Re: [asterisk-users] Asterisk 1.6
that is definitely another options, thanks for the range of options provided, Thanks On Sat, Apr 5, 2014 at 4:51 PM, Duncan Turnbull dun...@e-simple.co.nzwrote: Another option we like, but i depends on your preferences is to run them over openvpn. Works for Mac, Linux and Windows clients. Since all out clients are under our control we use openvpn a lot and yealink and other phones have it built in so they can connect directly once initially setup Cheers Duncan On 5/04/2014, at 4:36 am, motty cruz motty.c...@gmail.com wrote: that sounds feasible, Thanks Michelle, On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup...@ocg.ca wrote: If you know your users are all from with your country, or state, or even city, you could restrict geographic access in your secast.conf file like this: ruledefault=deny ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA The above would: - By default deny all source IP's anywhere in the world - Let in only source IP's from: 1. North America (continent), Canada (country), Ontario (region) 2. North America (continent), USA (country), Michigan (region), Detroit (city) 3. Any region called 'Ohio' anywhere in the world (not sure why you would do that but fun example) 4. Anywhere in North America So you can open up your system based solely on where you know your real users are located. -=Michelle=- -- *From:* asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of motty cruz motty.c...@gmail.com *Sent:* Friday, April 4, 2014 11:15 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Asterisk 1.6 Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- _ -- 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 -- _ -- 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] Asterisk 1.6
Another option we like, but i depends on your preferences is to run them over openvpn. Works for Mac, Linux and Windows clients. Since all out clients are under our control we use openvpn a lot and yealink and other phones have it built in so they can connect directly once initially setup Cheers Duncan On 5/04/2014, at 4:36 am, motty cruz motty.c...@gmail.com wrote: that sounds feasible, Thanks Michelle, On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup...@ocg.ca wrote: If you know your users are all from with your country, or state, or even city, you could restrict geographic access in your secast.conf file like this: ruledefault=deny ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA The above would: - By default deny all source IP's anywhere in the world - Let in only source IP's from: 1. North America (continent), Canada (country), Ontario (region) 2. North America (continent), USA (country), Michigan (region), Detroit (city) 3. Any region called 'Ohio' anywhere in the world (not sure why you would do that but fun example) 4. Anywhere in North America So you can open up your system based solely on where you know your real users are located. -=Michelle=- From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of motty cruz motty.c...@gmail.com Sent: Friday, April 4, 2014 11:15 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk 1.6 Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- _ -- 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
[asterisk-users] Asterisk 1.6
Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- _ -- 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] Asterisk 1.6
I don't know what platform you are on, but if you are on Linux (and possibly BSD) you could use fail2ban to block them at the network interface. On 04/04/2014 09:00 AM, motty cruz wrote: Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- Daniel Taylor VP OperationsVocal Laboratories, Inc. dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- 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] Asterisk 1.6
On 4 April 2014 15:00, motty cruz motty.c...@gmail.com wrote: Hello All, my asterisk server is constantly under attack Unfortunately you are not alone. [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Check out fail2ban. Works well. Hope this helps. -Barry Flanagan Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- _ -- 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] Asterisk 1.6
On Fri, Apr 4, 2014 at 10:05 AM, Daniel Taylor dtay...@vocalabs.com wrote: I don't know what platform you are on, but if you are on Linux (and possibly BSD) you could use fail2ban to block them at the network interface. I second fail2ban. If you need some ideas to configure it, you can steal them from the freepbx setup. How many sip phones do you have outside your network? If few and in well-known IPs, consider limiting access to only those (and the sip provider you are using). On 04/04/2014 09:00 AM, motty cruz wrote: Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- Daniel Taylor VP OperationsVocal Laboratories, inc.dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- 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] Asterisk 1.6
thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. On Fri, Apr 4, 2014 at 7:09 AM, Mauricio Tavares raubvo...@gmail.comwrote: On Fri, Apr 4, 2014 at 10:05 AM, Daniel Taylor dtay...@vocalabs.comwrote: I don't know what platform you are on, but if you are on Linux (and possibly BSD) you could use fail2ban to block them at the network interface. I second fail2ban. If you need some ideas to configure it, you can steal them from the freepbx setup. How many sip phones do you have outside your network? If few and in well-known IPs, consider limiting access to only those (and the sip provider you are using). On 04/04/2014 09:00 AM, motty cruz wrote: Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- Daniel Taylor VP OperationsVocal Laboratories, inc.dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- 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 -- _ -- 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] Asterisk 1.6
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/ It does everything fail2ban does and more, including blocking users by geography (we exclude all of Asia and Africa), detection of break-in patterns (even if someone guessed your un/pw), detect changes in dial rates, etc. Grab the free version - its a BIG step up from fail2ban. -=Michelle=-? All opions posted are my person ones. And personnally I like generationd products because I work for them :) From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of motty cruz motty.c...@gmail.com Sent: Friday, April 4, 2014 10:00 AM To: Asterisk Users List Subject: [asterisk-users] Asterisk 1.6 Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132[X]194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132[X]194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132[X]194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132[X]194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132[X]194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- _ -- 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] Asterisk 1.6
On Friday 04 Apr 2014, Michelle Dupuis wrote: Take a look a SecAst from www.generationd.comhttp://www.generationd.com/ It does everything fail2ban does and more, including blocking users by geography (we exclude all of Asia and Africa), detection of break-in patterns (even if someone guessed your un/pw), detect changes in dial rates, etc. Grab the free version - its a BIG step up from fail2ban. That link points towards a precompiled binary, which could have literally *anything* lurking in it. I politely advise you to back away slowly, and break into a run when you think you are out of sight. Precompiled binaries without Source Code should be treated like a bottle of glowing green liquid labelled drink me, or an offer to come and look at some puppies. No reputable software supplier would object to showing you what is on the inside. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Asterisk 1.6
What you are saying is only open source software is safe? You have just excluded most software in use in the business world. We have installed Norton antivirus on all of our workstation; I don't think Symantec will ever release the source code (since that would also show attackers how to get around it). Using the same logic releasing SecAst source would also seem foolish (and make it impossible for any commercial enterprise to sell software). I understand your point of view, and if your preference is to only use open source software that's great. However, that doesn't mean precompiled software is inherently dangerous or malevolent. -=Michelle=- From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of A J Stiles asterisk_l...@earthshod.co.uk Sent: Friday, April 4, 2014 10:38 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk 1.6 On Friday 04 Apr 2014, Michelle Dupuis wrote: Take a look a SecAst from www.generationd.comhttp://www.generationd.com/ It does everything fail2ban does and more, including blocking users by geography (we exclude all of Asia and Africa), detection of break-in patterns (even if someone guessed your un/pw), detect changes in dial rates, etc. Grab the free version - its a BIG step up from fail2ban. That link points towards a precompiled binary, which could have literally *anything* lurking in it. I politely advise you to back away slowly, and break into a run when you think you are out of sight. Precompiled binaries without Source Code should be treated like a bottle of glowing green liquid labelled drink me, or an offer to come and look at some puppies. No reputable software supplier would object to showing you what is on the inside. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Asterisk 1.6
absolutely right A J, thanks for the heads up. I do not intent to implement that solution in production server, I hope to learn it first, build a test server and monitor for a few days or weeks. Thanks again, On Fri, Apr 4, 2014 at 7:38 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Friday 04 Apr 2014, Michelle Dupuis wrote: Take a look a SecAst from www.generationd.com http://www.generationd.com/ It does everything fail2ban does and more, including blocking users by geography (we exclude all of Asia and Africa), detection of break-in patterns (even if someone guessed your un/pw), detect changes in dial rates, etc. Grab the free version - its a BIG step up from fail2ban. That link points towards a precompiled binary, which could have literally *anything* lurking in it. I politely advise you to back away slowly, and break into a run when you think you are out of sight. Precompiled binaries without Source Code should be treated like a bottle of glowing green liquid labelled drink me, or an offer to come and look at some puppies. No reputable software supplier would object to showing you what is on the inside. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Asterisk 1.6
On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Asterisk 1.6
Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Asterisk 1.6
Well in that case fail2ban gets my vote. On 4 April 2014 16:15, motty cruz motty.c...@gmail.com wrote: Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Asterisk 1.6
If you know your users are all from with your country, or state, or even city, you could restrict geographic access in your secast.conf file like this: ruledefault=deny ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA The above would: - By default deny all source IP's anywhere in the world - Let in only source IP's from: 1. North America (continent), Canada (country), Ontario (region) 2. North America (continent), USA (country), Michigan (region), Detroit (city) 3. Any region called 'Ohio' anywhere in the world (not sure why you would do that but fun example) 4. Anywhere in North America So you can open up your system based solely on where you know your real users are located. -=Michelle=- From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of motty cruz motty.c...@gmail.com Sent: Friday, April 4, 2014 11:15 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk 1.6 Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.ukmailto:i...@pack-net.co.uk wrote: On 4 April 2014 15:22, motty cruz motty.c...@gmail.commailto:motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994tel:%2B44%20%280%29845%20004%204994 f: +44 (0)161 660 9825tel:%2B44%20%280%29161%20660%209825 e: i...@pack-net.co.ukmailto:i...@pack-net.co.uk w: http://www.pack-net.co.ukhttp://www.pack-net.co.uk/ Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://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] Asterisk 1.6
Use allowguest=no And define ACLs for every SIP account. And obviously, fail2ban for blocking suspicious IPs. -- _ -- 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] Asterisk 1.6
that sounds feasible, Thanks Michelle, On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup...@ocg.ca wrote: If you know your users are all from with your country, or state, or even city, you could restrict geographic access in your secast.conf file like this: ruledefault=deny ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA The above would: - By default deny all source IP's anywhere in the world - Let in only source IP's from: 1. North America (continent), Canada (country), Ontario (region) 2. North America (continent), USA (country), Michigan (region), Detroit (city) 3. Any region called 'Ohio' anywhere in the world (not sure why you would do that but fun example) 4. Anywhere in North America So you can open up your system based solely on where you know your real users are located. -=Michelle=- -- *From:* asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of motty cruz motty.c...@gmail.com *Sent:* Friday, April 4, 2014 11:15 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Asterisk 1.6 Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- _ -- 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] Asterisk 1.6
Shouldn't the secast discussion be on the commercial list? Note that their free version works for five simultaneous calls-then the price goes 'way up. --Don (Top posting 'cause that's what's already being done.) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz Sent: Friday, April 04, 2014 10:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 that sounds feasible, Thanks Michelle, On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup...@ocg.ca wrote: If you know your users are all from with your country, or state, or even city, you could restrict geographic access in your secast.conf file like this: ruledefault=deny ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA The above would: - By default deny all source IP's anywhere in the world - Let in only source IP's from: 1. North America (continent), Canada (country), Ontario (region) 2. North America (continent), USA (country), Michigan (region), Detroit (city) 3. Any region called 'Ohio' anywhere in the world (not sure why you would do that but fun example) 4. Anywhere in North America So you can open up your system based solely on where you know your real users are located. -=Michelle=- _ From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of motty cruz motty.c...@gmail.com Sent: Friday, April 4, 2014 11:15 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk 1.6 Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 tel:%2B44%20%280%29845%20004%204994 f: +44 (0)161 660 9825 tel:%2B44%20%280%29161%20660%209825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk http://www.pack-net.co.uk/ Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 -- _ -- 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] [Asterisk 1.6] Mysql cdr addon doen't write full channel infomation when disposition is Failed
Hi All, Currently i'm facing with a cdr issue, When i originate a call (outbound call) to uncorrect/unregistration user, asterisk inform me that call was failed but in mysl-cdr (cdr-csv also) records. Here are 2 samples +-+--+-+-+--+-++--+-+---+ | calldate| clid | src | dst | dcontext | channel | dstchannel | duration | disposition | userfield | +-+--+-+-+--+-++--+-+---+ | 2013-03-30 11:01:20 | | | s | default | | |0 | FAILED | | | 2013-03-22 08:45:00 | | | 777 | from-avc | SIP/083777-0013 | | 19 | ANSWERED| 16| Thank and Appreciate if any social experiences can help me on this. BRs. -- TrungND -- _ -- 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] Asterisk 1.6 + Cisco AS5300
Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 Bad Extension back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-004d == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-004d' Do you have an explanation? Best regards, Mickael -- _ -- 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] Asterisk 1.6 + Cisco AS5300
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: Do you have an explanation? Put a SIP debug on and we may be able to find one.. Steve -- _ -- 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] Asterisk 1.6 + Cisco AS5300
Le 7/03/13 11:21, Steven Howes a écrit : On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: Do you have an explanation? Put a SIP debug on and we may be able to find one.. Steve Hello Steve, After checking, I confirm that the call is cut precisely to 900 seconds (15 min). 10.4.0.1 = Asterisk 10.4.0.10 = Cisco AS 5300 Info : debug start at 14min30sec set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Audio is at 10.4.0.1 port 11842 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 10.4.0.10:54789: INVITE sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE User-Agent: isdnbox1.1 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 207 v=0 o=root 1538728127 1538728127 IN IP4 10.4.0.1 s=Asterisk PBX 1.6.2.9-2+squeeze8 c=IN IP4 10.4.0.1 t=0 0 m=audio 11842 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 420 Bad Extension Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE Unsupported: timer Content-Length: 0 - --- (8 headers 0 lines) --- -- Got SIP response 420 Bad Extension back from 10.4.0.10 set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Transmitting (NAT) to 10.4.0.10:5060: ACK sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 ACK User-Agent: isdnbox1.1 Content-Length: 0 --- -- Stopped music on hold on SIP/as5300-1-0050 == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-0050' Reliably Transmitting (NAT) to 10.4.0.10:5060: OPTIONS sip:10.4.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport Max-Forwards: 70 From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10 Contact: sip:asterisk@10.4.0.1 Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 CSeq: 102 OPTIONS User-Agent: isdnbox1.1 Date: Thu, 07 Mar 2013 11:17:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10;tag=37A724C-211C Date: Sat, 01 Jan 2000 16:12:32 GMT Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp CSeq: 102 OPTIONS Supported: 100rel Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Accept: application/sdp Allow-Events: telephone-event Content-Length: 154 v=0 o=CiscoSystemsSIP-GW-UserAgent 793 5073 IN IP4 10.4.0.10 s=SIP Call c=IN IP4 10.4.0.10 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 10.4.0.10 - --- (14 headers 7 lines) --- Really destroying SIP dialog '6a43ad4b27d870d048e8425077bcc075@10.4.0.1' Method: OPTIONS --- SIP read from UDP:10.4.0.10:54336 --- BYE sip:65939191@10.4.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.10:5060 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Date: Sat, 01 Jan 2000 16:12:26 GMT Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 6 Timestamp: 946743153 CSeq: 102 BYE Content-Length: 0 - --- (11 headers 0 lines) --- --- Transmitting (NAT) to 10.4.0.10:54336 --- SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.4.0.10:5060;received=10.4.0.10 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 BYE Server: isdnbox1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 15 min (call ended) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
Can u debug on AS ? On Thu, Mar 7, 2013 at 9:20 AM, Mickael Monsieur mickael.monsi...@gmail.com wrote: Le 7/03/13 11:21, Steven Howes a écrit : On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: Do you have an explanation? Put a SIP debug on and we may be able to find one.. Steve Hello Steve, After checking, I confirm that the call is cut precisely to 900 seconds (15 min). 10.4.0.1 = Asterisk 10.4.0.10 = Cisco AS 5300 Info : debug start at 14min30sec set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Audio is at 10.4.0.1 port 11842 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 10.4.0.10:54789: INVITE sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE User-Agent: isdnbox1.1 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 207 v=0 o=root 1538728127 1538728127 IN IP4 10.4.0.1 s=Asterisk PBX 1.6.2.9-2+squeeze8 c=IN IP4 10.4.0.1 t=0 0 m=audio 11842 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 420 Bad Extension Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE Unsupported: timer Content-Length: 0 - --- (8 headers 0 lines) --- -- Got SIP response 420 Bad Extension back from 10.4.0.10 set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Transmitting (NAT) to 10.4.0.10:5060: ACK sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 ACK User-Agent: isdnbox1.1 Content-Length: 0 --- -- Stopped music on hold on SIP/as5300-1-0050 == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-0050' Reliably Transmitting (NAT) to 10.4.0.10:5060: OPTIONS sip:10.4.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport Max-Forwards: 70 From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10 Contact: sip:asterisk@10.4.0.1 Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 CSeq: 102 OPTIONS User-Agent: isdnbox1.1 Date: Thu, 07 Mar 2013 11:17:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10;tag=37A724C-211C Date: Sat, 01 Jan 2000 16:12:32 GMT Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp CSeq: 102 OPTIONS Supported: 100rel Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Accept: application/sdp Allow-Events: telephone-event Content-Length: 154 v=0 o=CiscoSystemsSIP-GW-UserAgent 793 5073 IN IP4 10.4.0.10 s=SIP Call c=IN IP4 10.4.0.10 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 10.4.0.10 - --- (14 headers 7 lines) --- Really destroying SIP dialog '6a43ad4b27d870d048e8425077bcc075@10.4.0.1' Method: OPTIONS --- SIP read from UDP:10.4.0.10:54336 --- BYE sip:65939191@10.4.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.10:5060 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Date: Sat, 01 Jan 2000 16:12:26 GMT Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 6 Timestamp: 946743153 CSeq: 102 BYE Content-Length: 0 - --- (11 headers 0 lines) --- --- Transmitting (NAT) to 10.4.0.10:54336 --- SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.4.0.10:5060;received=10.4.0.10 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 BYE Server: isdnbox1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
Le 7/03/13 11:12, Mickael Monsieur a écrit : Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 Bad Extension back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-004d == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-004d' Do you have an explanation? Best regards, Mickael Ok i solved : https://issues.asterisk.org/jira/browse/ASTERISK-15787 -- _ -- 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] Asterisk 1.6 / voicemail / final voice auth-thankyou
Hi, voicemail plays after hitting # as final file auth-thankyou. Is there any possibility to change this behaviour? Custom soundfile or disable it perhaps? Thanks for your answer(s)! -Thorsten- -- _ -- 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] Asterisk 1.6 and Outbound SIP Proxy configuration
Update: No luck with versions 1.6 and 1.8.7 I had to revert back to 1.4 which worked with no problem. Probably if I have some time, I will do more testing with version 1.8.7 to see what the difference is and what changes need to be made for this kind of setup to work in 1.8.7 Joseph On Mon, Aug 6, 2012 at 10:59 AM, Joseph Begumisa j.begum...@gmail.comwrote: Hello, Using asterisk 1.6 as sip client to register with sip provider and terminate calls through them. SIP Provider has provided sip proxy and sip server details. The problem is that the sip server FQDN does not resolve on the internet. So I can only presume that the SIP proxy knows how to reach the sip server. Asterisk 1.6 seems to have a problem with this. This is my config below: -- [trunk1] defaultuser=x...@sip.provider.com fromuser= fromdomain=sip.provider.com type=peer secret=a outboundproxy=10.10.10.10 ;(replaced actual ip) nat=no host=sip.provider.com dtmfmode=auto disallow=all context=from-internal canreinvite=no allow=g729 trustrpid=yes sendrpid=yes register = x...@sip.provider.com:a@10.10.10.10:5060 -- With the above config, I can register with the providers sip proxy, however, the error below is observed in the logs concerning the host when I try to make a call: -- [2012-08-02 23:37:31] WARNING[26155] acl.c: Unable to lookup ' sip.provider.com' [2012-08-02 23:37:31] ERROR[26155] chan_sip.c: srvlookup failed for host: sip.provider.com, on peer trunk1, removing peer -- I have done some research on this issue but not been able to find anything conclusive on why this would happen. I tested the sip details provided with a different sip client (actually an IP phone) and was able to register and send / receive calls with no problem. The problem just seems to be somewhere in my asterisk client configuration or a known bug with the version of asterisk I am using for this. Any pointers? Thanks. Joseph -- _ -- 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] Asterisk 1.6 and Outbound SIP Proxy configuration
Hello, Using asterisk 1.6 as sip client to register with sip provider and terminate calls through them. SIP Provider has provided sip proxy and sip server details. The problem is that the sip server FQDN does not resolve on the internet. So I can only presume that the SIP proxy knows how to reach the sip server. Asterisk 1.6 seems to have a problem with this. This is my config below: -- [trunk1] defaultuser=x...@sip.provider.com fromuser= fromdomain=sip.provider.com type=peer secret=a outboundproxy=10.10.10.10 ;(replaced actual ip) nat=no host=sip.provider.com dtmfmode=auto disallow=all context=from-internal canreinvite=no allow=g729 trustrpid=yes sendrpid=yes register = x...@sip.provider.com:a@10.10.10.10:5060 -- With the above config, I can register with the providers sip proxy, however, the error below is observed in the logs concerning the host when I try to make a call: -- [2012-08-02 23:37:31] WARNING[26155] acl.c: Unable to lookup ' sip.provider.com' [2012-08-02 23:37:31] ERROR[26155] chan_sip.c: srvlookup failed for host: sip.provider.com, on peer trunk1, removing peer -- I have done some research on this issue but not been able to find anything conclusive on why this would happen. I tested the sip details provided with a different sip client (actually an IP phone) and was able to register and send / receive calls with no problem. The problem just seems to be somewhere in my asterisk client configuration or a known bug with the version of asterisk I am using for this. Any pointers? Thanks. Joseph -- _ -- 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] Asterisk 1.6 AEL Macro vs GoSub
Hi,
I have recently run into the problem with macro implementation in AEL in
Asterisk 1.6. I have some older AEL dialplan which runs on 1.4 but it does not
on 1.6 and I'm not sure how to solve this correctly. Let me explain...
