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W52p Not accepting <sip:anonymous@********.co.za
Author Message
Derick_Syrex Offline
Junior Member
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Posts: 2
Joined: Oct 2016
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Post: #1
W52p Not accepting <sip:anonymous@********.co.za
Hi,

This is my 1st time posting on here, so excuse me if I don't make sense.

I have a client with a PRI card that connects to Asterisk, the line is managed by Telkom and incoming calls coming over this line comes in as Unknown.

When one of the extensions on the base station is called directly from outside, the base station rejects the call.

When I set up a temp Inbound route bypassing the Telkom PRI it is presented with a number and it works fine.

All software is up to date, the "do not accept anonymous calls" is disabled.

I did a Sip Debug :

Anonymous call:

-- Executing [s@macro-dial-one:43] Dial("DAHDI/i1/-5c0", "SIP/6014,15,trL(10800000:300000)") in new stack
> Limit Data for this call:
> timelimit = 10800000 ms (10800.000 s)
> play_warning = 300000 ms (300.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 0 ms (0.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 17544
Video is at 192.168.1.4:13544
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100011 (g726) to SDP
Adding video codec 200001 (h261) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding video codec 200004 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.135:5063:
INVITE sip:6014@192.168.1.135:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7f79f56c
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@bandagsa.co.za>;tag=as09a3c1e9
To: <sip:6014@192.168.1.135:5063>
Contact: <sip:anonymous@192.168.1.4:5060>
Call-ID: ****************************************@bandagsa.co.za
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.12.1)
Date: Wed, 26 Oct 2016 14:52:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 955

v=0
o=root 1808413219 1808413219 IN IP4 192.168.1.4
s=Asterisk PBX 11.12.1
c=IN IP4 192.168.1.4
b=CT:384
t=0 0
m=audio 17544 RTP/AVP 8 0 5 18 4 9 3 10 110 7 97 112 111 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 13544 RTP/AVP 31 34 98 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=sendrecv

---
-- Called SIP/6014

<--- SIP read from UDP:192.168.1.135:5063 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7f79f56c
From: "Anonymous" <sip:anonymous@bandagsa.co.za>;tag=as09a3c1e9
To: <sip:6014@192.168.1.135:5063>
Call-ID: *************************************@bandagsa.co.za
CSeq: 102 INVITE
User-Agent: Yealink SIP-W52P 25.73.0.40
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.135:5063 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7f79f56c

From: "Anonymous" <sip:anonymous@bandagsa.co.za>;tag=as09a3c1e9
To: <sip:6014@192.168.1.135:5063>;tag=219382457
Call-ID: *****************************************@bandagsa.co.za
CSeq: 102 INVITE
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-W52P 25.73.0.40
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
-- Got SIP response 486 "Busy Here" back from 192.168.1.135:5063
Transmitting (no NAT) to 192.168.1.135:5063:
ACK sip:6014@192.168.1.135:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7f79f56c
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@bandagsa.co.za>;tag=as09a3c1e9
To: <sip:6014@192.168.1.135:5063>;tag=219382457
Contact: <sip:anonymous@192.168.1.4:5060>
Call-ID: *********************************@bandagsa.co.za
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.12.1)
Content-Length: 0

Known call:
-- Executing [s@macro-dial-one:43] Dial("IAX2/Syrex-10117", "SIP/6014,15,trL(10800000:300000)") in new stack
> Limit Data for this call:
> timelimit = 10800000 ms (10800.000 s)
> play_warning = 300000 ms (300.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 0 ms (0.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 14586
Video is at 192.168.1.4:11526
Adding codec 100008 (g729) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100011 (g726) to SDP
Adding video codec 200001 (h261) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding video codec 200004 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.135:5063:
INVITE sip:6014@192.168.1.135:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7485ce12
Max-Forwards: 70
From: "Syrex" <sip:0861179739@bandagsa.co.za>;tag=as0de39992
To: <sip:6014@192.168.1.135:5063>
Contact: <sip:0861179739@192.168.1.4:5060>
Call-ID: ***********************************@bandagsa.co.za
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.12.1)
Date: Wed, 26 Oct 2016 14:55:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 953

v=0
o=root 999421842 999421842 IN IP4 192.168.1.4
s=Asterisk PBX 11.12.1
c=IN IP4 192.168.1.4
b=CT:384
t=0 0
m=audio 14586 RTP/AVP 18 8 0 5 4 9 3 10 110 7 97 112 111 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 11526 RTP/AVP 31 34 98 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=sendrecv

---
-- Called SIP/6014

<--- SIP read from UDP:192.168.1.135:5063 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7485ce12
From: "Syrex" <sip:0861179739@bandagsa.co.za>;tag=as0de39992
To: <sip:6014@192.168.1.135:5063>
Call-ID: ********************************@bandagsa.co.za
CSeq: 102 INVITE
User-Agent: Yealink SIP-W52P 25.73.0.40
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.135:5063 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7485ce12

From: "Syrex" <sip:0861179739@bandagsa.co.za>;tag=as0de39992
To: <sip:6014@192.168.1.135:5063>;tag=265546598
Call-ID: **************************@bandagsa.co.za
CSeq: 102 INVITE
Contact: <sip:6014@192.168.1.135:5063>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-W52P 25.73.0.40
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0
10-27-2016 01:31 PM
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W52p Not accepting <sip:anonymous@********.co.za - Derick_Syrex - 10-27-2016 01:31 PM

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