[asterisk-users] Asterisk 13 PJSIP with Snom 710
Madushan Geethanga
mgliyanage.rc at gmail.com
Fri Sep 9 12:03:18 CDT 2016
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 at gmail.com
> wrote:
> 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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