Re: [asterisk-users] Queue show : failed to extend from 240 to 327

[Richard Mudgett]

On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens 
wrote:

> Hello
>
> when I type on the Asterisk CLi 'queue show', I first get a list of my
> queues and then the following :
>
>
> failed to extend from 240 to 327
>



failed to extend from 240 to 334
>
>
> I could not find any information on this on the web, except this :
> https://issues.asterisk.org/jira/browse/ASTERISK-8828
>
> which is an old 'bug' that should have been fixed meanwhile.
>
> Any more thoughts on why I should be getting this message when asking
> information about queues (I don't see this message on any other command).
>

That message is a result of trying to build a string where the buffer is too
small to contain it.  I would expect that there is a truncated string in the
'queue show' output.

You haven't stated which Asterisk version you are using.  It may already be
fixed.

Richard
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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] Queue show : failed to extend from 240 to 327

[Jonas Kellens]

Hello

when I type on the Asterisk CLi 'queue show', I first get a list of my 
queues and then the following :



failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 323
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 323
failed to extend from 240 to 323
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 323
failed to extend from 240 to 334
failed to extend from 240 to 334
failed to extend from 240 to 334
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 334
failed to extend from 240 to 334


I could not find any information on this on the web, except this : 
https://issues.asterisk.org/jira/browse/ASTERISK-8828


which is an old 'bug' that should have been fixed meanwhile.

Any more thoughts on why I should be getting this message when asking 
information about queues (I don't see this message on any other command).




Kind regards


Jonas.



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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] Asterisk 13.11.2 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 13.11.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.11.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
  'REGISTER' failed (Reported by Dmitry Melekhov)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.2

Thank you for your continued support of Asterisk!

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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] How to get a list of DAHDI channels

[Carlos Chavez]

Anyone know an efficient way to get a list of the DAHDI channels?  
Is there an AMI or ARI variable to get a list of all the channels?



--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] Asterisk 13 PJSIP with Snom 710

[Madushan Geethanga]

thanks for the reply. if i config the extension in softphone it works fine.
but with snom its not working

Bet Regards,
Madushan

On Fri, Sep 9, 2016 at 10:31 PM, Madushan Geethanga  wrote:

> yes I have unchecked it.
>
> On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI 
> wrote:
>
>> Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :
>>
>>> Hi,
>>>
>>
>> If you're not using RTP encryption did you uncheck the option in your RTP
>> TAB from identity ?
>>
>>
>>> This is the log. ex dialling 0 from snom phone
>>>
>>>
>>> <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
>>>  --->
>>> INVITE sip:0@54.206.59.252 ;user=phone
>>> SIP/2.0
>>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
>>> From: "outburns00-nhvg5vjjn6-2001"
>>> >> >;tag=1bb809zgaa
>>> To: ;user=phone>
>>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>>> CSeq: 1 INVITE
>>> Max-Forwards: 70
>>> User-Agent: snom710/8.7.5.35 
>>> Contact: >> >;reg-id=1
>>>
>>> X-Serialnumber: 000413747C96
>>> P-Key-Flags: resolution="31x13", keys="4"
>>> Accept: application/sdp
>>> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
>>> PRACK, MESSAGE, INFO, UPDATE
>>> Allow-Events: talk, hold, refer, call-info
>>> Supported: timer, 100rel, replaces, from-change
>>> Session-Expires: 3600
>>> Min-SE: 90
>>> Content-Type: application/sdp
>>> Content-Length: 405
>>>
>>> v=0
>>> o=root 2136927789 2136927789 IN IP4 192.168.2.28
>>> s=call
>>> c=IN IP4 123.231.72.210
>>> t=0 0
>>> m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101
>>> a=rtpmap:9 G722/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:99 G726-32/8000
>>> a=rtpmap:112 AAL2-G726-32/8000
>>> a=rtpmap:18 G729/8000
>>> a=fmtp:18 annexb=no
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> <--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878
>>>  --->
>>> SIP/2.0 401 Unauthorized
>>> Via: SIP/2.0/UDP
>>> 123.231.72.210:45835;rport=33878;received=123.231.72.210;bra
>>> nch=z9hG4bK-bskkkx1t5bas
>>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>>> From: "outburns00-nhvg5vjjn6-2001"
>>> >> >;tag=1bb809zgaa
>>> To: >> ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
>>> CSeq: 1 INVITE
>>> WWW-Authenticate: Digest
>>> realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0c
>>> cea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"
>>> Server: Asterisk PBX certified/13.8-cert2
>>> Content-Length:  0
>>>
>>>
>>> <--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878
>>>  --->
>>> ACK sip:0@54.206.59.252 ;user=phone
>>> SIP/2.0
>>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
>>> From: "outburns00-nhvg5vjjn6-2001"
>>> >> >;tag=1bb809zgaa
>>> To: >> ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
>>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>>> CSeq: 1 ACK
>>> Max-Forwards: 70
>>> User-Agent: snom710/8.7.5.35 
>>> Contact: >> >;reg-id=1
>>> Content-Length: 0
>>>
>>>
>>> Best Regards,
>>> Madushan
>>>
>>>
>>>
>>> On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
>>> > wrote:
>>>
>>> Hi,
>>>
>>> I'm trying to setup snom 710 phone with asterisk 13 with PJSIP.
>>> inbound is working fine but i cannot dial out. i don't hear anything
>>> on the phone and asterisk CLI also does not show anything. my config
>>> is. please advice.
>>>
>>> [2001]
>>> type=endpoint
>>> context=out-local
>>> disallow=all
>>> allow=ulaw
>>> allow=alaw
>>> transport=system-udp
>>> auth=2001
>>> aors=2001
>>> direct_media=no
>>> rtp_symmetric=yes
>>> force_rport=yes
>>> allow=alaw
>>> allow=speex
>>> allow=speex16
>>> allow=speex32
>>> allow=gsm
>>>
>>>
>>> [2001]
>>> type=aor
>>> qualify_frequency=5000
>>> authenticate_qualify=yes

