Re: [asterisk-users] Bypass queue wrapup time
2012/10/31 Benny Amorsen benny+use...@amorsen.dk Olivier oza_4...@yahoo.fr writes: That's the point : to me, casual @pickupmark mechanism don't work with calls that entered into a queue : the extension rings but you can't pick the call up with a directed pickup. (For general pickup, that's another strory). (and I would be very pleased to be wrong) That seems to be fixed a long time ago, if I read the various issues correctly. I haven't actually tried it. The same for me: I haven't tried it lately ;-) I think a patch including Directed Pickup support in Queue was developped but it was not included yet in any branch, if I'm not mistaken. Reading AstriConDev notes, breaking Queue app into smaller parts is on Asterisk 12 menu. Maybe, this Directed Pickup support will come with this Queue re-factoring, though I'm sure a patch bringing it to current Queue exists. /Benny -- _ -- 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 does not re-register as a sip client after a sip reload if sip.conf or users.conf is changed
Hi, I just want to confirm that my problem is solved now and everything is working as expected . I used the patch provided in the following link: https://reviewboard.asterisk.org/r/2171/ Special thanks to Asterisk development team for great responsibility and quick reaction. regards Unfortunately this appears to be an issue with Asterisk 11. You can follow progress on solving it at https://issues.asterisk.org/**jira/browse/ASTERISK-20611https://issues.asterisk.org/jira/browse/ASTERISK-20611 -- _ -- 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 and OpenLDAP
Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username and password stored in ldap database. How can i configure this mechanism? Thanks in advice, Regards. -- _ -- 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 and OpenLDAP
You just need a program(C, PHP, Perl) to query LDAP and update SIP. The example you list requires realtime, but if you roll your own, you could update /etc/asterisk/sip.conf and issue an 'asterisk -rx sip reload' to update when needed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk and OpenLDAP Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/External Services_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username and password stored in ldap database. How can i configure this mechanism? Thanks in advice, Regards. -- _ -- 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 and OpenLDAP
2012/10/31 Danny Nicholas da...@debsinc.com: You just need a program(C, PHP, Perl) to query LDAP and update SIP. The example you list requires realtime, but if you roll your own, you could update /etc/asterisk/sip.conf and issue an 'asterisk -rx sip reload' to update when needed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk and OpenLDAP Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/External Services_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username and password stored in ldap database. How can i configure this mechanism? Thanks in advice, Regards. -- _ -- 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 Thanks for your help, i've only a question. How do i configure extensions? -- _ -- 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 and OpenLDAP
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 8:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP 2012/10/31 Danny Nicholas da...@debsinc.com: You just need a program(C, PHP, Perl) to query LDAP and update SIP. The example you list requires realtime, but if you roll your own, you could update /etc/asterisk/sip.conf and issue an 'asterisk -rx sip reload' to update when needed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk and OpenLDAP Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Ex ternal Services_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username and password stored in ldap database. How can i configure this mechanism? Thanks in advice, Regards. -- _ -- 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 Thanks for your help, i've only a question. How do i configure extensions? Just create a peer for each extension like this: [sipuser] type=friend context=default host=dynamic secret=xx canreinvite=yes directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = sipuser:xxx@xxx/sipuser defaultip=192.168.23.107 disallow=all allow=gsm -- _ -- 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 and OpenLDAP
With this configuration, the peer doesn't authenticate with ldap, right? 2012/10/31 Danny Nicholas da...@debsinc.com: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 8:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP 2012/10/31 Danny Nicholas da...@debsinc.com: You just need a program(C, PHP, Perl) to query LDAP and update SIP. The example you list requires realtime, but if you roll your own, you could update /etc/asterisk/sip.conf and issue an 'asterisk -rx sip reload' to update when needed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk and OpenLDAP Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Ex ternal Services_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username and password stored in ldap database. How can i configure this mechanism? Thanks in advice, Regards. -- _ -- 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 Thanks for your help, i've only a question. How do i configure extensions? Just create a peer for each extension like this: [sipuser] type=friend context=default host=dynamic secret=xx canreinvite=yes directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = sipuser:xxx@xxx/sipuser defaultip=192.168.23.107 disallow=all allow=gsm -- _ -- 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 and OpenLDAP