For example, in Asterisk 1.4 I have a macro like this:
macro read_digits(digits) {
Set(playlist=${SHELL(${PYTHON} ${SCRIPTS}/read_digits.py ${digits})});
if (${playlist}!=) {
Background(${playlist});
}
}
This macro calls a python script which generates a list of sound files which
are then played back by Background application. So whenever in my Dialplan I
need to read some digits, I simply do:
read_digits(20);
In 1.4 macro is implemented as macro and this is quite nice because I can use
it as follows:
context test {
s = {
read_digits(20);
}
h = {
// do something
}
}
Macro is executed in the original context and ordinary as well as special
extensions are handled by this context. As AEL is not much of a real
programming language and there aren't many possibilities how to make some parts
of code abstract, this was at least something.
But in 1.6 AEL macro has been reimplemented thru GoSub and it is translated
into context. So when the macro is performing it's work there is a need to
catch special extensions and so. The code above won't work because hangup in
read_digits macro is not catched. New macro should look like this:
macro read_digits(digits) {
Set(playlist=${SHELL(${PYTHON} ${SCRIPTS}/read_digits.py ${digits})});
if (${playlist}!=) {
Background(${playlist});
}
catch h {
// do something
}
}
But catching the h extension in the macro doesn't solve my problem as I need
to do different things in the h extension in different contexts. Only
possible workaround that comes to my mind is a copypaste of the code which
practically ruins any advantage of using a macro.
Any thoughts on how to do this in a nice way? Maybe I'm missing something...
Thanks,
Jiri Pokorny
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[asterisk-users] asterisk 1.6 agi problem with PHP
Hello everyone,
I am sure that someone can help with this. We decided to do a fresh install
of asterisk 1.6.2.19
And after we did that, the problem that we have is this - We cant run a
single Php file!
Here's the output:
-- Executing [8212@from-pstn:1] Answer(DAHDI/23-1, ) in new stack
-- Executing [8212@from-pstn:2] AGI(DAHDI/23-1, testera.agi) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/testera.agi
DAHDI/23-1AGI Tx agi_request: testera.agi
DAHDI/23-1AGI Tx agi_channel: DAHDI/23-1
DAHDI/23-1AGI Tx agi_language: en
DAHDI/23-1AGI Tx agi_type: DAHDI
DAHDI/23-1AGI Tx agi_uniqueid: 1310825293.10
DAHDI/23-1AGI Tx agi_version: 1.6.2.19
DAHDI/23-1AGI Tx agi_callerid: 112686649
DAHDI/23-1AGI Tx agi_calleridname: unknown
DAHDI/23-1AGI Tx agi_callingpres: 3
DAHDI/23-1AGI Tx agi_callingani2: 0
DAHDI/23-1AGI Tx agi_callington: 33
DAHDI/23-1AGI Tx agi_callingtns: 0
DAHDI/23-1AGI Tx agi_dnid: 8212
DAHDI/23-1AGI Tx agi_rdnis: unknown
DAHDI/23-1AGI Tx agi_context: from-pstn
DAHDI/23-1AGI Tx agi_extension: 8212
DAHDI/23-1AGI Tx agi_priority: 2
DAHDI/23-1AGI Tx agi_enhanced: 0.0
DAHDI/23-1AGI Tx agi_accountcode:
DAHDI/23-1AGI Tx agi_threadid: -1223132272
DAHDI/23-1AGI Tx
DAHDI/23-1AGI Rx verbose Failed to execute
'/var/lib/asterisk/agi-bin/testera.agi': No such file or directory 1
testera.agi: Failed to execute '/var/lib/asterisk/agi-bin/testera.agi': No
such file or directory
DAHDI/23-1AGI Tx 200 result=1
VERIFIED EVERYTHING:
[root@localhost agi-bin]# ls -l
total 48
-rwxr-xr-x 1 asterisk asterisk 1742 Jul 1 18:57 agi-test.agi
-rwxr-xr-x 1 asterisk asterisk 9909 Jul 1 18:57 eagi-sphinx-test
-rwxr-xr-x 1 asterisk asterisk 8724 Jul 1 18:57 eagi-test
-rwxr-xr-x 1 asterisk asterisk 14530 Jul 1 18:57 jukebox.agi
-rwxr-xr-x 1 asterisk asterisk 1508 Jul 16 16:04 testera.agi
[root@localhost agi-bin]# which php
/usr/bin/php
Here's the agi - simple test that we picked from the net.:
#!/usr/bin/php
?
ob_implicit_flush(false);
set_time_limit(6);
$stdin = fopen('php://stdin', 'r');
$stdlog = fopen('my_agi.log', 'w');
$debug = true;
/* Read input from Asterisk and output via $astOutput */
function astRead()
{
global $stdin, $debug, $stdlog;
$astOutput = str_replace(\n, , fgets($stdin, 4096));
if ($debug) fputs($stdlog, read: $input\n);
return $astOutput ;
}
/* Write AGI command to Asterisk */
function astWrite($agiCommand)
{
global $debug, $stdlog;
if ($debug) fputs($stdlog, write: $agiCommand\n);
echo $agiCommand.\n;
}
/* Handling execution input from Asterisk */
$agivar = array();
while (!feof($stdin))
{
$temp = fgets($stdin);
$temp = str_replace(\n,,$temp);
$s = explode(:,$temp);
$agivar[$s[0]] = trim($s[1]);
if ($temp == )
{
break;
}
}
/* Operational Code starts here */
/* Playback the demo-congrats.gsm file from the
* directory /var/lib/asterisk/sounds/
*/
astWrite(STREAM FILE /var/lib/asterisk/sounds/en/tt-monkeys #);
astRead();
/* Say the number 123456
astWrite(SAY NUMBER 123456 #);
astRead();*/
/* Finalization of AGI script and clean-ups */
fclose ($stdin);
fclose ($stdlog);
exit(0);
?
All help is appreciated.
Thanks,
Z. Zivanovic
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Re: [asterisk-users] asterisk 1.6 agi problem with PHP
On Sat, 16 Jul 2011, Zarko Zivanovic wrote: I am sure that someone can help with this. We decided to do a fresh install of asterisk 1.6.2.19 And after we did that, the problem that we have is this – We cant run a single Php file! testera.agi: Failed to execute '/var/lib/asterisk/agi-bin/testera.agi': No such file or directory If you try to execute the script as the user running the Asterisk binary from the command line, what do you get? For example: sudo -s -u asterisk /var/lib/asterisk/agi-bin/testera.agi /dev/null If that executes, I'd take a peek at the environment variables of the Asterisk process to ensure /usr/bin/ is in the PATH. For example: sudo cat /proc/$(pidof asterisk)/environ\ | tr '\0' '\n'\ | grep PATH Keep in mind, an AGI interfaces with Asterisk via STDIN and STDOUT so you can test an AGI (within obvious limitations) completely outside of Asterisk by redirecting STDIN and STDOUT. For example, given a file testera.stdin containing: agi_request: testera.agi agi_channel: DAHDI/23-1 agi_language: en agi_type: DAHDI agi_uniqueid: 1310825293.10 agi_version: 1.6.2.19 agi_callerid: 112686649 agi_calleridname: unknown agi_callingpres: 3 agi_callingani2: 0 agi_callington: 33 agi_callingtns: 0 agi_dnid: 8212 agi_rdnis: unknown agi_context: from-pstn agi_extension: 8212 agi_priority: 2 agi_enhanced: 0.0 agi_accountcode: agi_threadid: -1223132272 200 result=0 200 result=0 200 result=0 You can execute the AGI like: /var/lib/asterisk/agi-bin/testera.agi testera.stdin and your script should display: STREAM FILE /var/lib/asterisk/sounds/en/tt-monkeys # which, obviously, will not succeed because your AGI is 'talking' to your shell, not Asterisk. -- 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
[asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog telephone line coming on Wildcard TDM400P REV E/F Board 5 I can't get asterisk to dectect call coming from analog line. Here is my /etc/dahdi/system.conf fxsks=1 # global data loadzone = us defaultzone = us /etc/asterisk/chan_dahdi.conf [channels] language=en context=my-phones switchtype=national signalling=fxs_ks channel = 1 /etc/asterisk/extensions.conf [globals] CONSOLE=DAHDI/1 TRUNK=DAHDI/4 TRUNKMSD=1 [my-phone] exten = 2000,1,Dial(DAHDI/1/116) exten = 2000,2,cONGESTION exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline exten = 2001,2,HangUp() exten = 1001,1,Dial(DAHDI/1/7608514114) exten = 1001,2,HangUp() exten = ,1,Dial(DAHDI/1/7608514114) exten = l111,2,HangUp() /etc/asterisk/sip.conf [general] port = 5060 context = others [2000] type=friend context=my-phones secret=1234 host=dynamic [2001] type=friend context=my-phones secret=1234 host=dynamic [1001] type=friend context=my-phones secret=1234 [] type=friend context=my-phones secret=1234 [phonesys] type=friend username=user1 secret=1234 host=dynamic context=my-phones Any suggestions are welcome. Thanks, motty -- _ -- 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] Asterisk 1.6 Dahdi on Centos 5.2
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog telephone line coming on Wildcard TDM400P REV E/F Board 5 I can't get asterisk to dectect call coming from analog line. Here is my /etc/dahdi/system.conf fxsks=1 # global data loadzone = us defaultzone = us /etc/asterisk/chan_dahdi.conf [channels] language=en context=my-phones switchtype=national signalling=fxs_ks channel = 1 /etc/asterisk/extensions.conf [globals] CONSOLE=DAHDI/1 TRUNK=DAHDI/4 TRUNKMSD=1 [my-phone] exten = 2000,1,Dial(DAHDI/1/116) exten = 2000,2,cONGESTION exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline exten = 2001,2,HangUp() exten = 1001,1,Dial(DAHDI/1/7608514114) exten = 1001,2,HangUp() exten = ,1,Dial(DAHDI/1/7608514114) exten = l111,2,HangUp() The context in chan_dahdi.conf is my-phones which differs from the my-phone context in extensions.conf. Richard -- _ -- 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] Asterisk 1.6 Dahdi on Centos 5.2
A couple things - First, in extensions.con your context is [my-phone], but you're using my-phones in your dahdi and sip.conf files. Second, you need an 's' extension somewhere in your receiving context in order for asterisk to answer the incoming analog call. Third, I think you've got some issues with your Dial statements, but I'm on my phone right now and can't really diagnose them. I'll take a look later when I'm back at a desk, if no one else has commented by then. Thanks, --Warren Selby, dCAP On Jun 28, 2011, at 12:30 PM, motty.cruz motty.c...@gmail.com wrote: Hello, I have Asterisk 1.6 running on Centos, Also I have one analog telephone line coming on Wildcard TDM400P REV E/F Board 5 I can't get asterisk to dectect call coming from analog line. Here is my /etc/dahdi/system.conf fxsks=1 # global data loadzone = us defaultzone = us /etc/asterisk/chan_dahdi.conf [channels] language=en context=my-phones switchtype=national signalling=fxs_ks channel = 1 /etc/asterisk/extensions.conf [globals] CONSOLE=DAHDI/1 TRUNK=DAHDI/4 TRUNKMSD=1 [my-phone] exten = 2000,1,Dial(DAHDI/1/116) exten = 2000,2,cONGESTION exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline exten = 2001,2,HangUp() exten = 1001,1,Dial(DAHDI/1/7608514114) exten = 1001,2,HangUp() exten = ,1,Dial(DAHDI/1/7608514114) exten = l111,2,HangUp() /etc/asterisk/sip.conf [general] port = 5060 context = others [2000] type=friend context=my-phones secret=1234 host=dynamic [2001] type=friend context=my-phones secret=1234 host=dynamic [1001] type=friend context=my-phones secret=1234 [] type=friend context=my-phones secret=1234 [phonesys] type=friend username=user1 secret=1234 host=dynamic context=my-phones Any suggestions are welcome. Thanks, motty -- _ -- 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] Asterisk 1.6 Dahdi on Centos 5.2
My mistake I had fix that typo but no luck Thanks, motty -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett Sent: Tuesday, June 28, 2011 10:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2 Hello, I have Asterisk 1.6 running on Centos, Also I have one analog telephone line coming on Wildcard TDM400P REV E/F Board 5 I can't get asterisk to dectect call coming from analog line. Here is my /etc/dahdi/system.conf fxsks=1 # global data loadzone = us defaultzone = us /etc/asterisk/chan_dahdi.conf [channels] language=en context=my-phones switchtype=national signalling=fxs_ks channel = 1 /etc/asterisk/extensions.conf [globals] CONSOLE=DAHDI/1 TRUNK=DAHDI/4 TRUNKMSD=1 [my-phone] exten = 2000,1,Dial(DAHDI/1/116) exten = 2000,2,cONGESTION exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline exten = 2001,2,HangUp() exten = 1001,1,Dial(DAHDI/1/7608514114) exten = 1001,2,HangUp() exten = ,1,Dial(DAHDI/1/7608514114) exten = l111,2,HangUp() The context in chan_dahdi.conf is my-phones which differs from the my-phone context in extensions.conf. Richard -- _ -- 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 - No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1388 / Virus Database: 1516/3731 - Release Date: 06/28/11 -- _ -- 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] Asterisk 1.6 Dahdi on Centos 5.2
Thanks Warren,
I have gone ahead and correct my typo. Also, I created 's' extension as you
suggested.
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERIDNAME})
exten = s,n,Wait(4)
exten = s,n,Playback(tt-easels)
exten = s,n,Voicemail(@vm-test)
exten = s,n,Wait(2)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Wait(2)
exten = s,n,HangUp()
I actually followed this e.i
http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html
because I have the same Digium card tdm4oop four modules although I'm only
using one.
Thanks, in advance.
-motty
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Tuesday, June 28, 2011 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2
A couple things -
First, in extensions.con your context is [my-phone], but you're using
my-phones in your dahdi and sip.conf files.
Second, you need an 's' extension somewhere in your receiving context in
order for asterisk to answer the incoming analog call.
Third, I think you've got some issues with your Dial statements, but I'm on
my phone right now and can't really diagnose them. I'll take a look later
when I'm back at a desk, if no one else has commented by then.
Thanks,
--Warren Selby, dCAP
On Jun 28, 2011, at 12:30 PM, motty.cruz motty.c...@gmail.com wrote:
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi/system.conf
fxsks=1
# global data
loadzone = us
defaultzone = us
/etc/asterisk/chan_dahdi.conf
[channels]
language=en
context=my-phones
switchtype=national
signalling=fxs_ks
channel = 1
/etc/asterisk/extensions.conf
[globals]
CONSOLE=DAHDI/1
TRUNK=DAHDI/4
TRUNKMSD=1
[my-phone]
exten = 2000,1,Dial(DAHDI/1/116)
exten = 2000,2,cONGESTION
exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline
exten = 2001,2,HangUp()
exten = 1001,1,Dial(DAHDI/1/7608514114)
exten = 1001,2,HangUp()
exten = ,1,Dial(DAHDI/1/7608514114)
exten = l111,2,HangUp()
/etc/asterisk/sip.conf
[general]
port = 5060
context = others
[2000]
type=friend
context=my-phones
secret=1234
host=dynamic
[2001]
type=friend
context=my-phones
secret=1234
host=dynamic
[1001]
type=friend
context=my-phones
secret=1234
[]
type=friend
context=my-phones
secret=1234
[phonesys]
type=friend
username=user1
secret=1234
host=dynamic
context=my-phones
Any suggestions are welcome.
Thanks,
motty
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-
No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1388 / Virus Database: 1516/3731 - Release Date: 06/28/11
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[asterisk-users] Asterisk 1.6 - subscriptions.
Hi all, I try to figure out why I have empty : sip show subscriptions list in may asterisk 1.6. When device is registering to asterisk I can see in log: NOTICE[25603]: chan_sip.c:21518 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1010 but sip show subscriptions is just empty. May it be the problem because devices are registering to asterisk from behind NAT? Regards, Jarek -- _ -- 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] asterisk 1.6 - 511 Command not permitted causing high CPU usage
http://pastebin.com/vxGM2n5j We are getting those errors 100x per second in console when AGI set debug is on It is causing extremely high CPU usage, we've tried asterisk version 1.6.1.22 and 1.6.2.18 It seems the problem is worse in 1.6.2.18 Can someone advise how to fix this? Thank you. -- _ -- 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] asterisk 1.6 - 511 Command not permitted causing high CPU usage
On Sun, 5 Jun 2011, vip killa wrote: http://pastebin.com/vxGM2n5j We are getting those errors 100x per second in console when AGI set debug is on Can someone advise how to fix this? Thank you. Don't request 'WAIT FOR DIGIT 1000' from a dead channel. Don't ignore the error from 'WAIT FOR DIGIT 1000' Don't loop on the error. -- 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
[asterisk-users] Asterisk 1.6: Custom Name for Recordings file
Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -- _ -- 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] Asterisk 1.6: Custom Name for Recordings file
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
See the fname_base information below.
pbx*CLI core show application monitor
-= Info about application 'Monitor' =-
[Synopsis]
Monitor a channel.
[Description]
Used to start monitoring a channel. The channel's input and output voice
packets are logged to files until the channel hangs up or monitoring is stopped
by the StopMonitor application.
By default, files are stored to /var/spool/asterisk/monitor/. Returns
'-1' if monitor files can't be opened or if the channel is already monitored,
otherwise '0'.
[Syntax]
Monitor([file_format[:urlbase]][,fname_base[,options]])
[Arguments]
file_format
optional, if not set, defaults to 'wav'
fname_base
if set, changes the filename used to the one specified.
options
m: when the recording ends mix the two leg files into one and delete
the two leg files. If the variable ${MONITOR_EXEC} is set, the application
referenced in it will be executed instead of soxmix/sox and the raw leg
files will NOT be deleted automatically. soxmix/sox or ${MONITOR_EXEC}
is handed 3 arguments, the two leg files and a target mixed file name
which is the same as the leg file names only without the in/out designator.
If ${MONITOR_EXEC_ARGS} is set, the contents will be passed on as
additional arguments to ${MONITOR_EXEC}. Both ${MONITOR_EXEC} and the
Mix flag can be set from the administrator interface.
b: Don't begin recording unless a call is bridged to another channel.
i: Skip recording of input stream (disables 'm' option).
o: Skip recording of output stream (disables 'm' option).
[See Also]
StopMonitor()
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Re: [asterisk-users] Asterisk 1.6: Custom Name for Recordings file
On Fri, May 13, 2011 at 11:07 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
See the fname_base information below.
pbx*CLI core show application monitor
-= Info about application 'Monitor' =-
[Synopsis]
Monitor a channel.
[Description]
Used to start monitoring a channel. The channel's input and output voice
packets are logged to files until the channel hangs up or monitoring is
stopped
by the StopMonitor application.
By default, files are stored to /var/spool/asterisk/monitor/. Returns
'-1' if monitor files can't be opened or if the channel is already
monitored,
otherwise '0'.
