[asterisk-users] SOLVED: Asterisk12Beta- configure script/uuid missing??
On Fri, Oct 18, 2013 at 03:16:08PM -0400, Cassius Smith wrote: Hello, I'm trying to build Asterisk12 on a Centos 6.4 VM. The configure script is erring out with: … checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking for uuid_generate_random... no configure: error: *** uuid support not found (this typically means the uuid development package is missing) I have installed (using yum) uuid, uuidd and uuid-devel. No joy, still getting same error. Anyone else run into this? How did you get around it? libuuid-devel is what I think you need. As an aside, in the asterisk source there is an install_prereq script that can be used to install all the necessary packages for your platform: $ sudo contrib/scripts/install_prereq install Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer Thanks Shaun - the install_prereq script did the trick. Cassius -- _ -- 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] l2tp phones - only in China?
All, I'm looking for sip phones that support something other than openvpn. There are a lot of vendors in China (mainly Alibaba) that sell l2tp VPN phones. Are there any American vendors that support l2tp? Thanks, -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emik...@rimmkaufman.com -- http://www.rimmkaufman.com http://twitter.com/rimmkaufman http://www.linkedin.com/company/85385 http://plus.google.com/104980442218952272663/posts http://www.facebook.com/rimmkaufman http://www.RKGblog.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] Problem with call transfer from one server to another server
Dear All, I have pri with E1 facility that have 30 line and 100 pri number which is provided by service provider.Number started like 23568561,23568562,23568563 and so on. Service provider provide last four digit number for did mapping like 4561,4562,4563. exten = 8561,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8561,n,hangup() exten = 8562,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8562,n,hangup() Call comes into first server successful.But problem with second server when call came into second server i got following error: * chan_sip.c:20063 handle_request_invite: Call from '' to extension '4001' rejected because extension not found.* In one more scenario: when i create one extension and call forwarding with this extension that time I'm able to transfer call successful the code is given below: exten = 5001,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 5001,n,hangup() Regards Akhilesh -- _ -- 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] Problem with call transfer from one server to another server
Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd link here. Mitul On Oct 20, 2013 11:07 AM, akhilesh chand omakhileshch...@gmail.com wrote: Dear All, I have pri with E1 facility that have 30 line and 100 pri number which is provided by service provider.Number started like 23568561,23568562,23568563 and so on. Service provider provide last four digit number for did mapping like 4561,4562,4563. exten = 8561,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8561,n,hangup() exten = 8562,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8562,n,hangup() Call comes into first server successful.But problem with second server when call came into second server i got following error: * chan_sip.c:20063 handle_request_invite: Call from '' to extension '4001' rejected because extension not found.* In one more scenario: when i create one extension and call forwarding with this extension that time I'm able to transfer call successful the code is given below: exten = 5001,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 5001,n,hangup() Regards Akhilesh -- _ -- 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