Correct. LDAP can be queried to update the Asterisk configuration, but Asterisk itself is unaware of LDAP. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP With this configuration, the peer doesn't authenticate with ldap, right? 2012/10/31 Danny Nicholas da...@debsinc.com: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 8:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP 2012/10/31 Danny Nicholas da...@debsinc.com: You just need a program(C, PHP, Perl) to query LDAP and update SIP. The example you list requires realtime, but if you roll your own, you could update /etc/asterisk/sip.conf and issue an 'asterisk -rx sip reload' to update when needed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk and OpenLDAP Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/E x ternal Services_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username and password stored in ldap database. How can i configure this mechanism? Thanks in advice, Regards. -- _ -- 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 Thanks for your help, i've only a question. How do i configure extensions? Just create a peer for each extension like this: [sipuser] type=friend context=default host=dynamic secret=xx canreinvite=yes directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = sipuser:xxx@xxx/sipuser defaultip=192.168.23.107 disallow=all allow=gsm -- _ -- 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 and OpenLDAP
Based on my knowledge, the general section provides an interface to your LDAP server and the sipuser section sets up one static user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP This configuration is wrong? [_general] url=ldap://172.16.0.103:389 protocol=3 basedn=dc=shifteight,dc=org user=cn=admin,dc=shifteight,dc=org pass=canada [sipuser] name=cn type=friend context=default host=dynamic secret=AstAccountSecret canreinvite=yes directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = sipuser:xxx@xxx/sipuser defaultip=192.168.23.107 disallow=all allow=gsm -- _ -- 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 and OpenLDAP
I don't want update Asterisk configuration, i want to query LDAP only for name and secret field. -- _ -- 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 and OpenLDAP
I don't understand why in [_general] section of res_ldap.conf i need to put user and pass when i want to authenticate my extensions. 2012/10/31 Danny Nicholas da...@debsinc.com: Based on my knowledge, the general section provides an interface to your LDAP server and the sipuser section sets up one static user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP This configuration is wrong? [_general] url=ldap://172.16.0.103:389 protocol=3 basedn=dc=shifteight,dc=org user=cn=admin,dc=shifteight,dc=org pass=canada [sipuser] name=cn type=friend context=default host=dynamic secret=AstAccountSecret canreinvite=yes directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = sipuser:xxx@xxx/sipuser defaultip=192.168.23.107 disallow=all allow=gsm -- _ -- 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 and OpenLDAP
This allows asterisk to open an LDAP connection. Have you reviewed res_ldap.conf.sample in the configs folder? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 9:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP I don't understand why in [_general] section of res_ldap.conf i need to put user and pass when i want to authenticate my extensions. 2012/10/31 Danny Nicholas da...@debsinc.com: Based on my knowledge, the general section provides an interface to your LDAP server and the sipuser section sets up one static user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP This configuration is wrong? [_general] url=ldap://172.16.0.103:389 protocol=3 basedn=dc=shifteight,dc=org user=cn=admin,dc=shifteight,dc=org pass=canada [sipuser] name=cn type=friend context=default host=dynamic secret=AstAccountSecret canreinvite=yes directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = sipuser:xxx@xxx/sipuser defaultip=192.168.23.107 disallow=all allow=gsm -- _ -- 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 and OpenLDAP
Yes, but i think that's better to open an LDAP connection with extensions user and password. Or not? 2012/10/31 Danny Nicholas da...@debsinc.com: This allows asterisk to open an LDAP connection. Have you reviewed res_ldap.conf.sample in the configs folder? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 9:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP I don't understand why in [_general] section of res_ldap.conf i need to put user and pass when i want to authenticate my extensions. 2012/10/31 Danny Nicholas da...@debsinc.com: Based on my knowledge, the general section provides an interface to your LDAP server and the sipuser section sets up one static user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP This configuration is wrong? [_general] url=ldap://172.16.0.103:389 protocol=3 basedn=dc=shifteight,dc=org user=cn=admin,dc=shifteight,dc=org pass=canada [sipuser] name=cn type=friend context=default host=dynamic secret=AstAccountSecret canreinvite=yes directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = sipuser:xxx@xxx/sipuser defaultip=192.168.23.107 disallow=all allow=gsm -- _ -- 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 and OpenLDAP