[Syntax]
Monitor([file_format[:urlbase]][,fname_base[,options]])
[Arguments]
file_format
optional, if not set, defaults to 'wav'
fname_base
if set, changes the filename used to the one specified.
options
m: when the recording ends mix the two leg files into one and delete
the two leg files. If the variable ${MONITOR_EXEC} is set, the
application
referenced in it will be executed instead of soxmix/sox and the raw leg
files will NOT be deleted automatically. soxmix/sox or ${MONITOR_EXEC}
is handed 3 arguments, the two leg files and a target mixed file name
which is the same as the leg file names only without the in/out
designator.
If ${MONITOR_EXEC_ARGS} is set, the contents will be passed on as
additional arguments to ${MONITOR_EXEC}. Both ${MONITOR_EXEC} and the
Mix flag can be set from the administrator interface.
b: Don't begin recording unless a call is bridged to another channel.
i: Skip recording of input stream (disables 'm' option).
o: Skip recording of output stream (disables 'm' option).
[See Also]
StopMonitor()
Thx Eric.
I read the link e1*CLI core show application monitor but I could not
follow what I should do to customize the file name of the recording.
I guess some changes to the dialplan is required ?
Thx
S
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Re: [asterisk-users] Asterisk 1.6: Custom Name for Recordings file
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
On Fri, May 13, 2011 at 11:07 PM, Eric Wieling
ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
See the fname_base information below.
pbx*CLI core show application monitor
-= Info about application 'Monitor' =-
[Synopsis]
Monitor a channel.
[Description]
Used to start monitoring a channel. The channel's input
and output voice
packets are logged to files until the channel hangs up
or monitoring is stopped
by the StopMonitor application.
By default, files are stored to
/var/spool/asterisk/monitor/. Returns
'-1' if monitor files can't be opened or if the channel
is already monitored,
otherwise '0'.
[Syntax]
Monitor([file_format[:urlbase]][,fname_base[,options]])
[Arguments]
file_format
optional, if not set, defaults to 'wav'
fname_base
if set, changes the filename used to the one specified.
options
m: when the recording ends mix the two leg files
into one and delete
the two leg files. If the variable ${MONITOR_EXEC}
is set, the application
referenced in it will be executed instead of
soxmix/sox and the raw leg
files will NOT be deleted automatically. soxmix/sox
or ${MONITOR_EXEC}
is handed 3 arguments, the two leg files and a
target mixed file name
which is the same as the leg file names only without
the in/out designator.
If ${MONITOR_EXEC_ARGS} is set, the contents will be
passed on as
additional arguments to ${MONITOR_EXEC}. Both
${MONITOR_EXEC} and the
Mix flag can be set from the administrator interface.
b: Don't begin recording unless a call is bridged to
another channel.
i: Skip recording of input stream (disables 'm' option).
o: Skip recording of output stream (disables 'm' option).
[See Also]
StopMonitor()
Thx Eric.
I read the link e1*CLI core show application monitor but I
could not follow what I should do to customize the file name
of the recording.
I guess some changes to the dialplan is required ?
Re-read your message, and realized you are asking about a GUI for Asterisk.
Sorry, I can't help you with that.
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asterisk-users mailing list
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Re: [asterisk-users] Asterisk 1.6: Custom Name for Recordings file
On 5/13/11 10:57 AM, RSCL Mumbai wrote:
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
[snip..]
Thx Eric.
I read the link e1*CLI core show application monitor but I could not follow
what I should do to customize the file name of the recording.
I guess some changes to the dialplan is required ?
try something like:
Monitor(wav,${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${EXTEN})
--
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Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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[asterisk-users] Asterisk 1.6 Questions
I have a couple of questions about asterisk 1.6: Can you change codecs mid-call upon re-invite? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Thanks in advance for any insight. Gary -- _ -- 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] Asterisk 1.6 Questions
On 05/03/2011 12:43 PM, Gary Graves wrote: Can you change codecs mid-call upon re-invite? Do you mean: can Asterisk be configured to _initiate_ such a change at some point, mid-call? Or do you mean: Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Doesn't seem to. Looking at chan_sip.c in 1.6.2.13, there is no call to add_sdp() that is not made either in the context of 1) an initial INVITE request or 2) a re-INVITE or 3) the construction of a response. Nothing in the case of the production of an end-to-end ACK. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] Asterisk 1.6 Questions
Can you answer both? Can Asterisk be configured to _initiate_ such a change at some point, mid-call? and Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? On Tue, May 3, 2011 at 12:56 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/03/2011 12:43 PM, Gary Graves wrote: Can you change codecs mid-call upon re-invite? Do you mean: can Asterisk be configured to _initiate_ such a change at some point, mid-call? Or do you mean: Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Doesn't seem to. Looking at chan_sip.c in 1.6.2.13, there is no call to add_sdp() that is not made either in the context of 1) an initial INVITE request or 2) a re-INVITE or 3) the construction of a response. Nothing in the case of the production of an end-to-end ACK. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] Asterisk 1.6 Questions
On 05/03/2011 01:16 PM, Gary Graves wrote: Can you answer both? Can Asterisk be configured to _initiate_ such a change at some point, mid-call? I don't know of a way to do that. I suppose it might be possible if a call were asynchronously transferred to a SIP peer that had different codec requirements. and Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? It should. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] Asterisk 1.6 = No sound/voice when i redirect the call
Hi
very thanks, that's work
bye
olivier
2011/4/3 Mark Murawski markm-li...@intellasoft.net:
I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten = s,1,Playback(silence/1)
On 04/03/11 14:14, Olivier CALVANO wrote:
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawskimarkm-li...@intellasoft.net:
In that situation, I've had to do a pickup macro that kind of primes
the
audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the
callee's
channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten}
])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten =
_00339,7,Set(CALLERPRES()=prohib_not_screened)
exten =
_00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct,
asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
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[asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten = _00339,7,Set(CALLERPRES()=prohib_not_screened)
exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct, asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
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Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
In that situation, I've had to do a pickup macro that kind of primes
the audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the
callee's channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten = _00339,7,Set(CALLERPRES()=prohib_not_screened)
exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct, asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
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Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawski markm-li...@intellasoft.net:
In that situation, I've had to do a pickup macro that kind of primes the
audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the callee's
channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten = _00339,7,Set(CALLERPRES()=prohib_not_screened)
exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct,
asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
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Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten = s,1,Playback(silence/1)
On 04/03/11 14:14, Olivier CALVANO wrote:
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawskimarkm-li...@intellasoft.net:
In that situation, I've had to do a pickup macro that kind of primes the
audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the callee's
channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten =_00339,1,Set(foo=${SIP_HEADER(To)})
exten =_00339,2,Set(cut1=${CUT(foo,:,2)})
exten =_00339,3,Set(CLI=${CUT(cut1,,1)})
exten =_00339,4,Set(toexten=${CUT(CLI,@,1)})
exten =_00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten =_00339,6,AGI(Ddi-Network.agi,${toexten})
exten =_00339,7,Set(CALLERPRES()=prohib_not_screened)
exten =_00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten =_00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct,
asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
--
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[asterisk-users] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers = mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 And here's the error messages I get: voip2*CLI realtime mysql status localhost configured for mya2billing@localhost, port 3306 with username a2billinguser. mya2billing configured for mya2billing@localhost, port 3306 with username a2billinguser. [Mar 7 14:38:59] ERROR[15943]: res_config_mysql.c:1575 mysql_reconnect: MySQL RealTime: Failed to connect database server mya2billing on localhost (err 2002). Check debug for more info. [Mar 7 14:38:59] ERROR[15943]: res_config_mysql.c:1575 mysql_reconnect: MySQL RealTime: Failed to connect database server mya2billing on localhost (err 2002). Check debug for more info. This doesn't make any sense. res_mysql.conf contains working mysql credentials that I can verify with running mysql from the command line. -- _ -- 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] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
On Mon, Mar 7, 2011 at 5:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote: Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers = mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: Don't know what res_mysql.conf is, I think it should be res_config_mysql.conf? Sorry it's been a LONG time since I configured/used realtime and that also was with ODBC and TDS but I know that the file res_config_mysql.conf should definitely be there HTH \R -- _ -- 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] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config stanza and see if that helps (or whatever is the actual location of your mysql.sock file). -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config stanza and see if that helps (or whatever is the actual location of your mysql.sock file). Hmm. This appears to have fixed the problem, even though I swear I've done this already. (and for anyone reading this, on Debian the file is mysqld.sock) -- _ -- 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] Asterisk 1.6 and windows RTC
