Asterisk 13 PJSIP With Snom 710

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Asterisk Users 4 Comments

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 01
aors 01
direct_media=no
rtp_symmetric=yes
force_rport=yes
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm

[2001]
type=aor
qualify_frequencyP00
authenticate_qualify=yes
max_contacts=1
remove_existing=yes

[2001]
type=auth
auth_type=userpass
password=test
username=test

Best Regards, Madushan

4 thoughts on - Asterisk 13 PJSIP With Snom 710

  • 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:0@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@54.206.59.252>;tagb809zgaa To:
    Call-ID: 313437333433383639323238313539-ahn3begiq66q CSeq: 1 INVITE
    Max-Forwards: 70
    User-Agent: snom710/8.7.5.35
    Contact: ;reg-id=1
    X-Serialnumber: 000413747C96
    P-Key-Flags: resolution=”31×13″, 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

  • 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