Don't really know. My knowledge scale on this one is 99 percent asterisk 1 percent LDAP. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP Yes, but i think that's better to open an LDAP connection with extensions user and password. Or not? 2012/10/31 Danny Nicholas da...@debsinc.com: This allows asterisk to open an LDAP connection. Have you reviewed res_ldap.conf.sample in the configs folder? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 9:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP I don't understand why in [_general] section of res_ldap.conf i need to put user and pass when i want to authenticate my extensions. 2012/10/31 Danny Nicholas da...@debsinc.com: Based on my knowledge, the general section provides an interface to your LDAP server and the sipuser section sets up one static user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo Sent: Wednesday, October 31, 2012 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OpenLDAP This configuration is wrong? [_general] url=ldap://172.16.0.103:389 protocol=3 basedn=dc=shifteight,dc=org user=cn=admin,dc=shifteight,dc=org pass=canada [sipuser] name=cn type=friend context=default host=dynamic secret=AstAccountSecret canreinvite=yes directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=60 session-minse=120 session-refresher=uac register = sipuser:xxx@xxx/sipuser defaultip=192.168.23.107 disallow=all allow=gsm -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call drop weirdness
I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. So I've been watching this problem and was finally able to get a pcap while it happened. I've attached a sanitized text version of the SIP signaling surrounding the time the outbound RTP stream dropped on this particular call. I'm no SIP expert, so there may not be enough info in the file to tell anything. A few notes about the file: 1. X.X.X.X is the public IP our asterisk server is behind. 2. Y.Y.Y.Y is the IP given to us by our provider to use in our SIP trunk through which inbound calls arrive. 3. Z.Z.Z.Z is the IP of our provider's server involved in the RTP stream. 4. DID is our DID. 5. CID is the number of the incoming caller. 6. The outbound RTP stream appears to drop three packets prior to the SIP BYE request. Any thoughts on what might be going wrong? Do I need to post more info? Or am I on the wrong track altogether? Kind Regards, Chris OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24883b1c;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as395fd02c To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 59e57daf68d31595480ca5927505485f@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24883b1c;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as395fd02c To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 59e57daf68d31595480ca5927505485f@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 SIP/2.0 503 Unable to load gateways Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24883b1c;rport=5060 From: Unknown sip:Unknown@X.X.X.X;tag=as395fd02c To: sip:Y.Y.Y.Y;tag=71fd1b189ab888f8d5fb24b00af87228.acb1 Call-ID: 59e57daf68d31595480ca5927505485f@X.X.X.X:5060 CSeq: 102 OPTIONS Server: DFSGW Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 SIP/2.0 503 Unable to load gateways Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK10bfdfb7;rport=5060 From: Unknown sip:Unknown@X.X.X.X;tag=as789d30aa To: sip:Y.Y.Y.Y;tag=71fd1b189ab888f8d5fb24b00af87228.1bf8 Call-ID: 5309893149786df519e2ffa6282fb15d@X.X.X.X:5060 CSeq: 102 OPTIONS Server: DFSGW Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK65801b0b;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as46c7a0b8 To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 0d78056a4c3c4b5d167b013c41450be9@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 OPTIONS sip:Y.Y.Y.Y SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK65801b0b;rport Max-Forwards: 70 From: Unknown sip:Unknown@X.X.X.X;tag=as46c7a0b8 To: sip:Y.Y.Y.Y Contact: sip:Unknown@X.X.X.X:5060 Call-ID: 0d78056a4c3c4b5d167b013c41450be9@X.X.X.X:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.1(10.7.0) Date: Tue, 30 Oct 2012 17:49:30 GMT Allow: INVITE,
Re: [asterisk-users] Asterisk and OpenLDAP
Giuseppe wrote: Yes, but i think that's better to open an LDAP connection with extensions user and password. Or not? Better is not the right way to look at it. You questions is about early or late binding. Early binding requires a dedicated username and password to connect to LDAP before it can perform a query, and late can use the user provided credentials. I find that many applications will support only one or the other, so the choice is made for you. I do not know if Asterisk supports only early binding, but I suspect that it would be a better long term match for you. 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
[asterisk-users] Asterisk 11 and stdexten written in AEL invoked by pbx_config
Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten written in anything else but extensions.conf (AEL, LUA, etc, being these not able to define an arbitrary priority) will not work. The only way to workaround this is to fallback to Macro() and write macro-contexts in AEL with the stack limit implications of them so I'm proposing to add to asterisk.conf configuration the ability to invoke stdexten using AELSub() so stdexten can be again be written in AEL mantaining real backward compatiblity as it did the fact that you are able to fallback to Macro. ;stdexten = gosub ; How to invoke the extensions.conf stdexten. ; macro - Invoke the stdexten using a macro as ; done by legacy Asterisk versions. ; aelsub - Invoke the stdexten sutbroutine using AELSub ; when stdexten is defined in AEL. ; gosub - Invoke the stdexten using a gosub as ; documented in extensions.conf.sample. I've already started this conversation on the development lists, you can follow up it at: http://lists.digium.com/pipermail/asterisk-dev/2012-August/05.html and there is a working patch submited to JIRA here: https://issues.asterisk.org/jira/browse/ASTERISK-20355 I would like to read your comments. Best, -- _ -- 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] Multitenant opensouce application