Hello folks, for a customer of us we are trying to make asterisk and windows RTC library work together, but without success. We *need* to use gsm codec, so in the peer section we have disallow=all allow=gsm the sip signaling is ok, and when sniffing we got this session description: INVITE from windows RTC: v=0. o=- 0 0 IN IP4 172.31.9.130. s=session. c=IN IP4 172.31.9.130. b=CT:1000. t=0 0. m=audio 4632 RTP/AVP 97 111 112 6 0 8 4 5 3 101. k=base64:ftJemQZ2pTDV5gzzqxG6ps5Ol5qiOt2qbP9L9Or5JQg. a=rtpmap:97 red/8000. a=rtpmap:111 SIREN/16000. a=fmtp:111 bitrate=16000. a=rtpmap:112 G7221/16000. a=fmtp:112 bitrate=24000. a=rtpmap:6 DVI4/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=encryption:optional. a=direction:active. OK from asterisk 1.6 PBX: v=0. o=PBX 1705093286 1705093286 IN IP4 172.31.9.251. s=PBX. c=IN IP4 172.31.9.251. t=0 0. m=audio 14962 RTP/AVP 3 101. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. so, the rtp session should be GSM. But the audio does not work. In asterisk logs I see 'Got Siren7 offer at 24000 bps but only 32000 bps supported'. any hint? anyone with the same issue? unfortunately GSM is mandatory for us (we could not use alaw/ulaw, that seems working). thanks so much stefano -- _ -- 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] Asterisk 1.6 SSH Console Colors Debian Lenny
2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote: On Thursday 20 Jan 2011, JR Richardson wrote: Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I'm running Debian but have been running Asterisk since before there was a proper Debian package, and so I ended up writing my own init.d script. See attached. No guarantees or anything :) A number of things I did not like about it: 1. I don't trust safe_asterisk to properly handle being run twice and such. 2. Likewise with daemonization. safe_asterisk is still at the console. 3. You run asterisk as root. And use /var/run/asterisk.pid . Please use a non-root user and /var/run/asterisk/asterisk.pid . 4. On 'restart' you do nothing if the process was not running. That's not the standard semantics. 5. Even if a pid file exists, it does not mean that the process listed in it is your process. In short: A. Don't re-invent start-stop-daemon. B. Let's just move to upstart/systemd so there won't be a need for this stupid guardian safe asterisk. All these reasons seem fine for me. So the remaining question is how can we still get colors with ssh console ?. Is it compliant with start-stop-daemon, for instance ? Cheers -- _ -- 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] Asterisk 1.6 SSH Console Colors Debian Lenny
On Monday 24 January 2011 04:09:31 Olivier wrote: 2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote: On Thursday 20 Jan 2011, JR Richardson wrote: Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. In short: A. Don't re-invent start-stop-daemon. B. Let's just move to upstart/systemd so there won't be a need for this stupid guardian safe asterisk. All these reasons seem fine for me. So the remaining question is how can we still get colors with ssh console ?. Is it compliant with start-stop-daemon, for instance ? Why not just use the start script included with Asterisk? I solved this exact problem a while back, so unless somebody has broken the script since, it should still be working. -- Tilghman -- _ -- 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] Asterisk 1.6 SSH Console Colors Debian Lenny
On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote: On Thursday 20 Jan 2011, JR Richardson wrote: Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I'm running Debian but have been running Asterisk since before there was a proper Debian package, and so I ended up writing my own init.d script. See attached. No guarantees or anything :) A number of things I did not like about it: 1. I don't trust safe_asterisk to properly handle being run twice and such. 2. Likewise with daemonization. safe_asterisk is still at the console. 3. You run asterisk as root. And use /var/run/asterisk.pid . Please use a non-root user and /var/run/asterisk/asterisk.pid . 4. On 'restart' you do nothing if the process was not running. That's not the standard semantics. 5. Even if a pid file exists, it does not mean that the process listed in it is your process. In short: A. Don't re-invent start-stop-daemon. B. Let's just move to upstart/systemd so there won't be a need for this stupid guardian safe asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Asterisk 1.6 SSH Console Colors Debian Lenny
Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I found this in the init script: #snip-# # Mon Jun 04 2007 Iñaki Baz Castillo i...@in.ilimit.es # - Eliminated SAFE_ASTERISK since it doesn't work as LSB script (it could require a independent safe_asterisk init script). # If you DON'T want Asterisk to start up with terminal colors, comment # this out. COLOR=yes #snop# Commenting out COLOR=yes has no effect. The work around is to use the * 1.4 init script which does call safe_asterisk daemon and things seem to work as expected with the colors. So my question is, will this impact the stability of the system in reference to debian lenny using LSB scripts vs the older init scripts? Or is there another work around to get ssh console colors using the Debian * 1.6.0.28 LSB init script? I also tried 'nocolor = no' in the [options] section of asterisk.conf with no effect. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Asterisk 1.6 SSH Console Colors Debian Lenny
On 20 January 2011 18:01, JR Richardson jmr.richard...@gmail.com wrote: Or is there another work around to get ssh console colors using the Debian * 1.6.0.28 LSB init script? I also tried 'nocolor = no' in the [options] section of asterisk.conf with no effect. Try running asterisk using safe_asterisk.. Works for me with 1.4.22 and lenny.. -- _ -- 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] Asterisk 1.6 SSH Console Colors Debian Lenny
On Thursday 20 Jan 2011, JR Richardson wrote: Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I'm running Debian but have been running Asterisk since before there was a proper Debian package, and so I ended up writing my own init.d script. See attached. No guarantees or anything :) -- AJS Answers come *after* questions. asterisk Description: application/shellscript -- _ -- 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] Asterisk 1.6 iax auth rsa failed with policie not found
Hi, I had 2 Asterisk servers connected together in iax with auth=rsa and proper keys for user and peer in each direction. It worked well till I upgraded one of them to Asterisk 1.6.13 Since I get No authority found I thought that problem came from keys as the server with 1.6.13 was changed in the mean time, so I regenerated both keys on each server and copy the public of each one to the other: problem stays. What am I missing? What changes in 1.6 where made concerning this matter? Thanks for any hint. -- Daniel -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote: On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. I know what the issue is. Please open a report on https://issues.asterisk.org and I'll get a patch uploaded pronto. Please let us know the issue number once raised - I'd like to follow this one. Regards, Steve -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
On Wednesday 22 December 2010 08:23:19 MrHanMan wrote: On Wed, Dec 22, 2010 at 6:41 AM, Steve Davies davies...@gmail.com wrote: On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote: On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. I know what the issue is. Please open a report on https://issues.asterisk.org and I'll get a patch uploaded pronto. Please let us know the issue number once raised - I'd like to follow this one. I happened to see it pop up on the bug tracker. Issue #0018515. Very funny error message in the patch. It's a forward-port of a section of code that was in res_agi in 1.4. It was no longer needed in res_agi because AGIs can now continue to interact with Asterisk after a hangup event, transitioning gracefully into DeadAGI. -- Tilghman -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /var/log/syslog? -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /var/log/syslog? Not much. In /var/log/asterisk/messages here's a lot of lines like this: [Dec 17 19:10:13] NOTICE[25518] chan_sip.c: Registration from 'sip:xx...@voip.lightspeed.ca' failed for 'XX.XXX.X.XXX' - No matching peer found And /var/log/syslog has all the normal output from a2billing.php and making calls complete and such. The other funny thing is that except for the massive number of zombie processes, calls are being made and completed just fine. Even voice quality is quite high. -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
Your server is being brute-forced. Read this article (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) and set up fail2ban on your machine right now.Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brasil+55 54 2104-7000Information Security ManagerCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brazil+55 54 2104-7000 Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /var/log/syslog?Not much. In /var/log/asterisk/messages here's a lot of lines like this:[Dec 17 19:10:13] NOTICE[25518] chan_sip.c: Registration from'sip:xx...@voip.lightspeed.ca' failed for 'XX.XXX.X.XXX' - No matchingpeer foundAnd /var/log/syslog has all the normal output from a2billing.php andmaking calls complete and such.The other funny thing is that except for the massive number of zombieprocesses, calls are being made and completed just fine. Even voicequality is quite high.--_-- 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/helloasterisk-users mailing listTo 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] Asterisk 1.6 produces *many* zombie processes on Debian.
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. Simple In sip.conf please set alwaysauthreject = yes ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
Actually, no. This is part of a migration, and those are mostly customers' secondary lines (which for the most part, aren't even active). We get a lot of these bad logins because the retry times on the ATAs are quite short. Asterisk really *shouldn't* leave zombies around for every bad login, but if it does, then I suppose cleaning up these missing accounts might fix it. Your server is being brute-forced. Read this article (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) and set up fail2ban on your machine right now. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Mensagem original - Am 20.12.2010 21:39, schrieb Ernie Dunbar: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. What does /var/log/asterisk/messages say? And /var/log/syslog? Not much. In /var/log/asterisk/messages here's a lot of lines like this: [Dec 17 19:10:13] NOTICE[25518] chan_sip.c: Registration from 'sip:xx...@voip.lightspeed.ca' failed for 'XX.XXX.X.XXX' - No matching peer found And /var/log/syslog has all the normal output from a2billing.php and making calls complete and such. The other funny thing is that except for the massive number of zombie processes, calls are being made and completed just fine. Even voice quality is quite high. -- _ -- 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 -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. I know what the issue is. Please open a report on https://issues.asterisk.org and I'll get a patch uploaded pronto. -- Tilghman -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. Simple In sip.conf please set alwaysauthreject = yes Thanks for the tip, but we already did that a while ago. :) -- _ -- 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] Asterisk 1.6 produces *many* zombie processes on Debian.
