[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 at 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 at 54.206.59.252>;tag=1bb809zgaa
To: <sip:0 at 54.206.59.252;user=phone>
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35
Contact: <sip:outburns00-nhvg5vjjn6-2001 at 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 at 54.206.59.252>;tag=1bb809zgaa
To: <sip:0 at 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 --->
ACK sip:0 at 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 at 54.206.59.252>;tag=1bb809zgaa
To: <sip:0 at 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
Contact: <sip:outburns00-nhvg5vjjn6-2001 at 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 at 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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