is another way to build Multi Tenant system, have to design like Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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] multitenanat third party app
Is it possible to bul multitenant system using some third party opensouce application My design is like this. Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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] multitenanat third party app
Hi You will need change the names for your extensions 101-company_a 102-company_a ETC On Wed, Oct 31, 2012 at 2:23 PM, Darin Iv adari...@gmail.com wrote: Is it possible to bul multitenant system using some third party opensouce application My design is like this. Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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] Multitenant opensouce application
On 31/10/12 6:20 pm, Darin Iv wrote: is another way to build Multi Tenant system, have to design like Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. snip Is there any particular reason why it needs to be _exactly_ like that? FWIW, we use companyA-201, companyB-201, companyA-202, companyB-202 as our SIP usernames. Each companyX then has its own extensions.conf file which contains a specific [companyX] context. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] USB FXS device
Anyone manage to make one of these work *on* an asterisk server? Have been researching most of the morning and have only found windows-centric devices that talk SIP to asterisk (of course). I want one that has a Linux driver that preferably could be an asterisk channel itself. WIthout spending a gazillion dollars of course :) I found this: Broadtel UPA-1. I have email inquiries into them, but I saw in a blog post that they would provide Linux drivers on order, but nothing further... Cheers, j -- _ -- 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] USB FXS device
On 10/31/2012 01:38 PM, Jeff LaCoursiere wrote: Anyone manage to make one of these work *on* an asterisk server? Have been researching most of the morning and have only found windows-centric devices that talk SIP to asterisk (of course). I want one that has a Linux driver that preferably could be an asterisk channel itself. WIthout spending a gazillion dollars of course :) I found this: Broadtel UPA-1. I have email inquiries into them, but I saw in a blog post that they would provide Linux drivers on order, but nothing further... Cheers, j Also found the Digium S100U which is exactly what I want... but doesn't seem to be available anymore? Cheers, j -- _ -- 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] USB FXS device
On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote: Anyone manage to make one of these work *on* an asterisk server? Have been researching most of the morning and have only found windows-centric devices that talk SIP to asterisk (of course). I want one that has a Linux driver that preferably could be an asterisk channel itself. WIthout spending a gazillion dollars of course :) Xorcom's Astribanks have native support in DAHDI http://www.xorcom.com/telephony-interfaces/astribank-usb-channel-banks.html -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USB FXS device
On 10/31/2012 01:44 PM, Russ Meyerriecks wrote: On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote: Anyone manage to make one of these work *on* an asterisk server? Have been researching most of the morning and have only found windows-centric devices that talk SIP to asterisk (of course). I want one that has a Linux driver that preferably could be an asterisk channel itself. WIthout spending a gazillion dollars of course :) Xorcom's Astribanks have native support in DAHDI http://www.xorcom.com/telephony-interfaces/astribank-usb-channel-banks.html Yes, but that goes against the spending a gazillion dollars requirement, and though I didn't specify my needs, I am just looking for a single FXS port. Basically I would like to build an ATA out of a Raspberry Pi :) Ideally for $100. Cheers, j -- _ -- 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] USB FXS device
On 10/31/2012 02:49 PM, Jeff LaCoursiere wrote: On 10/31/2012 01:44 PM, Russ Meyerriecks wrote: On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote: Anyone manage to make one of these work *on* an asterisk server? Have been researching most of the morning and have only found windows-centric devices that talk SIP to asterisk (of course). I want one that has a Linux driver that preferably could be an asterisk channel itself. WIthout spending a gazillion dollars of course :) Xorcom's Astribanks have native support in DAHDI http://www.xorcom.com/telephony-interfaces/astribank-usb-channel-banks.html Yes, but that goes against the spending a gazillion dollars requirement, and though I didn't specify my needs, I am just looking for a single FXS port. Basically I would like to build an ATA out of a Raspberry Pi :) Ideally for $100. why punish yourself like that ? pap2t is 2xFXS hanging off a network jack for $50 Half your budget, twice the density, and a nice box too Cheers, j -- _ -- 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] USB FXS device
On 10/31/2012 02:38 PM, Jeff LaCoursiere wrote: why not just get a usb headset and use with one of the sip client apps ? if you're going to the trouble of having a phone to plug in the fxs why rely on the pc at all ? use one of the spa type routers and plug the pc into it and the phone or if you have a free network jack just use a pap2t Then it works whether you have to reboot the pc etc and does not steal cpu cycles from the pc. Anyone manage to make one of these work *on* an asterisk server? Have been researching most of the morning and have only found windows-centric devices that talk SIP to asterisk (of course). I want one that has a Linux driver that preferably could be an asterisk channel itself. WIthout spending a gazillion dollars of course :) I found this: Broadtel UPA-1. I have email inquiries into them, but I saw in a blog post that they would provide Linux drivers on order, but nothing further... Cheers, j -- _ -- 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] USB FXS device
