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Unable to receive Inbound calls

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Unable to receive Inbound calls

Postby shanepearce » Mon Nov 28, 2016 4:05 pm

Hello,

I am new to vicidial. I have been using it some time and i have attempted my first installation. i have managed to get the outbound calling to work perfectly. However, Inbound seems to be a problem and i cant wrap my head around the issue. Please see my settings below:

Linux vicibox7 4.1.31-30-default #1 SMP PREEMPT Wed Aug 24 06:20:09 UTC 2016 (de9ddf8) x86_64 x86_64 x86_64 GNU/Linux
VERSION: 2.12-565a
BUILD: 160827-0917
Asterisk 11.22.0-vici
Installed from ISO on vicibox website.
No customization
(do hope i stated everything there)

Register String:
register => 592231638xxx@gtt.co.gy:xxxxxxxx:592231638900@190.80.12.215:5060/592231638xxx

Account Entry:
[guyana]
disallow=all
nat=yes
canreinvite=no
context=trunkinbound
from=592231638xxx
fromdomain=gtt.co.gy
host=gtt.co.gy
insecure=invite,port
qualify=yes
type-friend
outboundproxy=190.80.12.215
externip=190.80.12.215
port=5060
allowguest=yes
secret=xxxxxxxx
username=592231638xxx
authname=592231638xxx
dtmfmode=rfc2833
allow=ulaw
allow=alaw
registername=592231638xxx


exten => _7592.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _7592.,2,Dial(SIP/${EXTEN:1}@guyana,,tTo)
exten => _7592.,3,Hangup

Please see error from CLI below:

Code: Select all
sip set debug on
SIP Debugging enabled
[Nov 28 17:03:19] Reliably Transmitting (NAT) to 192.168.1.104:60872:
[Nov 28 17:03:19] OPTIONS sip:2000@192.168.1.104:60872;rinstance=5535a6450363bcb6 SIP/2.0
[Nov 28 17:03:19] Via: SIP/2.0/UDP 192.168.1.117:5060;branch=z9hG4bK2dc233f6;rport
[Nov 28 17:03:19] Max-Forwards: 70
[Nov 28 17:03:19] From: "asterisk" <sip:asterisk@192.168.1.117>;tag=as460f219c
[Nov 28 17:03:19] To: <sip:2000@192.168.1.104:60872;rinstance=5535a6450363bcb6>
[Nov 28 17:03:19] Contact: <sip:asterisk@192.168.1.117:5060>
[Nov 28 17:03:19] Call-ID: 0a31fb5e2f9ccf5715b919ae1b318a56@192.168.1.117:5060
[Nov 28 17:03:19] CSeq: 102 OPTIONS
[Nov 28 17:03:19] User-Agent: Asterisk PBX 11.22.0-vici
[Nov 28 17:03:19] Date: Mon, 28 Nov 2016 21:03:19 GMT
[Nov 28 17:03:19] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 28 17:03:19] Supported: replaces, timer
[Nov 28 17:03:19] Content-Length: 0
[Nov 28 17:03:19]
[Nov 28 17:03:19]
[Nov 28 17:03:19] ---
[Nov 28 17:03:19]
[Nov 28 17:03:19] <--- SIP read from UDP:192.168.1.104:60872 --->
[Nov 28 17:03:19] SIP/2.0 200 OK
[Nov 28 17:03:19] Via: SIP/2.0/UDP 192.168.1.117:5060;branch=z9hG4bK2dc233f6;rport=5060
[Nov 28 17:03:19] Contact: <sip:192.168.1.104:60872>
[Nov 28 17:03:19] To: <sip:2000@192.168.1.104:60872;rinstance=5535a6450363bcb6>;tag=0d624133
[Nov 28 17:03:19] From: "asterisk"<sip:asterisk@192.168.1.117>;tag=as460f219c
[Nov 28 17:03:19] Call-ID: 0a31fb5e2f9ccf5715b919ae1b318a56@192.168.1.117:5060
[Nov 28 17:03:19] CSeq: 102 OPTIONS
[Nov 28 17:03:19] Accept: application/sdp
[Nov 28 17:03:19] Accept-Language: en
[Nov 28 17:03:19] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
[Nov 28 17:03:19] Supported: replaces
[Nov 28 17:03:19] Allow-Events: presence, message-summary, tunnel-info
[Nov 28 17:03:19] Content-Length: 0
[Nov 28 17:03:19]
[Nov 28 17:03:19] <------------->
[Nov 28 17:03:19] --- (13 headers 0 lines) ---
[Nov 28 17:03:19] Really destroying SIP dialog '0a31fb5e2f9ccf5715b919ae1b318a56@192.168.1.117:5060' Method: OPTIONS
[Nov 28 17:03:26]
[Nov 28 17:03:26] <--- SIP read from UDP:190.80.12.215:5060 --->
[Nov 28 17:03:26] INVITE sip:592231638900@190.108.212.250:5060 SIP/2.0
[Nov 28 17:03:26] Max-Forwards: 19
[Nov 28 17:03:26] Session-Expires: 3600;refresher=uas
[Nov 28 17:03:26] Min-SE: 600
[Nov 28 17:03:26] Supported: timer, 100rel,replaces,gin,x-nortel-sipvc,nosec,p-3rdpartycontrol,com.nortelnetworks.firewall
[Nov 28 17:03:26] To: "Sambora Comm" <sip:592231638900@190.108.212.250:5060>
[Nov 28 17:03:26] From: <sip:anonymous@anonymous.invalid>;tag=3689355806-188326
[Nov 28 17:03:26] Call-ID: 1474460-3689355806-188320@GYGTS30.cellinkgy.com
[Nov 28 17:03:26] CSeq: 1 INVITE
[Nov 28 17:03:26] Allow: UPDATE,REFER
[Nov 28 17:03:26] Via: SIP/2.0/UDP 190.80.12.215:5060;branch=z9hG4bKbf450d116ec1db7499855da4efc9856d
[Nov 28 17:03:26] Contact: <sip:anonymous@190.80.12.215:5060>
[Nov 28 17:03:26] Content-Type: application/sdp
[Nov 28 17:03:26] Accept: application/sdp
[Nov 28 17:03:26] User-Agent: Nortel SESM 18.0.26.1
[Nov 28 17:03:26] X-Nt-Corr-Id: 7130da26-0b1c-e017-91f9-000000000000
[Nov 28 17:03:26] Content-Length: 299
[Nov 28 17:03:26]
[Nov 28 17:03:26] v=0
[Nov 28 17:03:26] o=GYGTS30 303899648 1480367007 IN IP4 190.80.12.215
[Nov 28 17:03:26] s=sip call
[Nov 28 17:03:26] c=IN IP4 190.80.12.217
[Nov 28 17:03:26] t=0 0
[Nov 28 17:03:26] m=audio 64998 RTP/AVP 8 18 101
[Nov 28 17:03:26] a=rtpmap:8 PCMA/8000
[Nov 28 17:03:26] a=rtpmap:18 G729/8000
[Nov 28 17:03:26] a=fmtp:18 annexb=no
[Nov 28 17:03:26] a=rtpmap:101 telephone-event/8000
[Nov 28 17:03:26] a=fmtp:101 0-15
[Nov 28 17:03:26] a=ptime:20
[Nov 28 17:03:26] a=sqn: 0
[Nov 28 17:03:26] a=cdsc:1 image udptl t38
[Nov 28 17:03:26] <------------->
[Nov 28 17:03:26] --- (17 headers 14 lines) ---
[Nov 28 17:03:26] Sending to 190.80.12.215:5060 (NAT)
[Nov 28 17:03:26] Sending to 190.80.12.215:5060 (NAT)
[Nov 28 17:03:26] Using INVITE request as basis request - 1474460-3689355806-188320@GYGTS30.cellinkgy.com
[Nov 28 17:03:26] No matching peer for 'anonymous' from '190.80.12.215:5060'
[Nov 28 17:03:26]
[Nov 28 17:03:26] <--- Reliably Transmitting (NAT) to 190.80.12.215:5060 --->
[Nov 28 17:03:26] SIP/2.0 401 Unauthorized
[Nov 28 17:03:26] Via: SIP/2.0/UDP 190.80.12.215:5060;branch=z9hG4bKbf450d116ec1db7499855da4efc9856d;received=190.80.12.215;rport=5060
[Nov 28 17:03:26] From: <sip:anonymous@anonymous.invalid>;tag=3689355806-188326
[Nov 28 17:03:26] To: "Sambora Comm" <sip:592231638900@190.108.212.250:5060>;tag=as35d1c510
[Nov 28 17:03:26] Call-ID: 1474460-3689355806-188320@GYGTS30.cellinkgy.com
[Nov 28 17:03:26] CSeq: 1 INVITE
[Nov 28 17:03:26] Server: Asterisk PBX 11.22.0-vici
[Nov 28 17:03:26] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 28 17:03:26] Supported: replaces, timer
[Nov 28 17:03:26] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5cbdf557"
[Nov 28 17:03:26] Content-Length: 0
[Nov 28 17:03:26]
[Nov 28 17:03:26]
[Nov 28 17:03:26] <------------>
[Nov 28 17:03:26] Scheduling destruction of SIP dialog '1474460-3689355806-188320@GYGTS30.cellinkgy.com' in 32000 ms (Method: INVITE)
[Nov 28 17:03:26]
[Nov 28 17:03:26] <--- SIP read from UDP:190.80.12.215:5060 --->
[Nov 28 17:03:26] ACK sip:592231638900@190.108.212.250:5060 SIP/2.0
[Nov 28 17:03:26] Max-Forwards: 70
[Nov 28 17:03:26] To: "Sambora Comm" <sip:592231638900@190.108.212.250:5060>;tag=as35d1c510
[Nov 28 17:03:26] From: <sip:anonymous@anonymous.invalid>;tag=3689355806-188326
[Nov 28 17:03:26] Call-ID: 1474460-3689355806-188320@GYGTS30.cellinkgy.com
[Nov 28 17:03:26] CSeq: 1 ACK
[Nov 28 17:03:26] Via: SIP/2.0/UDP 190.80.12.215:5060;branch=z9hG4bKbf450d116ec1db7499855da4efc9856d
[Nov 28 17:03:26] Contact: <sip:anonymous@190.80.12.215:5060>
[Nov 28 17:03:26] Content-Length: 0
[Nov 28 17:03:26]
[Nov 28 17:03:26] <------------->
[Nov 28 17:03:26] --- (9 headers 0 lines) ---
[Nov 28 17:03:28]
[Nov 28 17:03:28] <--- SIP read from UDP:192.168.1.104:60872 --->
[Nov 28 17:03:28]
[Nov 28 17:03:28]
[Nov 28 17:03:28] <------------->
[Nov 28 17:03:35] Really destroying SIP dialog '1474457-3689355783-519997@GYGTS30.cellinkgy.com' Method: ACK
vicibox7*CLI>




