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 <http://123.231.72.210:33878> ---> INVITE sip:[email protected] <mailto: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] <mailto:sip%[email protected]>>;tag=1bb809zgaa To: <sip:[email protected] <mailto:sip%[email protected]>;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:[email protected]:45835 <http://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 <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:[email protected] <mailto:sip%[email protected]>>;tag=1bb809zgaa To: <sip:[email protected] <mailto: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 <http://123.231.72.210:33878> ---> ACK sip:[email protected] <mailto: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] <mailto:sip%[email protected]>>;tag=1bb809zgaa To: <sip:[email protected] <mailto: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 <http://8.7.5.35> Contact: <sip:[email protected]:45835 <http://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] <mailto:[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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