[asterisk-users] Asterisk 13 PJSIP with Snom 710

[Administrator TOOTAI]

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:0 at 54.206.59.252 <mailto:sip%3A0 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
> <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa
> To: <sip:0 at 54.206.59.252 <mailto:sip%3A0 at 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 at 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 at 54.206.59.252
> <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa
> To: <sip:0 at 54.206.59.252
> <mailto:sip%3A0 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
> <http://123.231.72.210:33878> --->
> ACK sip:0 at 54.206.59.252 <mailto:sip%3A0 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
> <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa
> To: <sip:0 at 54.206.59.252
> <mailto:sip%3A0 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 <http://8.7.5.35>
> Contact: <sip:outburns00-nhvg5vjjn6-2001 at 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 at gmail.com <mailto: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
>
>
>
>