Grateful if someone can point me in the right direction.

thank you
shanepearce
 
Posts: 23
Joined: Mon Nov 28, 2016 3:50 pm
Top

Re: Unable to receive Inbound calls

Postby ambiorixg12 » Tue Nov 29, 2016 9:26 pm

Your sip.conf file is misconfigured, allowguest=yes must be on the [general] section, if you are going to allow guest calls make sure also to send the calls to the context=trunkinbound , asterisk is challenging authentication (401), the allowguest should solve the issue, but that the best fix

[general]
context=trunkinbound ; Default context for incoming calls. Defaults to 'default'
allowguest=yes
ambiorixg12
 
Posts: 453
Joined: Tue Sep 17, 2013 10:35 pm
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Re: Unable to receive Inbound calls

Postby shanepearce » Wed Nov 30, 2016 9:50 am

Thank you for your response @ambiorixg12

I added the [general] reference to my Account Details and reloaded by core and sip. However i still get the same errors when i try to do the inbound call. Please see details below:

Account Entry:
[guyana]
disallow=all
nat=yes
canreinvite=no
from=592231638XXX
fromdomain=gtt.co.gy
host=gtt.co.gy
insecure=invite,port
qualify=yes
type=friend
outboundproxy=190.80.12.215
externip=190.80.12.215
port=5060
secret=XXXXXXX
username=592231638XXX
authname=592231638XXX
dtmfmode=rfc2833
allow=ulaw
allow=alaw
registername=592231638XXX

