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Madushan Geethanga

2016-Sep-09 16:23 UTC

[asterisk-users] Asterisk 13 PJSIP with Snom 710

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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Madushan Geethanga

2016-Sep-09 16:32 UTC

head link

[asterisk-users] 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 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>;tag=1bb809zgaa
To: <sip:0 at 54.206.59.252;user=phone>
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35
Contact: <sip:outburns00-nhvg5vjjn6-2001 at 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
--->
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>;tag=1bb809zgaa
To: <sip:0 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 --->
ACK sip:0 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>;tag=1bb809zgaa
To: <sip:0 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
Contact: <sip:outburns00-nhvg5vjjn6-2001 at 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>
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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Administrator TOOTAI

2016-Sep-09 16:57 UTC

head link

[asterisk-users] Asterisk 13 PJSIP with Snom 710

Le 09/09/2016 ? 18:32, Madushan Geethanga a ?crit :> 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 at
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 at
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
>
>
>
>

asterisk users - Sep 2016 - Asterisk 13 PJSIP with Snom 710

[asterisk-users] Asterisk 13 PJSIP with Snom 710

[asterisk-users] Asterisk 13 PJSIP with Snom 710

[asterisk-users] Asterisk 13 PJSIP with Snom 710