We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. We are using the following software: Debian Lenny 5.0 Asterisk 1.6.2.15 `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2 Libpri 1.4.11.4 A2Billing is also installed on this server, if that matters at all. Any help with this issue, including help in troubleshooting the cause, is highly appreciated. -- _ -- 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] Asterisk 1.6 (Web-meetme)
Hi, I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6 Any conference we create in web-meetme never expires. I am not sure if i am missing while configuring, though i didnt find anything in lib/define.php also checked in asterisk that can point me, which can help me in fixing this issue. Can someone please help me in fixing this. -- Thanks Regards Manmohan Singh -- _ -- 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] Asterisk 1.6 (Web-meetme)
Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6 Any conference we create in web-meetme never expires. I am not sure if i am missing while configuring, though i didnt find anything in lib/define.php also checked in asterisk that can point me, which can help me in fixing this issue. Since you can join the conference you created with WMM, the Realtime settings are likely correct. You do not mention which version of 1.6 you are on, so I would guess that you are on 1.6.2.7 or older. For a variety of reasons the realtime feature, in particular the scheduling code, was added and tweaked over a wide range of 1.6 releases. The first one I would consider feature complete for use with Web-MeetMe is 1.6.2.7, and even that version has a trivial, but nasty bug that causes segfaults (fixed in a recent 1.6.2 release) Dan -- _ -- 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] Asterisk 1.6 (Web-meetme)
Hi Dan, I am currently on Asterisk 1.6.2.14. Thanks Regards Manmohan Singh On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6 Any conference we create in web-meetme never expires. I am not sure if i am missing while configuring, though i didnt find anything in lib/define.php also checked in asterisk that can point me, which can help me in fixing this issue. Since you can join the conference you created with WMM, the Realtime settings are likely correct. You do not mention which version of 1.6 you are on, so I would guess that you are on 1.6.2.7 or older. For a variety of reasons the realtime feature, in particular the scheduling code, was added and tweaked over a wide range of 1.6 releases. The first one I would consider feature complete for use with Web-MeetMe is 1.6.2.7, and even that version has a trivial, but nasty bug that causes segfaults (fixed in a recent 1.6.2 release) Dan -- _ -- 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 -- Thanks Regards Manmohan Singh -- _ -- 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] Asterisk 1.6 (Web-meetme)
Manmohan wrote: I am currently on Asterisk 1.6.2.14. Do you have schedule=yes in meetme.conf? I incorrectly remembered/thought that all of the Realtime features were controlled by that option, only a small number, such as end times and CDR logging On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6 Any conference we create in web-meetme never expires. I am not sure if i am missing while configuring, though i didnt find anything in lib/define.php also checked in asterisk that can point me, which can help me in fixing this issue. Since you can join the conference you created with WMM, the Realtime settings are likely correct. You do not mention which version of 1.6 you are on, so I would guess that you are on 1.6.2.7 or older. For a variety of reasons the realtime feature, in particular the scheduling code, was added and tweaked over a wide range of 1.6 releases. The first one I would consider feature complete for use with Web-MeetMe is 1.6.2.7, and even that version has a trivial, but nasty bug that causes segfaults (fixed in a recent 1.6.2 release) Dan -- _ -- 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] Asterisk 1.6 (Web-meetme)
Hi Dan, In meetme.conf the schedule=yes was commented, after removing its working fine. But one strange thing had started now. I started getting segmentation fault. following are the errors which i see in it: warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/libodbcinst.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/libogg.so.0 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/libbluetooth.so.2 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations Reading symbols from /lib/libssl.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libssl.so.6 Reading symbols from /lib/libcrypto.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libcrypto.so.6 Reading symbols from /lib/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libc.so.6 Reading symbols from /usr/lib/libxml2.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libxml2.so.2 Reading symbols from /usr/lib/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libz.so.1 Thanks Regards Manmohan Singh On Sat, Dec 4, 2010 at 1:12 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am currently on Asterisk 1.6.2.14. Do you have schedule=yes in meetme.conf? I incorrectly remembered/thought that all of the Realtime features were controlled by that option, only a small number, such as end times and CDR logging On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6 Any conference we create in web-meetme never expires. I am not sure if i am missing while configuring, though i didnt find anything in lib/define.php also checked in asterisk that can point me, which can help me in fixing this issue. Since you can join the conference you created with WMM, the Realtime settings are likely correct. You do not mention which version of 1.6 you are on, so I would guess that you are on 1.6.2.7 or older. For a variety of reasons the realtime feature, in particular the scheduling code, was added and tweaked over a wide range of 1.6 releases. The first one I would consider feature complete for use with Web-MeetMe is 1.6.2.7, and even that version has a trivial, but nasty bug that causes segfaults (fixed in a recent 1.6.2 release) Dan -- _ -- 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 -- Thanks Regards Manmohan Singh -- _ -- 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] Asterisk 1.6 (Web-meetme)
The errors you posted do not point to a the problem. Did you build from source or are you using packages? If from source, grep for useropts in app_meetme.c and The second instance should be: char useropts[OPTIONS_LEN + 1] = ; If the line does not have the = , then the issue is that the bug I mentioned is still present. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Manmohan Singh Jandu Sent: Friday, December 03, 2010 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 (Web-meetme) Hi Dan, In meetme.conf the schedule=yes was commented, after removing its working fine. But one strange thing had started now. I started getting segmentation fault. following are the errors which i see in it: warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/libodbcinst.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/libogg.so.0 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/libbluetooth.so.2 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations Reading symbols from /lib/libssl.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libssl.so.6 Reading symbols from /lib/libcrypto.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libcrypto.so.6 Reading symbols from /lib/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libc.so.6 Reading symbols from /usr/lib/libxml2.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libxml2.so.2 Reading symbols from /usr/lib/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libz.so.1 Thanks Regards Manmohan Singh On Sat, Dec 4, 2010 at 1:12 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am currently on Asterisk 1.6.2.14. Do you have schedule=yes in meetme.conf? I incorrectly remembered/thought that all of the Realtime features were controlled by that option, only a small number, such as end times and CDR logging On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6 Any conference we create in web-meetme never expires. I am not sure if i am missing while configuring, though i didnt find anything in lib/define.php also checked in asterisk that can point me, which can help me in fixing this issue. Since you can join the conference you created with WMM, the Realtime settings are likely correct. You do not mention which version of 1.6 you are on, so I would guess that you are on 1.6.2.7 or older. For a variety of reasons the realtime feature, in particular the scheduling code, was added and tweaked over a wide range of 1.6 releases. The first one I would consider feature complete for use with Web-MeetMe is 1.6.2.7, and even that version has a trivial, but nasty bug that causes segfaults (fixed in a recent 1.6.2 release) Dan -- _ -- 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 -- Thanks Regards Manmohan Singh -- _ -- 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] Asterisk 1.6 and Username in Dial
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
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Re: [asterisk-users] Asterisk 1.6 and Username in Dial
No you can't
On Wed, Nov 24, 2010 at 2:34 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
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[asterisk-users] Asterisk 1.6 and Music on Hold
Hi i have a small problems on Asterisk 1.6 with the MusiconOld : musiconhold.conf: [Sound_1] mode=quietmp3 directory=/var/lib/asterisk/moh/Sound_1 in extensions.conf: exten = 0532xx,1,Answer exten = 0532xx,2,MusicOnHold(Sound_1) exten = 0532xx,3,Dial(SIP/ACCOUNT001,180,t) exten = 0532xx,4,Hangup When i call to the number, i have the Music Sound_1 but the SIP Phone don't ring ... Where is my error ? and second question, can i said at asterisk that when he receive the call, he play the music from first second ? and repeat at the end of the music. Thanks for your help -- _ -- 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] Asterisk 1.6 and Music on Hold
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote: Hi i have a small problems on Asterisk 1.6 with the MusiconOld : musiconhold.conf: [Sound_1] mode=quietmp3 directory=/var/lib/asterisk/moh/Sound_1 in extensions.conf: exten = 0532xx,1,Answer exten = 0532xx,2,MusicOnHold(Sound_1) exten = 0532xx,3,Dial(SIP/ACCOUNT001,180,t) exten = 0532xx,4,Hangup When i call to the number, i have the Music Sound_1 but the SIP Phone don't ring ... Where is my error ? and second question, can i said at asterisk that when he receive the call, he play the music from first second ? and repeat at the end of the music. Thanks for your help -- _ -- 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 First, if you don't use the Music on hold command prior to the dial, do you hear ringing? It seems to me that what's going on here is that you're overriding the progress notification that results from the device responding to the invite with TRYING or RINGING by running MOH. If the ringing doesn't occur even when you remove the MOH command, your device is probably not signaling properly and you'll need to use the r option in your Dial command. Cheers -- _ -- 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] Asterisk 1.6 and Music on Hold
2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com: On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote: Hi i have a small problems on Asterisk 1.6 with the MusiconOld : musiconhold.conf: [Sound_1] mode=quietmp3 directory=/var/lib/asterisk/moh/Sound_1 in extensions.conf: exten = 0532xx,1,Answer exten = 0532xx,2,MusicOnHold(Sound_1) exten = 0532xx,3,Dial(SIP/ACCOUNT001,180,t) exten = 0532xx,4,Hangup When i call to the number, i have the Music Sound_1 but the SIP Phone don't ring ... Where is my error ? and second question, can i said at asterisk that when he receive the call, he play the music from first second ? and repeat at the end of the music. Thanks for your help -- _ -- 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 First, if you don't use the Music on hold command prior to the dial, do you hear ringing? It seems to me that what's going on here is that you're overriding the progress notification that results from the device responding to the invite with TRYING or RINGING by running MOH. If the ringing doesn't occur even when you remove the MOH command, your device is probably not signaling properly and you'll need to use the r option in your Dial command. Hi Thanks for your help, yes, if i don't put the music on hold command, the phone ringing. I have change for put the r but no effect bye olivier -- _ -- 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] Asterisk 1.6 and Music on Hold
On Wed, Nov 24, 2010 at 5:35 AM, Olivier CALVANO o.calv...@gmail.com wrote: 2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com: On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote: Hi i have a small problems on Asterisk 1.6 with the MusiconOld : musiconhold.conf: [Sound_1] mode=quietmp3 directory=/var/lib/asterisk/moh/Sound_1 in extensions.conf: exten = 0532xx,1,Answer exten = 0532xx,2,MusicOnHold(Sound_1) exten = 0532xx,3,Dial(SIP/ACCOUNT001,180,t) exten = 0532xx,4,Hangup When i call to the number, i have the Music Sound_1 but the SIP Phone don't ring ... Where is my error ? and second question, can i said at asterisk that when he receive the call, he play the music from first second ? and repeat at the end of the music. Thanks for your help -- _ -- 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 First, if you don't use the Music on hold command prior to the dial, do you hear ringing? It seems to me that what's going on here is that you're overriding the progress notification that results from the device responding to the invite with TRYING or RINGING by running MOH. If the ringing doesn't occur even when you remove the MOH command, your device is probably not signaling properly and you'll need to use the r option in your Dial command. Hi Thanks for your help, yes, if i don't put the music on hold command, the phone ringing. I have change for put the r but no effect bye olivier -- _ -- 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 Olivier, Your MusicOnHold(Sound_1) command is overriding the progress indications that Asterisk would normally provide. Do you intend to play music on hold, or are you just wishing to set the class for that call? If the latter, use Set(CHANNEL(musicclass)=Sound_1). That would NOT play the Music on hold, thereby allowing Asterisk to provide the progress indications. If you mean to play the music, you're going to have to understand that you won't be able to hear indications (Please read http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial) such as ringing. Does that clear it up? Basically, you cna't have Music On Hold AND Ringing for a channel going at the same time, they're mutually exclusive -- _ -- 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] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
That was it! I had a value (412 and 413) set for each phone. This overwrote the
caller ID that I was setting in the dialplan. Removing the contents of the
fromuser field cleared this issue.
Thanks Olle!
Brett Woollum
br...@woollum.com
- Original Message -
From: Olle E. Johansson o...@edvina.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 11:30:27 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
10 nov 2010 kl. 02.38 skrev Brett Woollum:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack
[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is ringing
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412, 3)
exited non-zero on 'SIP/413-0005'
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Hangup(SIP/413-0005, ) in new stack
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1)
exited non-zero on 'SIP/413-0005'
As you can see on line 3, CallerID(num) was successfully set to 2.
SIP/412 is dialed next. It receives the call, but with 412 as the Caller ID
number - even though the real source of the call was extension 413. The name
I set in CallerID(name) works fine.
My Extensions.conf for that context:
[sipphones]
exten = 412,1,Set(CALLERID(num)=2)
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
exten = 412,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and 413
into sip.conf directly, this code works (ie: the CallerID(num) I set makes it
out to the destination phone properly).
Have you set the fromuser= field in the realtime database?