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

[Madushan Geethanga]

yes I have unchecked it.

On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI 
wrote:

> Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :
>
>> Hi,
>>
>
> If you're not using RTP encryption did you uncheck the option in your RTP
> TAB from identity ?
>
>
>> This is the log. ex dialling 0 from snom phone
>>
>>
>> <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
>>  --->
>> INVITE sip:0@54.206.59.252 ;user=phone
>> SIP/2.0
>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
>> From: "outburns00-nhvg5vjjn6-2001"
>> > >;tag=1bb809zgaa
>> To: ;user=phone>
>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>> CSeq: 1 INVITE
>> Max-Forwards: 70
>> User-Agent: snom710/8.7.5.35 
>> Contact: > >;reg-id=1
>>
>> X-Serialnumber: 000413747C96
>> P-Key-Flags: resolution="31x13", keys="4"
>> Accept: application/sdp
>> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
>> PRACK, MESSAGE, INFO, UPDATE
>> Allow-Events: talk, hold, refer, call-info
>> Supported: timer, 100rel, replaces, from-change
>> Session-Expires: 3600
>> Min-SE: 90
>> Content-Type: application/sdp
>> Content-Length: 405
>>
>> v=0
>> o=root 2136927789 2136927789 IN IP4 192.168.2.28
>> s=call
>> c=IN IP4 123.231.72.210
>> t=0 0
>> m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101
>> a=rtpmap:9 G722/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:99 G726-32/8000
>> a=rtpmap:112 AAL2-G726-32/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>> a=sendrecv
>>
>> <--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878
>>  --->
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 123.231.72.210:45835;rport=33878;received=123.231.72.210;bra
>> nch=z9hG4bK-bskkkx1t5bas
>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>> From: "outburns00-nhvg5vjjn6-2001"
>> > >;tag=1bb809zgaa
>> To: > ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
>> CSeq: 1 INVITE
>> WWW-Authenticate: Digest
>> realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0c
>> cea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"
>> Server: Asterisk PBX certified/13.8-cert2
>> Content-Length:  0
>>
>>
>> <--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878
>>  --->
>> ACK sip:0@54.206.59.252 ;user=phone SIP/2.0
>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
>> From: "outburns00-nhvg5vjjn6-2001"
>> > >;tag=1bb809zgaa
>> To: > ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>> CSeq: 1 ACK
>> Max-Forwards: 70
>> User-Agent: snom710/8.7.5.35 
>> Contact: > >;reg-id=1
>> Content-Length: 0
>>
>>
>> Best Regards,
>> Madushan
>>
>>
>>
>> On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
>> > wrote:
>>
>> Hi,
>>
>> I'm trying to setup snom 710 phone with asterisk 13 with PJSIP.
>> inbound is working fine but i cannot dial out. i don't hear anything
>> on the phone and asterisk CLI also does not show anything. my config
>> is. please advice.
>>
>> [2001]
>> type=endpoint
>> context=out-local
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> transport=system-udp
>> auth=2001
>> aors=2001
>> direct_media=no
>> rtp_symmetric=yes
>> force_rport=yes
>> allow=alaw
>> allow=speex
>> allow=speex16
>> allow=speex32
>> allow=gsm
>>
>>
>> [2001]
>> type=aor
>> qualify_frequency=5000
>> authenticate_qualify=yes
>> max_contacts=1
>> remove_existing=yes
>>
>> [2001]
>> type=auth
>> auth_type=userpass
>> password=test
>> username=test
>>
>> Best Regards,
>> Madushan
>>
>>
>>
>>
>>
> --
> _
> -- Bandwidth and Colocation 

