Re: [asterisk-users] Any way of "flattening out" 2 channels back into one?
Oh... I looked at that before, but I don't see how to play a warning
before the caller is disconnected with TIMEOUT?
On Sat, 28 Jul 2018 at 23:05, Social Boh wrote:
>
> TIMEOUT function:
>
> example
>
> same => n,Set(TIMEOUT(absolute)=600)
>
> after 600 seconds Asterisk Hankup the call
>
> Regards
>
> ---
> I'm SoCIaL, MayBe
>
> On 7/28/18 16:08, Jonathan H wrote:
> > Last question for today, I promise!
> >
> > The problem: In order to disconnect calls after x minutes, I need to do
> > this:
> >
> > [setup]
> > exten => setup,1,Answer()
> > same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
> > same =>
> > n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached)
> > same => n,Dial(Local/s@root/n,3,L(354:6))
> > same => n,Hangup()
> >
> > [root]
> > exten => s,1,Verbose(1,Call to: ${CALLERID(name)} from: ${CALLERID(num)})
> > same => n,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1)
> >
> > etc etc
> >
> > Works well, but the result is it looks like there are 2 active calls
> > in the console. Is there any way of forcing the drop of a call after x
> > minutes without doing this "double dialling" business?
> >
> > Thanks
> >
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Any way of "flattening out" 2 channels back into one?
TIMEOUT function:
example
same => n,Set(TIMEOUT(absolute)=600)
after 600 seconds Asterisk Hankup the call
Regards
---
I'm SoCIaL, MayBe
On 7/28/18 16:08, Jonathan H wrote:
Last question for today, I promise!
The problem: In order to disconnect calls after x minutes, I need to do this:
[setup]
exten => setup,1,Answer()
same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
same =>
n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached)
same => n,Dial(Local/s@root/n,3,L(354:6))
same => n,Hangup()
[root]
exten => s,1,Verbose(1,Call to: ${CALLERID(name)} from: ${CALLERID(num)})
same => n,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1)
etc etc
Works well, but the result is it looks like there are 2 active calls
in the console. Is there any way of forcing the drop of a call after x
minutes without doing this "double dialling" business?
Thanks
--
_
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Re: [asterisk-users] SRV with pjsip on Asterisk 15.5: yes or no?
On Sat, Jul 28, 2018, at 6:28 PM, Jonathan H wrote: > OK, thanks. Shall I file a ticket to get that example file updated? Sure! -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
On Sat, Jul 28, 2018, at 6:27 PM, Jonathan H wrote: > Thanks, but... whoah! I think I just found a bug! > > As soon as I changed > accepts_registrations = yes > to > sends_registrations = yes > > and did a pjsip reload, Asterisk crashed. I tried starting asterisk. > Nothing. In the syslog: > > Jul 28 22:20:41 televox kernel: [ 50.728769] asterisk[1504]: > segfault at 0 ip 7f4be3e00646 sp 7ffc32067388 error 4 in > libc-2.27.so[7f4be3d4f000+1e7000] > Jul 28 22:22:02 televox kernel: [ 132.413114] asterisk[1579]: > segfault at 0 ip 7f62a9ba2646 sp 7ffc9215d408 error 4 in > libc-2.27.so[7f62a9af1000+1e7000] > > Took that line back out, and Asterisk started again. Shall I file a bug? Yes, issues should be filed on the issue tracker[1]. It may be something particular about your config. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRV with pjsip on Asterisk 15.5: yes or no?
OK, thanks. Shall I file a ticket to get that example file updated? On Sat, 28 Jul 2018 at 21:50, Joshua Colp wrote: > > On Sat, Jul 28, 2018, at 5:42 PM, Jonathan H wrote: > > I'm trying to configure sip2sip, which says: > > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk > > "Asterisk, is currently unable to handle more that one result for a > > DNS SRV lookup, and the Asterisk configuration needed for getting it > > work with the SIP2SIP service is not trivial" > > > > It then gives a complex multi-section workaround in SIP. I remember > > reading there'd be the same issue with PJSIP, and then I found this > > post in the Asterisk blog from 2016: > > https://blogs.asterisk.org/2016/04/20/pjsip-dns-support/ which says: > > "chan_pjsip will now look for SRV records based on what transports are > > configured on the system". > > > > Does this mean there's now a way of doing it? Because > > https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip_wizard.conf.sample > > says: > > ; Hostnames must resolve to A, or CNAME records. > > ; SRV records are not currently supported. > > > > H... I'm confused! > > SRV support for inbound matching was added after that comment was written. > Identifying by IP address resolve a hostname down to all addresses (including > SRV) - not just a single one. > > Outgoing supports A, , SRV, and NAPTR automatically. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
Thanks, but... whoah! I think I just found a bug! As soon as I changed accepts_registrations = yes to sends_registrations = yes and did a pjsip reload, Asterisk crashed. I tried starting asterisk. Nothing. In the syslog: Jul 28 22:20:41 televox kernel: [ 50.728769] asterisk[1504]: segfault at 0 ip 7f4be3e00646 sp 7ffc32067388 error 4 in libc-2.27.so[7f4be3d4f000+1e7000] Jul 28 22:22:02 televox kernel: [ 132.413114] asterisk[1579]: segfault at 0 ip 7f62a9ba2646 sp 7ffc9215d408 error 4 in libc-2.27.so[7f62a9af1000+1e7000] Took that line back out, and Asterisk started again. Shall I file a bug? On Sat, 28 Jul 2018 at 21:55, Joshua Colp wrote: > > > > On Sat, Jul 28, 2018, at 3:06 PM, Jonathan H wrote: > > Using pjsip 2.7.2 on Asterisk 15.5 > > Really struggling to make sense of translating these old 1.8 SIP > > instructions into a neat pjsip_wizard conf suitable for 2018 > > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 > > > > In pjsip_wizard.conf, I have the following, which seems to get me > > registered, and it responds to an incoming call, but I always get > > this: > > > > [Jul 28 18:32:29] NOTICE[22492]: res_pjsip/pjsip_distributor.c:659 > > log_failed_request: Request 'INVITE' from '"demo" > > ' failed for 'x.x.x.x:5060' (callid: > > 5fa139428fef42d9bd0cd4063e10b047) - No matching endpoint found > > > > here's what I have in pjsip_wizard.conf > > > > [sip2sip] > > type = wizard > > sends_auth = yes > > accepts_registrations = yes > > transport = simpletrans > > outound_auth/username = myusern...@sip2sip.info > > outound_auth/password = password > > remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info > > endpoint/allow = alaw > > endpoint/context = fromsip2sip > > aor/max_contacts = 3 > > registration/contact_user = myusername > > outbound_proxy = proxy.sipthor.net > > endpoint/language=en_GB > > This is an ITSP trunk, you've configured it kind of as if it were a phone. > Instead of "accepts_registrations" you likely want "sends_registrations". > Asterisk needs to register to them. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any way of "flattening out" 2 channels back into one?
Last question for today, I promise!
The problem: In order to disconnect calls after x minutes, I need to do this:
[setup]
exten => setup,1,Answer()
same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
same =>
n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached)
same => n,Dial(Local/s@root/n,3,L(354:6))
same => n,Hangup()
[root]
exten => s,1,Verbose(1,Call to: ${CALLERID(name)} from: ${CALLERID(num)})
same => n,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1)
etc etc
Works well, but the result is it looks like there are 2 active calls
in the console. Is there any way of forcing the drop of a call after x
minutes without doing this "double dialling" business?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
On Sat, Jul 28, 2018, at 3:06 PM, Jonathan H wrote: > Using pjsip 2.7.2 on Asterisk 15.5 > Really struggling to make sense of translating these old 1.8 SIP > instructions into a neat pjsip_wizard conf suitable for 2018 > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 > > In pjsip_wizard.conf, I have the following, which seems to get me > registered, and it responds to an incoming call, but I always get > this: > > [Jul 28 18:32:29] NOTICE[22492]: res_pjsip/pjsip_distributor.c:659 > log_failed_request: Request 'INVITE' from '"demo" > ' failed for 'x.x.x.x:5060' (callid: > 5fa139428fef42d9bd0cd4063e10b047) - No matching endpoint found > > here's what I have in pjsip_wizard.conf > > [sip2sip] > type = wizard > sends_auth = yes > accepts_registrations = yes > transport = simpletrans > outound_auth/username = myusern...@sip2sip.info > outound_auth/password = password > remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info > endpoint/allow = alaw > endpoint/context = fromsip2sip > aor/max_contacts = 3 > registration/contact_user = myusername > outbound_proxy = proxy.sipthor.net > endpoint/language=en_GB This is an ITSP trunk, you've configured it kind of as if it were a phone. Instead of "accepts_registrations" you likely want "sends_registrations". Asterisk needs to register to them. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRV with pjsip on Asterisk 15.5: yes or no?