/O
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark
5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Set(SIP/413-0005, CALLERID(num)=2) in
new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:2] NoOp(SIP/413-0005, CallerID(num) is: 2)
in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:3] Dial(SIP/413-0005, SIP/412) in new stack
[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark
5
[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is
ringing
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension
(sipphones, 412, 3) exited non-zero on 'SIP/413-0005'
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Hangup(SIP/413-0005, ) in new stack
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension
(sipphones, h, 1) exited non-zero on 'SIP/413-0005'
As you can see on line 3, CallerID(num) was successfully set to
2. SIP/412 is dialed next. It receives the call, but with 412
as the Caller ID number - even though the real source of the call was
extension 413. The name I set in CallerID(name) works fine.
My Extensions.conf for that context:
[sipphones]
exten = 412,1,Set(CALLERID(num)=2)
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
exten = 412,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and
413 into sip.conf directly, this code works (ie: the CallerID(num) I
set makes it out to the destination phone properly).
Are you using the fromuser field in the realtime table? I had this
problem once when from user was set and user kept receiving that as the
callerid.
--
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Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Hi Carlos.
Yes I did have fromuser set, which was the problem. I removed this for each
extension and that solved the issue.
Thanks!
Brett Woollum
br...@woollum.com
- Original Message -
From: Carlos Chavez cur...@telecomabmex.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 10, 2010 8:47:38 AM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote: Good idea Paul.
My debug output: [Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP
CoS mark 5 [Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2) in new stack [Nov 9
17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack [Nov 9 17:33:39]
VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5 [Nov 9 17:33:39]
VERBOSE[4175] app_dial.c: -- Called 412 [Nov 9 17:33:40] VERBOSE[4175]
app_dial.c: -- SIP/412-0006 is ringing [Nov 9 17:33:44] VERBOSE[4175]
pbx.c: == Spawn extension (sipphones, 412, 3) exited non-zero on
'SIP/413-0005' [Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Hangup(SIP/413-0005, ) in new stack [Nov 9
17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1) exited
non-zero on 'SIP/413-0005' As you can see on line 3, CallerID(num) was
successfully set to 2. SIP/412 is dialed next. It receives the call,
but with 412 as the Caller ID number - even though the real source of the
call was extension 413. The name I set in CallerID(name) works fine. My
Extensions.conf for that context: [sipphones] exten =
412,1,Set(CALLERID(num)=2) exten =
412,1,Set(CALLERID(all)=TEST2) exten = 412,n,NoOp(CallerID(num) is:
${CALLERID(num)}) exten = 412,n,Dial(SIP/412) exten =
412,n,NoOp(${CALLERID(num)}) If I disable sippusers and sippeers in
extconfig.conf and put 412 and 413 into sip.conf directly, this code works
(ie: the CallerID(num) I set makes it out to the destination phone properly).
Are you using the fromuser field in the realtime table? I had this problem
once when from user was set and user kept receiving that as the callerid. --
Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director
de Tecnología +52-55-91169161 ext 2001
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[asterisk-users] Asterisk 1.6 and Username in Dial
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Nobody has any idea why the Caller ID is being overwritten when using Asterisk
Realtime for the SIP users?
Brett Woollum
br...@woollum.com
- Original Message -
From: Brett Woollum br...@woollum.com
To: asterisk-users@lists.digium.com
Sent: Sunday, November 7, 2010 3:08:50 PM GMT -08:00 US/Canada Pacific
Subject: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num)
Problem
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point everything works in terms of phones registering, placing calls
between them, etc. However, I am having a problem setting the Caller ID number
whenever I am using the Realtime database for the SIP users/peers. If I use a
static sip.conf configuration instead of the database, everything works fine.
Unfortunately a static sip.conf file won't work in my application.
In this example:
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)}) ;;;PS: This shows the
correct number of 2 on the CLI console...
exten = 412,n,Dial(SIP/412)
Whenever another phone calls extension 412, the call is forwarded to SIP/412
and should have TEST as the CallerID name and 2 as the CallerID number.
But, whenever I am using the realtime backend, the caller ID number always
displays on the destination phone as that phone's username. Meaning, if phone
SIP/412 receives the call from the example above, the caller ID name displayed
is TEST but the caller ID number is always 412.
What could be causing this?
Brett Woollum
br...@woollum.com
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum br...@woollum.com wrote: Nobody has any idea why the Caller ID is being overwritten when using Asterisk Realtime for the SIP users? No, perhaps you can _show_ us the problem. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2 ) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack
[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is ringing
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412, 3)
exited non-zero on 'SIP/413-0005'
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Hangup(SIP/413-0005, ) in new stack
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1)
exited non-zero on 'SIP/413-0005'
As you can see on line 3, CallerID(num) was successfully set to 2.
SIP/412 is dialed next. It receives the call, but with 412 as the Caller ID
number - even though the real source of the call was extension 413. The name I
set in CallerID(name) works fine.
My Extensions.conf for that context:
[sipphones]
exten = 412,1,Set(CALLERID(num)=2)
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
exten = 412,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and 413 into
sip.conf directly, this code works (ie: the CallerID(num) I set makes it out to
the destination phone properly).
Brett Woollum
br...@woollum.com
- Original Message -
From: Paul Belanger paul.belan...@polybeacon.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 5:18:36 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum br...@woollum.com wrote:
Nobody has any idea why the Caller ID is being overwritten when using
Asterisk Realtime for the SIP users?
No, perhaps you can _show_ us the problem.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
10 nov 2010 kl. 02.38 skrev Brett Woollum:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack
[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is ringing
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412,
3) exited non-zero on 'SIP/413-0005'
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Hangup(SIP/413-0005, ) in new stack
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1)
exited non-zero on 'SIP/413-0005'
As you can see on line 3, CallerID(num) was successfully set to 2.
SIP/412 is dialed next. It receives the call, but with 412 as the Caller ID
number - even though the real source of the call was extension 413. The name
I set in CallerID(name) works fine.
My Extensions.conf for that context:
[sipphones]
exten = 412,1,Set(CALLERID(num)=2)
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
exten = 412,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and 413
into sip.conf directly, this code works (ie: the CallerID(num) I set makes it
out to the destination phone properly).
Have you set the fromuser= field in the realtime database?
/O
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[asterisk-users] Asterisk 1.6 and Kamailio 3.1 realtime integration tutorial
Hello, I got the time to upgrade my tutorial about Asterisk and Kamailio realtime integration to latest stable release of Kamailio, version 3.1.0 (out on Oct 6, 2010). You can find the document at: * http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb Besides making it work for v3.1.x, the Kamailio config file has some new features included: * IP authentication - can be enabled via define WITH_IPAUTH * TLS support - can be enabled via define WITH_TLS - TLS to UDP translation and vice-versa is done automatically by Kamailio in case you configure Asterisk on UDP * detection of DoS attacks - can be enabled via define WITH_ANTIFLOOD - banning automatically traffic from attacker IP addresses for a specific time interval * restructuring of configuration file for better modularity and highlighting of functionalities such as registrar, location server, within-dialog request routing Hope it is useful for some people within this community. Next step, naturally, is to upgrade the tutorial for latest Asterisk, 1.8.0, just needs some time to get familiar with it. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com -- _ -- 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] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point everything works in terms of phones registering, placing calls
between them, etc. However, I am having a problem setting the Caller ID number
whenever I am using the Realtime database for the SIP users/peers. If I use a
static sip.conf configuration instead of the database, everything works fine.
Unfortunately a static sip.conf file won't work in my application.
In this example:
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)}) ;;;PS: This shows the
correct number of 2 on the CLI console...
exten = 412,n,Dial(SIP/412)
Whenever another phone calls extension 412, the call is forwarded to SIP/412
and should have TEST as the CallerID name and 2 as the CallerID number.
But, whenever I am using the realtime backend, the caller ID number always
displays on the destination phone as that phone's username. Meaning, if phone
SIP/412 receives the call from the example above, the caller ID name displayed
is TEST but the caller ID number is always 412.
What could be causing this?
Brett Woollum
br...@woollum.com
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Re: [asterisk-users] asterisk 1.6 and Firefox 4 Beta
Has anyone started using Firefox 4 beta versions? We started today and find that many of the GUI's attached to Asterisk respond differently and in many cases not at all? We have found that details cannot be saves and that the screens become very unstable. While we appreciate this is a beta Firefox it would appear they have deviated from their 3.x format with regards to interfacing. Rupert Utteridge -- _ -- 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] Asterisk 1.6 Overlap dialling timeout?
Hello, I'm experimenting with Overlap Dialling in asterisk 1.6. I've enabled this in sip.conf and on the SNOM 300 phone. My problem is that asterisk dials out as soon as it matches an extension without waiting to see if the user is going to type in more digits. Is there a way to set a timeout per channel or globally? I'd like Asterisk to wait for a few seconds once its found a match in case the user needs to key in more digits. Thank You. Regards, Veselin K -- _ -- 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] asterisk 1.6 and BLF
Hello list, are there special things that needs to be done when converting BLF from asterisk 1.4 tot 1.6.2 ?! I have replaced call-limit with call-counter, but it seems that the lights on the phone no longer give the status of the extension they monitor. On Snom phones, when the lights should be blinking (indicating a ringing phone) the lights are lighting up constantly (as if the extension is busy). I have not changed my hints in the dialplan. What other steps do I need to take ? Kind regards, Jonas. -- _ -- 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] Asterisk 1.6 and fax
Hello! I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5. When i try to receive fax I get: [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel 'SIP/crocus-ua-0004' refused to negotiate T.38 [Sep 13 00:46:02] WARNING[3283]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Sep 13 00:46:02] WARNING[3283]: app_fax.c:817 transmit: Transmission error I definitely know that this peer supports T.38 because it works on Lynksys PAP2T. Dialplan is such: answer() wait(6) ReceiveFAX(/var/spool/asterisk/test.tif) Am I doing something wrong here? Thanks! -- Stas Korsei On Thu, Sep 9, 2010 at 12:17 AM, David Backeberg dbackeb...@gmail.comwrote: On Wed, Sep 8, 2010 at 4:18 PM, Stanislav Korsei kor...@rinogo.com wrote: Can you recommend any specific solution to this problem or way to install app_fax? Not without specific debugging about what problems you're seeing. You get a lot of information when faxes succeed or fail. Try a fax and paste in the debug. -- _ -- 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