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

[Administrator TOOTAI]

Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :

Hi,


If you're not using RTP encryption did you uncheck the option in your 
RTP TAB from identity ?




This is the log. ex dialling 0 from snom phone


<--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
 --->
INVITE sip:0@54.206.59.252 ;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001"
>;tag=1bb809zgaa
To: ;user=phone>
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 
Contact: >;reg-id=1
X-Serialnumber: 000413747C96
P-Key-Flags: resolution="31x13", keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600
Min-SE: 90
Content-Type: application/sdp
Content-Length: 405

v=0
o=root 2136927789 2136927789 IN IP4 192.168.2.28
s=call
c=IN IP4 123.231.72.210
t=0 0
m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878
 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
123.231.72.210:45835;rport=33878;received=123.231.72.210;branch=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
From: "outburns00-nhvg5vjjn6-2001"
>;tag=1bb809zgaa
To: ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
CSeq: 1 INVITE
WWW-Authenticate: Digest
realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0ccea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"
Server: Asterisk PBX certified/13.8-cert2
Content-Length:  0


<--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878
 --->
ACK sip:0@54.206.59.252 ;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001"
>;tag=1bb809zgaa
To: ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 
Contact: >;reg-id=1
Content-Length: 0


Best Regards,
Madushan



On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
> wrote:

Hi,

I'm trying to setup snom 710 phone with asterisk 13 with PJSIP.
inbound is working fine but i cannot dial out. i don't hear anything
on the phone and asterisk CLI also does not show anything. my config
is. please advice.

[2001]
type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
aors=2001
direct_media=no
rtp_symmetric=yes
force_rport=yes
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm


[2001]
type=aor
qualify_frequency=5000
authenticate_qualify=yes
max_contacts=1
remove_existing=yes

[2001]
type=auth
auth_type=userpass
password=test
username=test

Best Regards,
Madushan






--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] Asterisk 13 PJSIP with Snom 710

[Madushan Geethanga]

Hi,

This is the log. ex dialling 0 from snom phone


<--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878 --->
INVITE sip:0@54.206.59.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001" <
sip:outburns00-nhvg5vjjn6-2001@54.206.59.252>;tag=1bb809zgaa
To: 
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35
Contact: ;reg-id=1
X-Serialnumber: 000413747C96
P-Key-Flags: resolution="31x13", keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600
Min-SE: 90
Content-Type: application/sdp
Content-Length: 405

v=0
o=root 2136927789 2136927789 IN IP4 192.168.2.28
s=call
c=IN IP4 123.231.72.210
t=0 0
m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 123.231.72.210:45835
;rport=33878;received=123.231.72.210;branch=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
From: "outburns00-nhvg5vjjn6-2001" <
sip:outburns00-nhvg5vjjn6-2001@54.206.59.252>;tag=1bb809zgaa
To: ;tag=z9hG4bK-bskkkx1t5bas
CSeq: 1 INVITE
WWW-Authenticate: Digest
realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0ccea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"
Server: Asterisk PBX certified/13.8-cert2
Content-Length:  0


<--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878 --->
ACK sip:0@54.206.59.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001" <
sip:outburns00-nhvg5vjjn6-2001@54.206.59.252>;tag=1bb809zgaa
To: ;tag=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom710/8.7.5.35
Contact: ;reg-id=1
Content-Length: 0


Best Regards,
Madushan



On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga 
wrote:

> Hi,
>
> I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is
> working fine but i cannot dial out. i don't hear anything on the phone and
> asterisk CLI also does not show anything. my config is. please advice.
>
> [2001]
> type=endpoint
> context=out-local
> disallow=all
> allow=ulaw
> allow=alaw
> transport=system-udp
> auth=2001
> aors=2001
> direct_media=no
> rtp_symmetric=yes
> force_rport=yes
> allow=alaw
> allow=speex
> allow=speex16
> allow=speex32
> allow=gsm
>
>
> [2001]
> type=aor
> qualify_frequency=5000
> authenticate_qualify=yes
> max_contacts=1
> remove_existing=yes
>
> [2001]
> type=auth
> auth_type=userpass
> password=test
> username=test
>
> Best Regards,
> Madushan
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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] Asterisk 13 PJSIP with Snom 710

[Madushan Geethanga]

Hi,

I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is
working fine but i cannot dial out. i don't hear anything on the phone and
asterisk CLI also does not show anything. my config is. please advice.

[2001]
type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
aors=2001
direct_media=no
rtp_symmetric=yes
force_rport=yes
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm


[2001]
type=aor
qualify_frequency=5000
authenticate_qualify=yes
max_contacts=1
remove_existing=yes

[2001]
type=auth
auth_type=userpass
password=test
username=test

Best Regards,
Madushan
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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] (no subject)

[Madushan Geethanga]

Hi,

Im trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is
working fine but i cannot dial out. i don't hear anything on the phone and
asterisk CLI also does not show anything. my config is. please advice.

[2001]
type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
aors=2001
direct_media=no
rtp_symmetric=yes
force_rport=yes
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm


[2001]
type=aor
qualify_frequency=5000
authenticate_qualify=yes
max_contacts=1
remove_existing=yes

[2001]
type=auth
auth_type=userpass
password=test
username=test

Best Regards,
Madushan
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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] Asterisk 13 and WebRTC

[Annus Fictus]

Hello,

I mean a working configuration (SIP o PJSIP) without patches or code 
corrections.


Thank you

Regards


El 09/09/2016 a las 03:47, marek cervenka escribió:

using in production

last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search 
pjsip conf) + sipml5 version from roginvs


https://github.com/DoubangoTelecom/sipml5/pull/238


Dne 08/09/2016 v 23:36 Annus Fictus napsal(a):

Hello list,

before to lost my time, I'd like know if someone have a WebRTC 
working configuration on Asterisk 13.11.0 SIP or PJSIP channel.


Thank you

Regards









--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] [SOLVED] Re: Feature Request: what about "core stop panic" ?

[Olivier]

2016-09-09 13:24 GMT+02:00 Jacek Konieczny :

> On 2016-09-09 11:34, Olivier wrote:
>
>> Adding an /etc/sysctl.d/foobar.conf file with the bellow content allowed
>> me to at last produce core dump files (in /var/tmp directory), even if
>> asterisk is run by asterisk user (and by root).
>> I choosed this /var/tmp directory to make sure core dumps are not erased
>> after a reboot and because this directory is "world-writable".
>> To trigger core dumping, previously recommended "pkill -SEGV asterisk"
>> was used.
>>
>> /etc/sysctl.d/foobar.conf content is simply:
>> kernel.core_pattern=/var/tmp/core.%e.%t
>>
>> Maybe taming systemd to consider /var/lib/asterisk as a current
>> directory when running asterisk daemon would be a better solution ?
>>
>> Maybe Asterisk or more generally long running daemons, should warn when
>> they are run with "-g option" and from a current directory where it
>> can't write any file (or any file matching core pattern) ?
>> Maybe this is already done but I overlooked it or looked in the wrong
>> place ?
>>
>
>
> Why not just use the systemd journal and coredumpctl for core files
> management?  systemd solves that quite well.
>

Do you have any pointer showing how systemd journal or coredumpctl can help
?
How can any of those help to specify the where core files are dumped ?



>
> Jacek
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>  http://www.asterisk.org/community/astricon-user-conference
>
> 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 --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

[Dmitry Melekhov]

09.09.2016 15:18, Joshua Colp пишет:

Dmitry Melekhov wrote:

09.09.2016 14:08, Dmitry Melekhov пишет:






And, as I already said, there was no such messages while using asterisk
13.10.
I'll open bug report.


I was mistaken and it is indeed a bug. I've got a fix up but even 
without the fix it isn't blocking requests as you've noticed.



Just compiled with fix, thank you!



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] [SOLVED] Re: Feature Request: what about "core stop panic" ?

[Jacek Konieczny]

On 2016-09-09 11:34, Olivier wrote:

Adding an /etc/sysctl.d/foobar.conf file with the bellow content allowed
me to at last produce core dump files (in /var/tmp directory), even if
asterisk is run by asterisk user (and by root).
I choosed this /var/tmp directory to make sure core dumps are not erased
after a reboot and because this directory is "world-writable".
To trigger core dumping, previously recommended "pkill -SEGV asterisk"
was used.

/etc/sysctl.d/foobar.conf content is simply:
kernel.core_pattern=/var/tmp/core.%e.%t

Maybe taming systemd to consider /var/lib/asterisk as a current
directory when running asterisk daemon would be a better solution ?

Maybe Asterisk or more generally long running daemons, should warn when
they are run with "-g option" and from a current directory where it
can't write any file (or any file matching core pattern) ?
Maybe this is already done but I overlooked it or looked in the wrong
place ?



Why not just use the systemd journal and coredumpctl for core files 
management?  systemd solves that quite well.


Jacek

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

[Joshua Colp]

Dmitry Melekhov wrote:

09.09.2016 14:08, Dmitry Melekhov пишет:






And, as I already said, there was no such messages while using asterisk
13.10.
I'll open bug report.


I was mistaken and it is indeed a bug. I've got a fix up but even 
without the fix it isn't blocking requests as you've noticed.


--
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 --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

[Dmitry Melekhov]

09.09.2016 14:13, Joshua Colp пишет:

Dmitry Melekhov wrote:

09.09.2016 14:08, Dmitry Melekhov пишет:

09.09.2016 13:45, Joshua Colp пишет:

Dmitry Melekhov wrote:

Hello!


Upgraded 13.10 to 13.11.1 today and now I see messages in log:


[Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request
'REGISTER' from '"3563" ' failed for
'192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No
matching endpoint found


or

[Sep 9 12:56:14] NOTICE[10163] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"3567" ' failed for
'192.168.32.108:5060' (callid: 0_2410349837@192.168.32.108) - No
matching endpoint found


No complains from users though


Could you tell me what is this?


It means that the request could not be matched to an endpoint, just
like the message says :D this could be because endpoints in
pjsip.conf could not be loaded or if from a database, they couldn't
be loaded from there.


These endpoints are already registered



And, as I already said, there was no such messages while using asterisk
13.10.
I'll open bug report.


I don't believe this is actually a bug. 


I just reported :-)
Are the messages happening constantly? 


No, but quite often:

[Sep  9 13:48:41] NOTICE[16626] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"3567" ' failed for 
'192.168.32.108:5060' (callid: 0_806567588@192.168.32.108) - No matching 
endpoint

 found
[Sep  9 13:48:41] NOTICE[16626] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"3567" ' failed for 
'192.168.32.108:5060' (callid: 0_806567588@192.168.32.108) - No matching 
endpoint

 found
[Sep  9 13:48:48] NOTICE[16626] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"3566" ' failed for 
'192.168.32.111:5060' (callid: 0_3498933275@192.168.32.111) - No 
matching endpo

int found
[Sep  9 13:48:48] NOTICE[16626] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"3566" ' failed for 
'192.168.32.111:5060' (callid: 0_3498933275@192.168.32.111) - No 
matching endpo

int found
[Sep  9 14:03:16] NOTICE[18406] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"3563" ' failed for 
'192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No 
matching endpo

int found
[Sep  9 14:03:16] NOTICE[18406] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"3563" ' failed for 
'192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No 
matching endpo

int found
[Sep  9 14:11:29] NOTICE[19410] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"3560" ' failed for 
'192.168.32.106:5060' (callid: 0_4177963995@192.168.32.106) - No 
matching endpo

int found
[Sep  9 14:11:29] NOTICE[19410] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"3560" ' failed for 
'192.168.32.106:5060' (callid: 0_4177963995@192.168.32.106) - No 
matching endpo

int found
[Sep  9 14:13:56] NOTICE[19705] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"3560" ' failed for 
'192.168.32.106:5060' (callid: 0_1192076852@192.168.32.106) - No 
matching endpoin

t found
[Sep  9 14:13:56] NOTICE[19705] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"3560" ' failed for 
'192.168.32.106:5060' (callid: 0_1192076852@192.168.32.106) - No 
matching endpoin

t found
[Sep  9 14:14:11] NOTICE[19705] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"3560" ' failed for 
'192.168.32.106:5060' (callid: 0_3557536029@192.168.32.106) - No 
matching endpoin

t found
[Sep  9 14:14:11] NOTICE[19705] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"3560" ' failed for 
'192.168.32.106:5060' (callid: 0_3557536029@192.168.32.106) - No 
matching endpoin

t found
[Sep  9 14:15:27] NOTICE[19705] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"3560" ' failed for 
'192.168.32.106:5060' (callid: 0_2728336985@192.168.32.106) - No 
matching endpoin

t found
[Sep  9 14:15:27] NOTICE[19705] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"3560" ' failed for 
'192.168.32.106:5060' (callid: 0_2728336985@192.168.32.106) - No 
matching endpoin

t found


Are there other successful calls and registrations?

Yes.

Does pjsip show endpoints show the endpoints?


Yes.

 Endpoint:  3560/3560 In use1 of inf
 InAuth:  3560/3560
Aor:  3560   1
  Contact:  3560/sip:3560@192.168.32.106:5060 4d5f4b26a2 
Unknown nan
  Transport:  transport-udp udp  0  0 
192.168.32.254:5060

Channel: PJSIP/3560-0042/Dial Up00:01:19
Exten: 984922778704  CLCID: "" <>

Others are OK too...




--

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

[Joshua Colp]

Dmitry Melekhov wrote:

09.09.2016 14:08, Dmitry Melekhov пишет:

09.09.2016 13:45, Joshua Colp пишет:

Dmitry Melekhov wrote:

Hello!


Upgraded 13.10 to 13.11.1 today and now I see messages in log:


[Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request
'REGISTER' from '"3563" ' failed for
'192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No
matching endpoint found


or

[Sep 9 12:56:14] NOTICE[10163] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"3567" ' failed for
'192.168.32.108:5060' (callid: 0_2410349837@192.168.32.108) - No
matching endpoint found


No complains from users though


Could you tell me what is this?


It means that the request could not be matched to an endpoint, just
like the message says :D this could be because endpoints in
pjsip.conf could not be loaded or if from a database, they couldn't
be loaded from there.


These endpoints are already registered



And, as I already said, there was no such messages while using asterisk
13.10.
I'll open bug report.


I don't believe this is actually a bug. Are the messages happening 
constantly? Are there other successful calls and registrations? Does 
pjsip show endpoints show the endpoints?


--
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 --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

[Dmitry Melekhov]

09.09.2016 14:08, Dmitry Melekhov пишет:

09.09.2016 13:45, Joshua Colp пишет:

Dmitry Melekhov wrote:

Hello!


Upgraded 13.10 to 13.11.1 today and now I see messages in log:


[Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request
'REGISTER' from '"3563" ' failed for
'192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No
matching endpoint found


or

[Sep 9 12:56:14] NOTICE[10163] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"3567" ' failed for
'192.168.32.108:5060' (callid: 0_2410349837@192.168.32.108) - No
matching endpoint found


No complains from users though


Could you tell me what is this?


It means that the request could not be matched to an endpoint, just 
like the message says :D this could be because endpoints in 
pjsip.conf could not be loaded or if from a database, they couldn't 
be loaded from there.



These endpoints are already registered



And, as  I already said, there was no such messages while using asterisk 
13.10.

I'll open bug report.



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

[Dmitry Melekhov]

09.09.2016 13:45, Joshua Colp пишет:

Dmitry Melekhov wrote:

Hello!


Upgraded 13.10 to 13.11.1 today and now I see messages in log:


[Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request
'REGISTER' from '"3563" ' failed for
'192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No
matching endpoint found


or

[Sep 9 12:56:14] NOTICE[10163] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"3567" ' failed for
'192.168.32.108:5060' (callid: 0_2410349837@192.168.32.108) - No
matching endpoint found


No complains from users though


Could you tell me what is this?


It means that the request could not be matched to an endpoint, just 
like the message says :D this could be because endpoints in pjsip.conf 
could not be loaded or if from a database, they couldn't be loaded 
from there.



These endpoints are already registered



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

[Joshua Colp]

Dmitry Melekhov wrote:

Hello!


Upgraded 13.10 to 13.11.1 today and now I see messages in log:


[Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request
'REGISTER' from '"3563" ' failed for
'192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No
matching endpoint found


or

[Sep 9 12:56:14] NOTICE[10163] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"3567" ' failed for
'192.168.32.108:5060' (callid: 0_2410349837@192.168.32.108) - No
matching endpoint found


No complains from users though


Could you tell me what is this?


It means that the request could not be matched to an endpoint, just like 
the message says :D this could be because endpoints in pjsip.conf could 
not be loaded or if from a database, they couldn't be loaded from there.


--
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 --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] [SOLVED] Re: Feature Request: what about "core stop panic" ?

[Olivier]

Adding an /etc/sysctl.d/foobar.conf file with the bellow content allowed me
to at last produce core dump files (in /var/tmp directory), even if
asterisk is run by asterisk user (and by root).
I choosed this /var/tmp directory to make sure core dumps are not erased
after a reboot and because this directory is "world-writable".
To trigger core dumping, previously recommended "pkill -SEGV asterisk" was
used.

/etc/sysctl.d/foobar.conf content is simply:
kernel.core_pattern=/var/tmp/core.%e.%t

Maybe taming systemd to consider /var/lib/asterisk as a current directory
when running asterisk daemon would be a better solution ?

Maybe Asterisk or more generally long running daemons, should warn when
they are run with "-g option" and from a current directory where it can't
write any file (or any file matching core pattern) ?
Maybe this is already done but I overlooked it or looked in the wrong place
?

Anyway, thank you very much to all.

2016-09-08 16:22 GMT+02:00 Olivier :

> I think were getting closer:
>
> I did:
> - I edited /etc/default/asterisk to include :
> AST_USER="root"
> AST_GROUP="root"
>
> # systemctl daemon-reload
> # systemctl start asterisk
> # ps aux | grep asterisk
> root  3602  7.1  2.5  60332 26012 ?Ssl  16:00   0:03
> /usr/sbin/asterisk -U root -G root -g
> # rasterisk
> # pkill -SEGV asterisk
>
> Then console showed:
> Segmentation error (core dumped)
>
> and a /core file is created
>
>
> I also tried with asterisk run asterisk user and group and looked for any
> newly created (find / -cmin -2) : nothing looking like a core file, though
> I might overlooked it.
>
> So, it really look like a configuration issue: "how to specify, with
> asterisk and systemd among others, where core files are dumped ?".
>
> I'll read again this thread advices and report back here.
>
> 2016-09-07 21:44 GMT+02:00 George Joseph :
>
>>
>>
>> On Wed, Sep 7, 2016 at 11:15 AM, Olivier  wrote:
>>
>>>
>>>
>>> 2016-09-06 17:48 GMT+02:00 Tzafrir Cohen :
>>>
 On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote:
 > On Tue, Sep 6, 2016 at 1:55 AM, Olivier  wrote:

 > > Where should core file be created when Asterisk is run as a daemon
 by
 > > asterisk user and group ?
 > > Is there a setting I can use to specify the directory used (so that
 we can
 > > make sure appropriate ownership is set) ?
 > >
 >
 > "$ sysctl kernel.core_pattern" will show you where core files are
 written.
 > For Asterisk to produce the core file, it has to be started with the
 '-g'
 > option so make sure your asterisk.service file is adding the option.

