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[asterisk-users] Asterisk 13 PJSIP with Snom 710


 
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PostPosted: Fri Sep 09, 2016 11:23 am    Post subject: [asterisk-users] Asterisk 13 PJSIP with Snom 710 Reply with quote

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
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mgliyanage.rc at gmail...
Guest





PostPosted: Fri Sep 09, 2016 11:32 am    Post subject: [asterisk-users] Asterisk 13 PJSIP with Snom 710 Reply with quote

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 ([email]sip%3A0@54.206.59.252[/email]);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 ([email]sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email]);user=phone>
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35
Contact: <sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>;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 ([email]sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email]);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 ([email]sip%3A0@54.206.59.252[/email]);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 ([email]sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email]);user=phone>;tag=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom710/8.7.5.35
Contact: <sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>;reg-id=1
Content-Length: 0



Best Regards,

Madushan






On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga <mgliyanage.rc@gmail.com (mgliyanage.rc@gmail.com)> wrote:
Quote:
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
Back to top
admin at tootai.net
Guest





PostPosted: Fri Sep 09, 2016 11:57 am    Post subject: [asterisk-users] Asterisk 13 PJSIP with Snom 710 Reply with quote

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

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

Quote:

This is the log. ex dialling 0 from snom phone


<--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
<http://123.231.72.210:33878> --->
INVITE sip:0@54.206.59.252 <mailto:sip%3A0@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
<mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252>>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 <mailto:sip%3A0@54.206.59.252>;user=phone>
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 <http://8.7.5.35>
Contact: <sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835
<http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;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
<http://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
<mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252>>;tag=1bb809zgaa
To: <sip:0@54.206.59.252
<mailto:sip%3A0@54.206.59.252>;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
<http://123.231.72.210:33878> --->
ACK sip:0@54.206.59.252 <mailto:sip%3A0@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
<mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252>>;tag=1bb809zgaa
To: <sip:0@54.206.59.252
<mailto:sip%3A0@54.206.59.252>;user=phone>;tag=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 <http://8.7.5.35>
Contact: <sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835
<http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;reg-id=1
Content-Length: 0


Best Regards,
Madushan



On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
<mgliyanage.rc@gmail.com <mailto:mgliyanage.rc@gmail.com>> 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
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mgliyanage.rc at gmail...
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PostPosted: Fri Sep 09, 2016 12:01 pm    Post subject: [asterisk-users] Asterisk 13 PJSIP with Snom 710 Reply with quote

yes I have unchecked it.


On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI <admin@tootai.net (admin@tootai.net)> wrote:
Quote:
Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :
Quote:
Hi,

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

Quote:

This is the log. ex dialling 0 from snom phone


<--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
<http://123.231.72.210:33878> --->
INVITE sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email]) <mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;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 ([email]sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])
<mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252 ([email]sip%253Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])>>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email]) <mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;user=phone>
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 <http://8.7.5.35>
Contact: <sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835
<http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;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 [url=tel:2136927789]2136927789[/url] [url=tel:2136927789]2136927789[/url] 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


<http://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 ([email]sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])
<mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252 ([email]sip%253Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])>>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email])
<mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;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
<http://123.231.72.210:33878> --->
ACK sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email]) <mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;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 ([email]sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])
<mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252 ([email]sip%253Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])>>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email])
<mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;user=phone>;tag=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 <http://8.7.5.35>
Contact: <sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835
<http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;reg-id=1
Content-Length: 0


Best Regards,
Madushan



On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
<mgliyanage.rc@gmail.com (mgliyanage.rc@gmail.com) <mailto:mgliyanage.rc@gmail.com (mgliyanage.rc@gmail.com)>> 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
Back to top
mgliyanage.rc at gmail...
Guest





PostPosted: Fri Sep 09, 2016 12:03 pm    Post subject: [asterisk-users] Asterisk 13 PJSIP with Snom 710 Reply with quote

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 <mgliyanage.rc@gmail.com (mgliyanage.rc@gmail.com)> wrote:
Quote:
yes I have unchecked it.


On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI <admin@tootai.net (admin@tootai.net)> wrote:
Quote:
Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :
Quote:
Hi,

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

Quote:

This is the log. ex dialling 0 from snom phone


<--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
<http://123.231.72.210:33878> --->
INVITE sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email]) <mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;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 ([email]sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])
<mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252 ([email]sip%253Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])>>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email]) <mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;user=phone>
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 <http://8.7.5.35>
Contact: <sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835
<http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;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 [url=tel:2136927789]2136927789[/url] [url=tel:2136927789]2136927789[/url] 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


<http://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 ([email]sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])
<mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252 ([email]sip%253Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])>>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email])
<mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;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
<http://123.231.72.210:33878> --->
ACK sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email]) <mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;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 ([email]sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])
<mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252 ([email]sip%253Aoutburns00-nhvg5vjjn6-2001@54.206.59.252[/email])>>;tag=1bb809zgaa
To: <sip:0@54.206.59.252 ([email]sip%3A0@54.206.59.252[/email])
<mailto:sip%3A0@54.206.59.252 ([email]sip%253A0@54.206.59.252[/email])>;user=phone>;tag=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 <http://8.7.5.35>
Contact: <sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835
<http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;reg-id=1
Content-Length: 0


Best Regards,
Madushan



On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
<mgliyanage.rc@gmail.com (mgliyanage.rc@gmail.com) <mailto:mgliyanage.rc@gmail.com (mgliyanage.rc@gmail.com)>> 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







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