[general]
context=trunkinbound
allowguest=yes

Code: Select all
vicibox7:~ # asterisk -r
[Nov 30 10:49:05] Asterisk 11.22.0-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
[Nov 30 10:49:05] Created by Mark Spencer <markster@digium.com>
[Nov 30 10:49:05] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
[Nov 30 10:49:05] This is free software, with components licensed under the GNU General Public
[Nov 30 10:49:05] License version 2 and other licenses; you are welcome to redistribute it under
[Nov 30 10:49:05] certain conditions. Type 'core show license' for details.
[Nov 30 10:49:05] =========================================================================
[Nov 30 10:49:05] Connected to Asterisk 11.22.0-vici currently running on vicibox7 (pid = 2418)
[Nov 30 10:49:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov 30 10:49:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov 30 10:49:12]
[Nov 30 10:49:12] <--- SIP read from UDP:190.80.12.215:5060 --->
[Nov 30 10:49:12] INVITE sip:592231638900@190.108.212.250:5060 SIP/2.0
[Nov 30 10:49:12] Max-Forwards: 19
[Nov 30 10:49:12] Session-Expires: 3600;refresher=uas
[Nov 30 10:49:12] Min-SE: 600
[Nov 30 10:49:12] Supported: timer, 100rel,replaces,gin,x-nortel-sipvc,nosec,p-3rdpartycontrol,com.nortelnetworks.firewall
[Nov 30 10:49:12] To: "Sambora Comm" <sip:592231638900@190.108.212.250:5060>
[Nov 30 10:49:12] From: <sip:anonymous@anonymous.invalid>;tag=3689506152-539469
[Nov 30 10:49:12] Call-ID: 1507850-3689506152-539464@GYGTS30.cellinkgy.com
[Nov 30 10:49:12] CSeq: 1 INVITE
[Nov 30 10:49:12] Allow: UPDATE,REFER
[Nov 30 10:49:12] Via: SIP/2.0/UDP 190.80.12.215:5060;branch=z9hG4bK7f0b6cacbb5683a741e1ee1a079bfc7e
[Nov 30 10:49:12] Contact: <sip:anonymous@190.80.12.215:5060>
[Nov 30 10:49:12] Content-Type: application/sdp
[Nov 30 10:49:12] Accept: application/sdp
[Nov 30 10:49:12] User-Agent: Nortel SESM 18.0.26.1
[Nov 30 10:49:12] X-Nt-Corr-Id: 2a360c7e-0b1e-e017-cdfa-000000000000
[Nov 30 10:49:12] Content-Length: 299
[Nov 30 10:49:12]
[Nov 30 10:49:12] v=0
[Nov 30 10:49:12] o=GYGTS30 308749312 1480517353 IN IP4 190.80.12.215
[Nov 30 10:49:12] s=sip call
[Nov 30 10:49:12] c=IN IP4 190.80.12.217
[Nov 30 10:49:12] t=0 0
[Nov 30 10:49:12] m=audio 31402 RTP/AVP 8 18 101
[Nov 30 10:49:12] a=rtpmap:8 PCMA/8000
[Nov 30 10:49:12] a=rtpmap:18 G729/8000
[Nov 30 10:49:12] a=fmtp:18 annexb=no
[Nov 30 10:49:12] a=rtpmap:101 telephone-event/8000
[Nov 30 10:49:12] a=fmtp:101 0-15
[Nov 30 10:49:12] a=ptime:20
[Nov 30 10:49:12] a=sqn: 0
[Nov 30 10:49:12] a=cdsc:1 image udptl t38
[Nov 30 10:49:12] <------------->
[Nov 30 10:49:12] --- (17 headers 14 lines) ---
[Nov 30 10:49:12] Sending to 190.80.12.215:5060 (NAT)
[Nov 30 10:49:12] Sending to 190.80.12.215:5060 (NAT)
[Nov 30 10:49:12] Using INVITE request as basis request - 1507850-3689506152-539464@GYGTS30.cellinkgy.com
[Nov 30 10:49:12] No matching peer for 'anonymous' from '190.80.12.215:5060'
[Nov 30 10:49:12]
[Nov 30 10:49:12] <--- Reliably Transmitting (NAT) to 190.80.12.215:5060 --->
[Nov 30 10:49:12] SIP/2.0 401 Unauthorized
[Nov 30 10:49:12] Via: SIP/2.0/UDP 190.80.12.215:5060;branch=z9hG4bK7f0b6cacbb5683a741e1ee1a079bfc7e;received=190.80.12.215;rport=5060
[Nov 30 10:49:12] From: <sip:anonymous@anonymous.invalid>;tag=3689506152-539469
[Nov 30 10:49:12] To: "Sambora Comm" <sip:592231638900@190.108.212.250:5060>;tag=as50dbd0f9
[Nov 30 10:49:12] Call-ID: 1507850-3689506152-539464@GYGTS30.cellinkgy.com
[Nov 30 10:49:12] CSeq: 1 INVITE
[Nov 30 10:49:12] Server: Asterisk PBX 11.22.0-vici
[Nov 30 10:49:12] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 30 10:49:12] Supported: replaces, timer
[Nov 30 10:49:12] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0640d51e"
[Nov 30 10:49:12] Content-Length: 0
[Nov 30 10:49:12]
[Nov 30 10:49:12]
[Nov 30 10:49:12] <------------>
[Nov 30 10:49:12] Scheduling destruction of SIP dialog '1507850-3689506152-539464@GYGTS30.cellinkgy.com' in 32000 ms (Method: INVITE)
[Nov 30 10:49:12]
[Nov 30 10:49:12] <--- SIP read from UDP:190.80.12.215:5060 --->
[Nov 30 10:49:12] ACK sip:592231638900@190.108.212.250:5060 SIP/2.0
[Nov 30 10:49:12] Max-Forwards: 70
[Nov 30 10:49:12] To: "Sambora Comm" <sip:592231638900@190.108.212.250:5060>;tag=as50dbd0f9
[Nov 30 10:49:12] From: <sip:anonymous@anonymous.invalid>;tag=3689506152-539469
[Nov 30 10:49:12] Call-ID: 1507850-3689506152-539464@GYGTS30.cellinkgy.com
[Nov 30 10:49:12] CSeq: 1 ACK
[Nov 30 10:49:12] Via: SIP/2.0/UDP 190.80.12.215:5060;branch=z9hG4bK7f0b6cacbb5683a741e1ee1a079bfc7e
[Nov 30 10:49:12] Contact: <sip:anonymous@190.80.12.215:5060>
[Nov 30 10:49:12] Content-Length: 0
[Nov 30 10:49:12]
[Nov 30 10:49:12] <------------->
[Nov 30 10:49:12] --- (9 headers 0 lines) ---
[Nov 30 10:49:18]
[Nov 30 10:49:18] <--- SIP read from UDP:192.168.1.104:54922 --->
[Nov 30 10:49:18]
[Nov 30 10:49:18]
[Nov 30 10:49:18] <--


