+- Yealink Forums (http://forum.yealink.com/forum)
+-- Forum: Dect Phone Series (/forumdisplay.php?fid=6)
+--- Forum: W52P (Wireless) (/forumdisplay.php?fid=23)
+--- Thread: W52p Not accepting <sip:anonymous@********.co.za (/showthread.php?tid=33094)
W52p Not accepting <sip:anonymous@********.co.za - Derick_Syrex - 10-27-2016 01:31 PM
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
RE: W52p Not accepting <sip:anonymous@********.co.za - Lucia_Yealink - 10-27-2016 02:33 PM
Hi Derick,
About your problem do you mean the phone will reject all of the incoming call from outside? What is the number show in the call log? But the incoming call from the inside can work normally?
If the answer is yes, please share us more information to do trouble shooting:
1. What is the number show in the LCD screen when there are an outside incoming? Is it show "Unknow" or show others? It would be better if you can provide a picture.
2. Please update the firmware to the latest version(http://download.support.yealink.com/download?path=upload%2Fattachment%2F2016-7-14%2F6%2F982e354e-5822-4088-9e3c-c8550c415dc9%2FBase%20for%20W52P%26W56P-25.80.0.15.zip) and do test again. If the problem still exist, please provide the level 6 syslog, config.bin and pace trace when you reproduce the issue to Lucia@yealink.com.
About how to get the three files, please refer to the FAQ below:
http://support.yealink.com/faq/faqInfo?id=313
3.Please also provide the three files of the normal incoming call(inside call) to make a comparison. Thank you.
4. How many phones you have in total? And how many of them have the issue?
Thank you for your information in advanced.
Best Regards,
Lucia Lu
RE: W52p Not accepting <sip:anonymous@********.co.za - Derick_Syrex - 10-27-2016 04:02 PM
(10-27-2016 02:33 PM)Yealink_Lucia Wrote: Hi Derick,
About your problem do you mean the phone will reject all of the incoming call from outside? What is the number show in the call log? But the incoming call from the inside can work normally?
If the answer is yes, please share us more information to do trouble shooting:
1. What is the number show in the LCD screen when there are an outside incoming? Is it show "Unknow" or show others? It would be better if you can provide a picture.
2. Please update the firmware to the latest version(http://download.support.yealink.com/download?path=upload%2Fattachment%2F2016-7-14%2F6%2F982e354e-5822-4088-9e3c-c8550c415dc9%2FBase%20for%20W52P%26W56P-25.80.0.15.zip) and do test again. If the problem still exist, please provide the level 6 syslog, config.bin and pace trace when you reproduce the issue to Lucia@yealink.com.
About how to get the three files, please refer to the FAQ below:
http://support.yealink.com/faq/faqInfo?id=313
3.Please also provide the three files of the normal incoming call(inside call) to make a comparison. Thank you.
4. How many phones you have in total? And how many of them have the issue?
Thank you for your information in advanced.
Best Regards,
Lucia Lu
Hi,
There is no number showing on the LCD screen, the call gets rejected by the base station before the handset rings.
I have updated the firmware and the problem is still there.
Normal incoming calls work as it presents the base station with the internal extension the call is coming from. It is only calls from outside the company it rejects.
This is only happening on the W52p with 4 phones connected to the base station, all the other brand of phones work fine when dialing from outside.
The other phones on the network are Snom and Cisco. They also get presented with a "anonymous" source number and the call goes through.
It is only on the Yealing base station where the "anonymous" source gets rejected. In the post I did supply both scenarios. One "anonymous" and one "known".
RE: W52p Not accepting <sip:anonymous@********.co.za - Lucia_Yealink - 10-28-2016 02:04 AM
Dear Derick,
Thank you for your detail information.
Please also provide the level 6 syslog, config.bin and pace trace when you reproduce the issue to Lucia@yealink.com, so that our R&D can do more trouble shooting with your issue.
About how to get the three files, please refer to the FAQ below:
http://support.yealink.com/faq/faqInfo?id=313
Please also provide the three files of the normal incoming call(inside call) to make a comparison. Thank you.
Best Regards,
Lucia