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Jonathan H

2018-Jul-28 18:06 UTC

[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"
<sip:myusername at sip2sip.info>' 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 = myusername at 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)

    [myusername at sip2sip.info]
    exten => _.,1,Verbose(answered)

    [demo]
    exten => _.,1,Verbose(answered)

Joshua Colp

2018-Jul-28 20:55 UTC

head link

[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"
> <sip:myusername at sip2sip.info>' 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 = myusername at 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

Jonathan H

2018-Jul-28 21:27 UTC

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[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 00007f4be3e00646 sp 00007ffc32067388 error 4 in
libc-2.27.so[7f4be3d4f000+1e7000]
Jul 28 22:22:02 televox kernel: [  132.413114] asterisk[1579]:
segfault at 0 ip 00007f62a9ba2646 sp 00007ffc9215d408 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 <jcolp at digium.com>
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"
> > <sip:myusername at sip2sip.info>' 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 = myusername at 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
>
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> To UNSUBSCRIBE or update options visit:
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asterisk users - Jul 2018 - Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?

[asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?

[asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?

[asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?