On Sat, Jul 28, 2018, at 5:42 PM, Jonathan H wrote: > I'm trying to configure sip2sip, which says: > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk > "Asterisk, is currently unable to handle more that one result for a > DNS SRV lookup, and the Asterisk configuration needed for getting it > work with the SIP2SIP service is not trivial" > > It then gives a complex multi-section workaround in SIP. I remember > reading there'd be the same issue with PJSIP, and then I found this > post in the Asterisk blog from 2016: > https://blogs.asterisk.org/2016/04/20/pjsip-dns-support/ which says: > "chan_pjsip will now look for SRV records based on what transports are > configured on the system". > > Does this mean there's now a way of doing it? Because > https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip_wizard.conf.sample > says: > ; Hostnames must resolve to A, or CNAME records. > ; SRV records are not currently supported. > > H... I'm confused! SRV support for inbound matching was added after that comment was written. Identifying by IP address resolve a hostname down to all addresses (including SRV) - not just a single one. Outgoing supports A, , SRV, and NAPTR automatically. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRV with pjsip on Asterisk 15.5: yes or no?
I'm trying to configure sip2sip, which says: http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk "Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it work with the SIP2SIP service is not trivial" It then gives a complex multi-section workaround in SIP. I remember reading there'd be the same issue with PJSIP, and then I found this post in the Asterisk blog from 2016: https://blogs.asterisk.org/2016/04/20/pjsip-dns-support/ which says: "chan_pjsip will now look for SRV records based on what transports are configured on the system". Does this mean there's now a way of doing it? Because https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip_wizard.conf.sample says: ; Hostnames must resolve to A, or CNAME records. ; SRV records are not currently supported. H... I'm confused! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)
I've not needed to do a dialplan reload for a while, so I don't know exactly which version is stopped working, but on 15.5, I'm not seeing ANY debug info at any debug level. So I'm not really sure how to find mistakes in the dialplan. This is all I get... how do I enable this debug mode to see the previous behaviour? Thanks asterisk -rvd (enters console) dialplan reload Dialplan reloaded. [...] -- pbx_config successfully loaded 125 contexts (enable debug for details). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29] NOTICE[22492]: res_pjsip/pjsip_distributor.c:659 log_failed_request: Request 'INVITE' from '"demo" ' failed for 'x.x.x.x:5060' (callid: 5fa139428fef42d9bd0cd4063e10b047) - No matching endpoint found here's what I have in pjsip_wizard.conf [sip2sip] type = wizard sends_auth = yes accepts_registrations = yes transport = simpletrans outound_auth/username = myusern...@sip2sip.info outound_auth/password = password remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info endpoint/allow = alaw endpoint/context = fromsip2sip aor/max_contacts = 3 registration/contact_user = myusername outbound_proxy = proxy.sipthor.net endpoint/language=en_GB in pjsip.conf [simpletrans] type = transport protocol = UDP bind = 0.0.0.0 [acl] type = acl deny = 0.0.0.0/0.0.0.0 ; next 3 are for sip2sip permit = 81.23.228.129 permit = 85.17.186.7 permit = 81.23.228.150 in extensions.conf, I've got a bit OTT and covered every possible base to match an endpoint! Every single item in the "to" or "from" header is accounted for somewhere, so why can't it find this endpoint? Would be really grateful. Thanks. extensions.conf [fromsip2sip] exten => _.,1,Verbose(answered) [myusername] exten => _.,1,Verbose(answered) [myusern...@sip2sip.info] exten => _.,1,Verbose(answered) [demo] exten => _.,1,Verbose(answered) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
