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[mgliyanage.rc at gmail...]

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

[mgliyanage.rc at gmail...]

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:

[admin at tootai.net]

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

[mgliyanage.rc at gmail...]

yes I have unchecked it.


On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI <admin@tootai.net (admin@tootai.net)> wrote:

[mgliyanage.rc at gmail...]

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: