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:[email protected];user=phone SIP/2.0 Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport From: "outburns00-nhvg5vjjn6-2001" < sip:[email protected]>;tag=1bb809zgaa To: <sip:[email protected];user=phone> Call-ID: 313437333433383639323238313539-ahn3begiq66q CSeq: 1 INVITE Max-Forwards: 70 User-Agent: snom710/8.7.5.35 Contact: <sip:[email protected]: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:[email protected]>;tag=1bb809zgaa To: <sip:[email protected];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:[email protected];user=phone SIP/2.0 Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport From: "outburns00-nhvg5vjjn6-2001" < sip:[email protected]>;tag=1bb809zgaa To: <sip:[email protected];user=phone>;tag=z9hG4bK-bskkkx1t5bas Call-ID: 313437333433383639323238313539-ahn3begiq66q CSeq: 1 ACK Max-Forwards: 70 User-Agent: snom710/8.7.5.35 Contact: <sip:[email protected]:45835>;reg-id=1 Content-Length: 0 Best Regards, Madushan On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga <[email protected]> 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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