[asterisk-users] Asterisk 13 PJSIP with Snom 710
Madushan Geethanga
mgliyanage.rc at gmail.com
Fri Sep 9 12:01:24 CDT 2016
yes I have unchecked it.
On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI <admin at tootai.net>
wrote:
> 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;bra
>> nch=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/ef923d25464dbedc1dbd85e0c
>> cea08b7",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
>>
>>
>>
>>
>>
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