>>>
>>> My asterisk.service is :
>>>
>>> # cat /run/systemd/generator.late/asterisk.service
>>> # Automatically generated by systemd-sysv-generator
>>>
>>> [Unit]
>>> SourcePath=/etc/init.d/asterisk
>>> Description=LSB: Asterisk PBX
>>> Before=runlevel2.target runlevel3.target runlevel4.target
>>> runlevel5.target shutdown.target
>>> After=network-online.target systemd-journald-dev-log.socket
>>> nss-lookup.target local-fs.target remote-fs.target dahdi.service
>>> misdn.service lcr.service wanrouter.service mysql.service postgresql.service
>>> Wants=network-online.target
>>> Conflicts=shutdown.target
>>>
>>> [Service]
>>> Type=forking
>>> Restart=no
>>> TimeoutSec=5min
>>> IgnoreSIGPIPE=no
>>> KillMode=process
>>> GuessMainPID=no
>>> RemainAfterExit=yes
>>> SysVStartPriority=2
>>> ExecStart=/etc/init.d/asterisk start
>>> ExecStop=/etc/init.d/asterisk stop
>>> ExecReload=/etc/init.d/asterisk reload
>>>
>>>
>>> My /etc/init.d/asterisk file is sourcing my /etc/default/asterisk which
>>> includes an (uncommented) line COREDUMP=yes.
>>> I also have
>>> # grep core /etc/asterisk/asterisk.conf
>>> dumpcore = yes
>>>
>>> make menuselect shows:
>>> [*] DONT_OPTIMIZE
>>> [*] COMPILE_DOUBLE
>>> [ ] DEBUG_THREADS
>>> [*] LOADABLE_MODULES
>>> [ ] DEBUG_FD_LEAKS
>>> [*] BETTER_BACKTRACES
>>> [ ] LOTS_OF_SPANS
>>> [ ] MALLOC_DEBUG
>>> [ ] DEBUG_CHAOS
>>> [*] BUILD_NATIVE
>>> --- Extended ---
>>> [ ] REF_DEBUG
>>> [ ] AO2_DEBUG
>>> [ ] STATIC_BUILD
>>> XXX REBUILD_PARSERS
>>> [ ] LOW_MEMORY
>>> [ ] DISABLE_INLINE
>>> [*] OPTIONAL_API
>>> XXX USE_HOARD_ALLOCATOR
>>> [ ] RADIO_RELAX
>>> [ ] G711_NEW_ALGORITHM
>>> < > G711_REDUCED_BRANCHING
>>> < > TEST_CODING_TABLES
>>> < > TEST_TANDEM_TRANSCODING
>>> [ ] ADDRESS_SANITIZER
>>> [ ] THREAD_SANITIZER
>>> [ ] LEAK_SANITIZER
>>> [ ] UNDEFINED_SANITIZER
>>> [ ] BUSYDETECT_TONEONLY
>>> [ ] BUSYDETECT_COMPARE_TONE_AND_SILENCE
>>> [ ] BUSYDETECT_DEBUG
>>> [ ] INTEGER_CALLERID
>>>
>>>
>>> Is there a way to read Compiler Flags from a running system without
>>> looking at source file directory ?
>>>
>>> If someone forgets to select appropriate Compiler Flags, does it prevent
>>> Asterisk to produce a core file (even an empty one) ?
>>>
>>
>> If -g is specified a core file should 

[asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

[Dmitry Melekhov]

Hello!


Upgraded 13.10 to 13.11.1 today and now I see messages in log:


[Sep  9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 
'REGISTER' from '"3563" ' failed for 
'192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No 
matching endpoint found



or

[Sep  9 12:56:14] NOTICE[10163] res_pjsip/pjsip_distributor.c: Request 
'INVITE' from '"3567" ' failed for 
'192.168.32.108:5060' (callid: 0_2410349837@192.168.32.108) - No 
matching endpoint found



No complains from users though


Could you tell me what is this?


Thank you!



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] Asterisk 13 and WebRTC

[marek cervenka]

using in production

last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search 
pjsip conf) + sipml5 version from roginvs


https://github.com/DoubangoTelecom/sipml5/pull/238


Dne 08/09/2016 v 23:36 Annus Fictus napsal(a):

Hello list,

before to lost my time, I'd like know if someone have a WebRTC working 
configuration on Asterisk 13.11.0 SIP or PJSIP channel.


Thank you

Regards






--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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] Asterisk 13 and WebRTC

[Yuriy Gorlichenko]

Hi. It have big audio delay because using extenral ICE servers.
Better to use kamailio/opensips + rpenigne infront

2016-09-09 0:36 GMT+03:00 Annus Fictus :

> Hello list,
>
> before to lost my time, I'd like know if someone have a WebRTC working
> configuration on Asterisk 13.11.0 SIP or PJSIP channel.
>
> Thank you
>
> Regards
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>  http://www.asterisk.org/community/astricon-user-conference
>
> 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 --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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] Different cachertclasses setting for different Music on Hold

[Leandro Dardini]

As you know, there is the following settings

[general]
cachertclasses=yes ; use 1 instance of moh class for all users who are
using it,
; decrease consumable cpu cycles and memory
; disabled by default

It allows to use a single instance of MOH for all users. I'd like to have
this setting different for each Music on Hold class.

Is it possible?

Leandro
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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