Is there anything else that i can try? This is happening for all calls.

thank you.
shanepearce
 
Posts: 23
Joined: Mon Nov 28, 2016 3:50 pm
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Re: Unable to receive Inbound calls

Postby ambiorixg12 » Wed Nov 30, 2016 12:19 pm

The issue is pretty clear

[Nov 30 10:49:12] No matching peer for 'anonymous' from '190.80.12.215:5060'
[Nov 30 10:49:12]


This is the reason why Asterisk is Asking for Authentication

[Nov 30 10:49:12] SIP/2.0 401 Unauthorized


Dont know if this matter but general section should be on the top of the sip.conf file
ambiorixg12
 
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Re: Unable to receive Inbound calls

Postby shanepearce » Wed Nov 30, 2016 2:17 pm

Okay let me try that and see.
shanepearce
 
Posts: 23
Joined: Mon Nov 28, 2016 3:50 pm
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Re: Unable to receive Inbound calls

Postby shanepearce » Wed Nov 30, 2016 2:19 pm

Still have the same issue.
shanepearce
 
Posts: 23
Joined: Mon Nov 28, 2016 3:50 pm
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Re: Unable to receive Inbound calls

Postby ambiorixg12 » Wed Nov 30, 2016 2:39 pm

On linux shell run this command asterisk -x " sip show settings" | less and then verify if you have Allow unknown access set to yes




Allow unknown access: Yes
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Re: Unable to receive Inbound calls

Postby shanepearce » Wed Nov 30, 2016 3:23 pm

It is not.

Allow unknown access: No
shanepearce
 
Posts: 23
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Re: Unable to receive Inbound calls

Postby ambiorixg12 » Wed Nov 30, 2016 3:35 pm

This is the reason why calls are not working
ambiorixg12
 
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Re: Unable to receive Inbound calls

Postby shanepearce » Wed Nov 30, 2016 3:43 pm

How do i turn that on please?
shanepearce
 
Posts: 23
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Re: Unable to receive Inbound calls

Postby ambiorixg12 » Wed Nov 30, 2016 3:57 pm

I just told you before. On the general section. And then run this command on the asterisk command line core restart now and then verify if setting is enabled

[general]
context=trunkinbound ; Default context for incoming calls
allowguest=yes
Last edited by ambiorixg12 on Wed Nov 30, 2016 3:58 pm, edited 1 time in total.
ambiorixg12
 
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Re: Unable to receive Inbound calls

Postby shanepearce » Wed Nov 30, 2016 3:58 pm

thank you.

Let me try that again
shanepearce
 
Posts: 23
Joined: Mon Nov 28, 2016 3:50 pm
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Re: Unable to receive Inbound calls

Postby shanepearce » Thu Dec 01, 2016 11:58 am

thanks again for the help. That did the trick.

