Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Hello, On Thu, May 19, 2011 at 3:40 PM, A E [Gmail] all.efor...@gmail.com wrote: That's WAYY too much info for me to go through right now, and I don't know anything about TLS registration but what I would ask for is if you have the following lines in your sip.conf domain=IP/FQDN of your asterisk server:TLS port so in your case add the lines domain=pbx.domain.com:5061 and then do a sip reload So far, all problems I've had, have been solved because of this. At the end of your sip.conf add those lines and it should fix your problem. Thanks, but it didn't worked. :-( Any other possible solution for this problem? Regards, GNUbie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
On 11-05-22 07:44 AM, GNUbie wrote: Hello, On Thu, May 19, 2011 at 3:40 PM, A E [Gmail]all.efor...@gmail.com wrote: That's WAYY too much info for me to go through right now, and I don't know anything about TLS registration but what I would ask for is if you have the following lines in your sip.conf domain=IP/FQDN of your asterisk server:TLS port so in your case add the lines domain=pbx.domain.com:5061 and then do a sip reload So far, all problems I've had, have been solved because of this. At the end of your sip.conf add those lines and it should fix your problem. Thanks, but it didn't worked. :-( Any other possible solution for this problem? It is possible this is a regressions, if you roll back to 1.8.3 does it work? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
see: https://issues.asterisk.org/view.php?id=19182 On May 22, 2011, at 7:00 PM, asterisk-users-requ...@lists.digium.com wrote: Date: Sun, 22 May 2011 19:44:21 +0800 From: GNUbie gnu...@gmail.com Subject: Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: BANLkTin9TROuRB4_-g7JHfGYa59YemsW=g...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Hello, On Thu, May 19, 2011 at 3:40 PM, A E [Gmail] all.efor...@gmail.com wrote: That's WAYY too much info for me to go through right now, and I don't know anything about TLS registration but what I would ask for is if you have the following lines in your sip.conf domain=IP/FQDN of your asterisk server:TLS port so in your case add the lines domain=pbx.domain.com:5061 and then do a sip reload So far, all problems I've had, have been solved because of this. At the end of your sip.conf add those lines and it should fix your problem. Thanks, but it didn't worked. :-( Any other possible solution for this problem? Regards, GNUbie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: FW: realtime mysql - p4]
On Fri, 2011-05-20 at 10:05 +0100, Andrew Thomas wrote: Post your cdr_mysql.conf and res_mysql.conf and we'll take it from there. Don't forget to remove any 'private' info first (like passwords). Cheers Tnx for the offer, Wil get the files when got back at the office. I presume that cdr_mysql.conf is only relevant for storing call-data-records? Perhaps that is something for later on. For now, i have to show a working *, with all sip-details in a mysql-DB. Other people pointed out that other means (postgres, ldap) might work better, but that's not an option for me. Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call files .vbs
This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? If I were to build a call file script (described in this link http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then how does it work if my Asterisk machine is running on Centos 5.5? I simply want to execute a script that helps me automate the voice broadcasting/IVR of up to 1 phone numbers. Thank you Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
Thomas Perron wrote: Can a vb script run somehow on a Linux machine or does it only work on Windows? Visual Basic is Windows specific. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
Hi Doug, Yes. I have sorted that part out. Also, it seems like the pscp function is the way that I can tie together the vb script with the logic of the Asterisk call files learning curve!! Thanks On Sun, May 22, 2011 at 8:37 PM, Doug Lytle supp...@drdos.info wrote: Thomas Perron wrote: Can a vb script run somehow on a Linux machine or does it only work on Windows? Visual Basic is Windows specific. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
On Sun, 22 May 2011, Thomas Perron wrote: Can a vb script run somehow on a Linux machine or does it only work on Windows? Virtual machines or Wine may have some possibilities. I simply want to execute a script that helps me automate the voice broadcasting/IVR of up to 1 phone numbers. Write a script that executes on the Asterisk box. Where do the 10,000 numbers come from? Executing a script on the Asterisk box will enable you to monitor the status of the process better. Like only dumping xxx scripts at a time into the spool directory and sending you an email if Asterisk stops processing them for some reason. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
On Sun, 22 May 2011, Thomas Perron wrote: Also, it seems like the pscp function is the way that I can tie together the vb script with the logic of the Asterisk call files learning curve!! pscp is a program, not a function. Part of or related to putty as I remember. Not a good idea. One of the 'bugaboos' of call files is that you are supposed to create the files in a temporary directory and move them into the spool directory. Also, you will have limited error detection ability if you are only dumping files 'willy-nilly.' Much better to do it all on the Asterisk host. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] difference between SIP peer and SIP user ?
hello: please refer this link: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Sat, 21 May 2011 17:49:37 +0530 From: virbh...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] difference between SIP peer and SIP user ? Hi list, I am confuse about these CLI commands sip show users sip show peers Can someone clear my doubt . what are the difference between them? - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Hello Paul, On Sun, May 22, 2011 at 11:55 PM, Paul Belanger pabelan...@digium.com wrote: It is possible this is a regressions, if you roll back to 1.8.3 does it work? I'm using the binary .deb packages from the upstream project (Digium). *CLI core show version Asterisk 1.8.3.3-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-04-22 17:50:44 UTC Unless, you want me to test install version 1.8.3 from source. Please advice. Regards, GNUbie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Hello Marcelo, On Mon, May 23, 2011 at 2:48 AM, Marcello Ceschia marcello.cesc...@gmx.net wrote: see: https://issues.asterisk.org/view.php?id=19182 That explains. Thanks. Is the patch already applied in the Asterisk version 1.8.4 because I can't find it on its changelog? Regards, GNUbie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
I'm the original author of said VB Script. Steve is right, I had lots of errors - related to the fact that asterisk watches it too closely and reads the files even before they are complete - and have since updated it that it first dumps it to a temp directory, then use a bash script on the linux machine that moves all files from the temp directory to the call directory using plink. Both pscp and plink are windoz programs that utilize ssh for their functions. Pscp xfers files, and plink executes any remote commands. In the newer version pscp in the VB Script dumps it to /root/calltemps/ and /root/mvcallfiles.sh moves the files from /root/calltemps/* to /var/spool/asterisk/outgoing/ change this line: strcmd=C:\pscp -pw password c:\direcotry\strcnt\* root@asterisk:/var/spool/asterisk/outgoing to: strcmd=C:\pscp -pw password c:\directory\strcnt\* root@asterisk:/root/calltemps make sure the dir exists then add: Set objShell2 = CreateObject(WScript.Shell) strcmd2=C:\plink -pw password root@asterisk /root/mvcallfiles.sh objShell2.Run strcmd2 /root/movcallfiles.sh: #/bin/bash mv /root/calltemps/* /var/spool/asterisk/outgoing/ Hope this helps. On Sun, May 22, 2011 at 8:55 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 22 May 2011, Thomas Perron wrote: Also, it seems like the pscp function is the way that I can tie together the vb script with the logic of the Asterisk call files learning curve!! pscp is a program, not a function. Part of or related to putty as I remember. Not a good idea. One of the 'bugaboos' of call files is that you are supposed to create the files in a temporary directory and move them into the spool directory. Also, you will have limited error detection ability if you are only dumping files 'willy-nilly.' Much better to do it all on the Asterisk host. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
