[asterisk-users] FreePBX, Asterisk and Twinkle - Testing a new setup
I am experimenting with Asterisk having downloaded and installed the FreePBX i386 CentOS-6.3 based distro and updated it. The current package level on this system is: asterisk11-11.3.0-49_centos6 freepbx-2.11.0beta2-112 I am using twinkle-1.4.2-7.el6 as a softphone testing tool. There is no firewall on the asterisk host and SELinux is disabled on it. Fail2Ban is installed but I have made no alterations to the default configuration, whatever it is. The asterisk host is configured as 192.168.6.122. The softphone is configured on a separate host with a routable IP on our 216.xxx.xxx/24 netblock. Both networks pass though an internal switch and are firewalled from the outside world by a centos-6.4 based gateway host using IPTables. I have no difficulty in connecting to the asterisk host either by ssh or by https. I have initialised the FreePBX config and have selected the user/device approach as this seems to fit our firm's employee requirements more closely than the extension based configuration. We have several employees who frequently telecommute. For the purposes of testing I have created two users, 11 and 12. I have configured a twinkle user profile for user 12. I can place a call to user 11 from twinkle and I get the IVR message for 'the number you have called is not in service'. I have tried to register Twinkle and this always fails. If I do : # asterisk -r CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Description 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] CLI sip show users Username Secret Accountcode Def.Context ACL ForcerPort Which seems to say to me that I have nothing configured albeit I have tried to through FreePBX. At this point I am not trying to get a call out to our PTSN, although I have the FXO port plugged into a live analogue line. What I am trying to understand is the relationship between asterisk devices and users. The twinkle softphone has two lines (1 and 2). It seems to me that I should be able to configure each line as a separate extension and to call one from the other. What I cannot seem to discover is how to do it. Is it possible to do this? How is it done? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Feature request: Need to INVITE to peer with other domain without peer domain addition
31.03.2013 23:15, Barry Flanagan ?: On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com mailto:serov@gmail.com wrote: Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten = s,n,Dial(SIP/peer1/num...@domain2.com mailto:num...@domain2.com,60,r) [peer1] type=friend host=domain1.com http://domain1.com fromdomain=domain1.com http://domain1.com As a result in SIP packet uri: num...@domain2.com@domain1.com http://domain1.com I need: num...@domain2.com mailto:num...@domain2.com I can't use SIP uri dial, i need authorization (peer1) I think asterisk can't do that. Is where work around? Would it work if you created a sip peer [domain2.com http://domain2.com] and set outboundproxy=domain1.com http://domain1.com then sent the call to SIP/num...@domain2.com mailto:num...@domain2.com ? -Barry does not matter. [skype.ippi.com](srv-options-common) type=friend secret=xxx host=ippi.fr fromdomain=ippi.fr outboundproxy=ippi.fr exten = 22,n,Dial(SIP/lo...@skype.ippi.com,60,rS(1200)) INVITE sip:lo...@ippi.fr SIP/2.0 Via: SIP/2.0/UDP 109.60.163.xx:5060;branch=z9hG4bK60e845b5;rport Max-Forwards: 70 From: demon sip:usern...@ippi.fr;tag=as518b59df To: sip:lo...@ippi.fr and exten = 22,n,Dial(SIP/skype.ippi.com/lo...@skype.ippi.com,60,rS(1200)) do: INVITE sip:lo...@skype.ippi.com@ippi.fr SIP/2.0 I studied the source code and found no ways to implement it :( Dmitriy. -- _ -- 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] Feature request: Need to INVITE to peer with other domain without peer domain addition
On 13-04-01 03:16 PM, Dmitriy Serov wrote: 31.03.2013 23:15, Barry Flanagan ?: On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com mailto:serov@gmail.com wrote: Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten = s,n,Dial(SIP/peer1/num...@domain2.com mailto:num...@domain2.com,60,r) [peer1] type=friend host=domain1.com http://domain1.com fromdomain=domain1.com http://domain1.com As a result in SIP packet uri: num...@domain2.com@domain1.com http://domain1.com I need: num...@domain2.com mailto:num...@domain2.com I can't use SIP uri dial, i need authorization (peer1) I think asterisk can't do that. Is where work around? Would it work if you created a sip peer [domain2.com http://domain2.com] and set outboundproxy=domain1.com http://domain1.com then sent the call to SIP/num...@domain2.com mailto:num...@domain2.com ? -Barry does not matter. [skype.ippi.com](srv-options-common) type=friend secret=xxx host=ippi.fr fromdomain=ippi.fr outboundproxy=ippi.fr exten = 22,n,Dial(SIP/lo...@skype.ippi.com,60,rS(1200)) INVITE sip:lo...@ippi.fr SIP/2.0 Via: SIP/2.0/UDP 109.60.163.xx:5060;branch=z9hG4bK60e845b5;rport Max-Forwards: 70 From: demon sip:usern...@ippi.fr;tag=as518b59df To: sip:lo...@ippi.fr and exten = 22,n,Dial(SIP/skype.ippi.com/lo...@skype.ippi.com,60,rS(1200)) do: INVITE sip:lo...@skype.ippi.com@ippi.fr SIP/2.0 I studied the source code and found no ways to implement it :( Dmitriy. How about: exten =22,n,Dial(SIP/skype.ippi.com!lo...@skype.ippi.com,60,rS(1200)) -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://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] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
On 31/03/13 23:43, Joshua Colp wrote: Daniel Pocock wrote: I'm trying to call from DruCall to Asterisk and I get this error: WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F 103 104 111 0 8 107 106 105 13 126' == Problem setting up ssl connection: error::lib(0):func(0):reason(0) I'm guessing my Asterisk is too old (it is 1.8 from Debian). Can you confirm which version is needed to parse a media descriptor with SAVPF? Do I need to upgrade all the way to v11 with WebRTC support, or was avpf support added in some intermediate version? Asterisk 1.8 does not have any knowledge of AVPF, and since it's a new feature it was only added to Asterisk 11. You could try to backport the changes but chan_sip has changed quite a bit, so it could be rough. Thanks for the fast reply. I agree backporting full support for AVPF would not be justified for many use cases (including my own). What I was more curious about is whether the F can be tolerated (in other words, ignored or silently removed), as described here: http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html 1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling gateway to legacy will change that by removing the F to AVP or SAVP. and whether such behavior is possible even without setting avpf=yes on a per-peer basis? -- _ -- 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] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
Daniel Pocock wrote: Thanks for the fast reply. I agree backporting full support for AVPF would not be justified for many use cases (including my own). What I was more curious about is whether the F can be tolerated (in other words, ignored or silently removed), as described here: From a code perspective, it could. Still not something I would be comfortable with putting in Asterisk 1.8. http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html 1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling gateway to legacy will change that by removing the F to AVP or SAVP. and whether such behavior is possible even without setting avpf=yes on a per-peer basis? This is fine for incoming but what about outgoing to a device? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
[asterisk-users] Getting DIALSTATUS via agi
Hi all,
Hopefully, I just need a second set of eyes on this one, but I just can't
figure out what I'm doing wrong. I'm using an agi script to dial a number,
check the dial result, and act accordingly.
The problem is that I'm not getting anything back from DIALSTATUS, or
HANGUPCAUSE.
Here is the relevant perl code:
===
$main::agi-verbose(
, 1);
$main::agi-exec(dial,
sip/$t-{trunk}/$t-{prefix}$o-{number},240,KrTWX);
$result = $main::agi-get_variable(DIALSTATUS);
$cause = $main::agi-get_variable(HANGUPCAUSE);
$o-{DIALSTATUS} = $result;
$o-{HANGUPCAUSE} = $cause;
$main::agi-verbose(= The
result of the last call was ($o-{DIALSTATUS}) ($o-{HANGUPCAUSE}), 1);
$main::agi-verbose(
, 1);
===
Here is the relevant output with agi debug turned on:
===
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx 200 result=0
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx VERBOSE
1
[Apr 1 15:01:55] magic_switch.pl:
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx 200 result=1
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx EXEC dial
sip/didlogic/15058457900,240,KrTWX
[Apr 1 15:01:55] -- AGI Script Executing Application: (dial) Options:
(sip/didlogic/15058457900,240,KrTWX)
[Apr 1 15:01:55] == Using SIP RTP TOS bits 24
[Apr 1 15:01:55] == Using SIP RTP CoS mark 5
[Apr 1 15:01:55] -- Called sip/didlogic/15058457900
[Apr 1 15:01:55] -- Got SIP response 603 Declined back from
178.63.143.236:5060
[Apr 1 15:01:55] -- SIP/didlogic-7788 is busy
[Apr 1 15:01:55] == Everyone is busy/congested at this time (1:1/0/0)
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx 200 result=0
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx GET VARIABLE
DIALSTATUS
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx 200 result=1 (BUSY)
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx GET VARIABLE
HANGUPCAUSE
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx 200 result=1 (21)
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx VERBOSE
= The result of the last call was () () 1
[Apr 1 15:01:55] magic_switch.pl: = The
result of the last call was () ()
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx 200 result=1
[Apr 1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx VERBOSE
1
[Apr 1 15:01:55] magic_switch.pl:
===
So, the result of my previous AGI command was 200. Then I can clearly see
that the result of this dial is BUSY/21, which I understand. But, my perl
script isn't getting this value. I do the assignment to/from $result and
$cause because I use those values later in the script, but have to pass
them back to a global routine, as well.
Can anyone see what I'm doing wrong?
Mike Diehl.
--
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Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
On 01/04/13 22:06, Joshua Colp wrote: Daniel Pocock wrote: Thanks for the fast reply. I agree backporting full support for AVPF would not be justified for many use cases (including my own). What I was more curious about is whether the F can be tolerated (in other words, ignored or silently removed), as described here: From a code perspective, it could. Still not something I would be comfortable with putting in Asterisk 1.8. http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html 1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling gateway to legacy will change that by removing the F to AVP or SAVP. and whether such behavior is possible even without setting avpf=yes on a per-peer basis? This is fine for incoming but what about outgoing to a device? Excellent question... I've seen one of my Polycom devices reboot itself each time it receives a raw SDP from WebRTC, so if such a hack is implemented, I'd guess that stripping the F is better than ignoring it. -- _ -- 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