On 10/31/2012 02:00 PM, jon pounder wrote: On 10/31/2012 02:49 PM, Jeff LaCoursiere wrote: On 10/31/2012 01:44 PM, Russ Meyerriecks wrote: On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote: Anyone manage to make one of these work *on* an asterisk server? Have been researching most of the morning and have only found windows-centric devices that talk SIP to asterisk (of course). I want one that has a Linux driver that preferably could be an asterisk channel itself. WIthout spending a gazillion dollars of course :) Xorcom's Astribanks have native support in DAHDI http://www.xorcom.com/telephony-interfaces/astribank-usb-channel-banks.html Yes, but that goes against the spending a gazillion dollars requirement, and though I didn't specify my needs, I am just looking for a single FXS port. Basically I would like to build an ATA out of a Raspberry Pi :) Ideally for $100. why punish yourself like that ? pap2t is 2xFXS hanging off a network jack for $50 Half your budget, twice the density, and a nice box too Doesn't support OpenVPN... Our architecture (hosted PBX offering) works entirely over OpenVPN - the phones we provide do it natively (Yealink). Haven't been able to find an ATA that will. So our customers that have analog devices today need an ATA that can reach their hosted PBX. Today we front a PAP2T (actually the Cisco equiv now that the Linksys line is EOL) with a DD-WRT box whose only function is to provide OpenVPN access to their hosted PBX. Now I have this awesome little box - the Raspberry Pi - that can run asterisk natively for $35 (well closer to $50 with the needed parts). Obviously it can also run OpenVPN. Gives me all kinds of possibilities for tracking link quality. If I could find a USB FXS dongle that would work as an asterisk channel (much like Xorcom devices are supported by dahdi), I would be all set. I wouldn't expect such a device to cost much more than $30 - $50, which gets me what I need for $100, which is currently what I spend on the DD-WRT/Cisco combo. The basic question was has anyone made a USB FXS device work with asterisk. Now that I have additionally defended my architecture decisions, can anyone actually answer the question? Thanks, j -- _ -- 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] USB FXS device
Jeff LaCoursiere j...@sunfone.com writes: The basic question was has anyone made a USB FXS device work with asterisk. Now that I have additionally defended my architecture decisions, can anyone actually answer the question? The Open USB FXS project is exactly what you want. It seems to be discontinued. Depending on volume, it might be worth resurrecting the project -- it looks like the price could get reasonable if you need a few thousand... It does not seem like there is anything commercially available right now. /Benny -- _ -- 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] multitenanat third party app
Stop asking same questions !!! On Oct 31, 2012 11:54 PM, Darin Iv adari...@gmail.com wrote: Is it possible to bul multitenant system using some third party opensouce application My design is like this. Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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] multitenanat third party app
Indeed this is getting ridiculous. This person also called me (!!) for some free consulting after I had posted the answer a few days ago. NOTE: We aren't going to engineer your system for you! We as a group will provide help and some basic code to get you started. If you don't know how to start working with the fully working stuff I provided already, you're not ready to deploy a system this complex. On Wed, Oct 31, 2012 at 2:59 PM, Mitul Limbani mi...@enterux.in wrote: Stop asking same questions !!! On Oct 31, 2012 11:54 PM, Darin Iv adari...@gmail.com wrote: Is it possible to bul multitenant system using some third party opensouce application My design is like this. Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] SIP - Authenticated vs Unauthenticated Calls
Greetings- I'm running into an issue as follows, in simplified form: A remote Asterisk box, when registered/peered via SIP to a central server, and makes a call to that central server, is *sometimes* authenticated and calls go through properly (via from-internal context), and *sometimes* is unauthenticated, and all calls are greeted with congestion() via the from-sip-external context. Yes, as you can tell, FreePBX is in play here too. Grabbing captures of a working call vs a non-working call, I'm seeing on the working call, the central Asterisk server is responding to the INVITE with a 407 Proxy Authentication Needed, box responds, call goes through. On the non-working calls, the central Asterisk server is responding with a simple 100 Trying, then 200 OKs the session as it throws it into from-sip-external assuming the box is not authenticated. So... and pardon my rambling above... why is this the case? In what circumstances would Asterisk respond to the same peer differently, seemingly at random? I'm happy to provide any details required, but I'm having a brain freeze on what would be relevant at this point. Thanks for any pointers or ideas! --Tim -- _ -- 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
