[asterisk-users] No chan_sip in compiled asterisk-11.13.0
Hello asterisk users, Compiled asterisk-11.13.0 on openSUSE 13.1, however Channel driver chan_sip is XXX in menuselect --- it depends on: chan_local(M), res_crypto(M), res_http_websocket(M) chan_local is [*] chan_local in menuselect, res_crypto is in Resource Modules, Depends on: openssl(E) --- I don't know what (E) means ??? res_http_websocket is [*] res_http_websocket in menuselect. So this means that openssl(E) is holding everything? Can someone give me some help on this? 10q. -- _ -- 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] No chan_sip in compiled asterisk-11.13.0
This is the openssl I have: openssl-1.0.1i-11.52.1.i586 libopenssl1_0_0-1.0.1e-11.2.1.i586 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 04 October 2014 15:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No chan_sip in compiled asterisk-11.13.0 Anthony Azzopardi wrote: So this means that openssl(E) is holding everything? Can someone give me some help on this? On my Debian install, openssl shows as: openssl 1.0.1e-2+deb7u12 My guess is, if you'd do a: rpm -qa|grep -i openssl You'll find that your openssl version is before the 'e' revision. If that's the case, you'll have to upgrade your ssl. Doug -- _ -- 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] Unable to connect to remote asterisk
solved, permissions problem. Asterisks run with user asterisk at default, I changed to asteriskpbx as the book says ;) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Azzopardi Sent: 03 September 2014 20:57 To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -- _ -- 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] (no subject)
Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -- _ -- 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] compiling dahdi and exporting it to another system
Hello asterisk-users, I need to compile dahdi and then export it to another system. I managed to do this with DESTDIR=/root/destDir, then make a tar file and extract in / of the other system. However the module is not loading and /dev/dahdi is not created. Anyone done this? Thank you, Anthony. -- _ -- 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] How do I monitor the whole conversation on a Zap channel ...
How do I monitor the whole conversation on a Zap channel without answering it - the channel is hanging up, I think it's because it's not answered. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I monitor the whole conversation on a Zap channel ...
I want to monitor the zap channel while the phone is answered by a human not by asterisk. Tzafrir Cohen wrote: On Wed, May 10, 2006 at 08:53:36AM +0200, Anthony Azzopardi wrote: How do I monitor the whole conversation on a Zap channel without answering it - the channel is hanging up, I think it's because it's not answered. If the channel is not answered, there is no (useful) audio in it to monitor. What exactly are you trying to do? -- Tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Anthony Azzopardi. Tel: 79713618 Email:[EMAIL PROTECTED] Sign up for free voip from http://line.sytes.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I monitor a Zap channel ...
How do I monitor a Zap channel as soon as the telephone is off the hook, till it is on the hook again? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I monitor a Zap channel ...
Maybe I did'nt put the problem correctly. First of all I'm using an X100P card, and the calls are not bridged. I used both monitor and record and I record audio however the channel hangs up. I cannot find a method to avoid this, and the channel is not answered. Cosmin Prund wrote: I'm using ChanSpy. Set up an extension with ChanSpy, dial the given extension and don't hang up (put it on speakerphone). When there's no one on the given zap channel you'll here silence. As soon as someone's on the channel you'll be listening to them. If you don't like ChanSpy there's ZapBarge. And if by monitor you mean record there's Monitor! Anthony Azzopardi wrote: How do I monitor a Zap channel as soon as the telephone is off the hook, till it is on the hook again? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Anthony Azzopardi. Tel: 79713618 Email:[EMAIL PROTECTED] Sign up for free voip from http://line.sytes.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I have my asterisk machine behind a Linux Nat ...
Hello ppl, I have my asterisk machine behind a Linux Nat router which is connected to the internet. Please tell me the iptables rules and other configurations that I need so that a sip phones on the internet can access asterisk. Best regards, Anthony. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where can I get the tar.gz sources of libnewt?
Where can I get the tar.gz sources of libnewt? Reg, Anthony Azzopardi. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I configure the console to ring on one sound card and the headset on another sound card?
Can I configure the console to ring on one sound card and the headset on another sound card? Best regards, Anthony. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I send DTMF from the console?
How can I send DTMF from the console? Anthony. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How come I don't have the MeetMe application registered?
How come I don't have the MeetMe application registered? Regards, Anthony. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I configure to call from the console by means of a sip phone,
How can I configure to call from the console by means of a sip phone, any docs on this. Regards, Anthony. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How can I configure to call from the console bymeans of a sip phone,
I can call from the console by means of the 'dial' command, now I need to know how to call the console itself. Anthony. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
