT320P
Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local [EMAIL PROTECTED] callerid=221 221 accountcode=221 qualify=yes Any ideas? Thanks to all. -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PA168S/AT320P
Hi, thanks to reply: 1)SIP 2)yes. I've used the original 1.46 for SIP protocol Also your solution do not work. Are 2 days that I'm trying configurations and googling for this problem, but nothing! Always: LOG ON FAILED I've saw about problems with this phone, but my hope was that with the new firmware something could be solved. Thanks again. 2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]: 1)What is the protocol you are using? SIP or IAX2? 2)Have you applied the correct firmware to the Phone? Pa168 phones are falwless when connecting to Asterisk. Start the configuration as asimple entry as under. I have added Port address and allowed codecs in the config below: [221] type=friend username=221 secret=secret context=local host=dynamic dtmfmode=rfc2833 nat=yes Port=5060 Disallow=all Allow=g729 Allow=ulaw Allow=gsm Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PA168S/AT320P Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local [EMAIL PROTECTED] callerid=221 221 accountcode=221 qualify=yes Any ideas? Thanks to all. -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PA168S/AT320P
Right now, but nothing changed. 2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]: have you configured the STUN server on the phone to any one of the available stun servers like stun.xten.net? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PA168S/AT320P Hi, thanks to reply: 1)SIP 2)yes. I've used the original 1.46 for SIP protocol Also your solution do not work. Are 2 days that I'm trying configurations and googling for this problem, but nothing! Always: LOG ON FAILED I've saw about problems with this phone, but my hope was that with the new firmware something could be solved. Thanks again. 2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]: 1)What is the protocol you are using? SIP or IAX2? 2)Have you applied the correct firmware to the Phone? Pa168 phones are falwless when connecting to Asterisk. Start the configuration as asimple entry as under. I have added Port address and allowed codecs in the config below: [221] type=friend username=221 secret=secret context=local host=dynamic dtmfmode=rfc2833 nat=yes Port=5060 Disallow=all Allow=g729 Allow=ulaw Allow=gsm Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PA168S/AT320P Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local [EMAIL PROTECTED] callerid=221 221 accountcode=221 qualify=yes Any ideas? Thanks to all. -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming calls
Hi, stupid question: how can I let to call an extensions from outside? Untill now, I've just the possibility to call our number and then, after the system answer, dial the extension. My sistem is like this: SER - internal extensions Asterisk - incoming/outgoing gateway. FaberK -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] netappel
Hi guys, does anybody succesfully connect Asterisk with netappel? I've tryed using voipbuster settings, but doesn't work. Any suggestions, are wellcome. Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voipbuster advise
Hi, I'm using voipbuster at work, and I've got 2 questions: 1) Is it possible to send faxes using voipbuster connex? 2) Is it possible to cut off or cover the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when I'm with clients. Any links, suggestions? Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR problem
Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] more accounts
Hi guys, one question: I've got 2 IAX accounts, and I would like to let use them in the same time, so that if one is busy I can call using the other? Thanks-- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Messagenet.it
Hi to all, I need help to setting up messagenet.it account in Asterisk. My * is connected with static IP to the net. No other cards at the moment, just the network-card. I'm able to receive call on the geographical number, but I'm not able to setup the outgoing calls. All that it does is: I dial the number(not voip), * accepts my call, I can hear the ring tone, but the telephone called do not receive the call. I'm sure, because I called my number and I've picked up the phone while is was ringing, and the call was not there... I'm using port 5061 and rtp port 8000. Username and password are correct. I'm sure because I've setup a voip phone with those and it worked. Any ideas? Thanks a lot-- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323 build problems
Hi Read README file first. You will get a clue. thanks for you suggestions, but I always read README file, before starting any installation. I've also googled my problem before post here. Thaks again -- .:FaberK:. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323 build problems
What versions of OpenH323/Pwlib/asterisk-oh323 are you trying to install? OpenH323=1.12.2 Pwlib=1.5.2 asterisk-oh323=0.2 Thank you Michael Michael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323 building problems
Hello Guys, first of all, I'm very new with asterisk. I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7 Now I'm trying with asterisk-oh323 I've already installed pwlib, oh323 and I've already set the variables. Now, when I try to make asterisk-oh323 I receive this error messagge: for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/root/voip/asterisk/asterisk-oh323/wrapper' g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3 -DNDEBUG -I/usr/include -I/usr/include/crypto -I/usr/lib/pwlib/include/ptlib/unix -I/usr/lib/pwlib/include -I/usr/lib/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o wrapper.o wrapper.cxx: In constructor `WrapH323Connection::WrapH323Connection(WrapH323EndPoint, unsigned int, int, int, short unsigned int)': wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this function) wrapper.cxx:563: (Each undeclared identifier is reported only once for each function it appears in.) wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)': wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread' make[1]: *** [wrapper.o] Error 1 make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper' make: *** [subdirs_all] Error 1 What's wrong? Thanks -- .:FaberK:. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323 build problems
Hello Guys, first of all, I'm very new with asterisk. I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7 Now I'm trying with asterisk-oh323 I've already installed pwlib, oh323 and I've already set the variables. Now, when I try to make asterisk-oh323 I receive this error messagge: for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/root/voip/asterisk/asterisk-oh323/wrapper' g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3 -DNDEBUG -I/usr/include -I/usr/include/crypto -I/usr/lib/pwlib/include/ptlib/unix -I/usr/lib/pwlib/include -I/usr/lib/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o wrapper.o wrapper.cxx: In constructor `WrapH323Connection::WrapH323Connection(WrapH323EndPoint, unsigned int, int, int, short unsigned int)': wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this function) wrapper.cxx:563: (Each undeclared identifier is reported only once for each function it appears in.) wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)': wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread' make[1]: *** [wrapper.o] Error 1 make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper' make: *** [subdirs_all] Error 1 What's wrong? Thanks -- .:FaberK:. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