However, as luck (or lack of it) would have it, i have another road block.

The call is being accepted now. But there is an issue with codec. My provider stated that they support, g729a, g711a and T38. I do not have g729 on my server. So the call is coming in and is not processed because it comes in g729 and i dont support this. Is there a way to have it converted to maybe g711a or something?

Please see results from CLI below. I am not sure what to do here:

Code: Select all
[Dec  1 12:46:15]
[Dec  1 12:46:15] <--- SIP read from UDP:190.80.12.215:5060 --->
[Dec  1 12:46:15] INVITE sip:592231638900@190.108.212.250:5060 SIP/2.0
[Dec  1 12:46:15] Max-Forwards: 19
[Dec  1 12:46:15] Session-Expires: 3600;refresher=uas
[Dec  1 12:46:15] Min-SE: 600
[Dec  1 12:46:15] Supported: timer, 100rel,replaces,gin,x-nortel-sipvc,nosec,p-3rdpartycontrol,com.nortelnetworks.firewall
[Dec  1 12:46:15] To: "Sambora Comm" <sip:592231638900@190.108.212.250:5060>
[Dec  1 12:46:15] From: <sip:anonymous@anonymous.invalid>;tag=3689599575-658531
[Dec  1 12:46:15] Call-ID: 1525729-3689599575-658526@GYGTS30.cellinkgy.com
[Dec  1 12:46:15] CSeq: 1 INVITE
[Dec  1 12:46:15] Allow: UPDATE,REFER
[Dec  1 12:46:15] Via: SIP/2.0/UDP 190.80.12.215:5060;branch=z9hG4bK9a84c3e6f217e1c07944be0da2d822ce
[Dec  1 12:46:15] Contact: <sip:anonymous@190.80.12.215:5060>
[Dec  1 12:46:15] Content-Type: application/sdp
[Dec  1 12:46:15] Accept: application/sdp
[Dec  1 12:46:15] User-Agent: Nortel SESM 18.0.26.1
[Dec  1 12:46:15] X-Nt-Corr-Id: cc350f9a-0c01-e017-f9ae-000000000000
[Dec  1 12:46:15] Content-Length: 299
[Dec  1 12:46:15]
[Dec  1 12:46:15] v=0
[Dec  1 12:46:15] o=GYGTS30 320929792 1480610776 IN IP4 190.80.12.215
[Dec  1 12:46:15] s=sip call
[Dec  1 12:46:15] c=IN IP4 190.80.12.217
[Dec  1 12:46:15] t=0 0
[Dec  1 12:46:15] m=audio 14492 RTP/AVP 8 18 101
[Dec  1 12:46:15] a=rtpmap:8 PCMA/8000
[Dec  1 12:46:15] a=rtpmap:18 G729/8000
[Dec  1 12:46:15] a=fmtp:18 annexb=no
[Dec  1 12:46:15] a=rtpmap:101 telephone-event/8000
[Dec  1 12:46:15] a=fmtp:101 0-15
[Dec  1 12:46:15] a=ptime:20
[Dec  1 12:46:15] a=sqn: 0
[Dec  1 12:46:15] a=cdsc:1 image udptl t38
[Dec  1 12:46:15] <------------->
[Dec  1 12:46:15] --- (17 headers 14 lines) ---
[Dec  1 12:46:15] Sending to 190.80.12.215:5060 (NAT)
[Dec  1 12:46:15] Sending to 190.80.12.215:5060 (NAT)
[Dec  1 12:46:15] Using INVITE request as basis request - 1525729-3689599575-658526@GYGTS30.cellinkgy.com
[Dec  1 12:46:15] No matching peer for 'anonymous' from '190.80.12.215:5060'
[Dec  1 12:46:15]   == Using SIP RTP CoS mark 5
[Dec  1 12:46:15] Found RTP audio format 8
[Dec  1 12:46:15] Found RTP audio format 18
[Dec  1 12:46:15] Found RTP audio format 101
[Dec  1 12:46:15] Found audio description format PCMA for ID 8
[Dec  1 12:46:15] Found audio description format G729 for ID 18
[Dec  1 12:46:15] Found audio description format telephone-event for ID 101
[Dec  1 12:46:15] NOTICE[22421][C-00000000]: chan_sip.c:10541 process_sdp: No compatible codecs, not accepting this offer!
[Dec  1 12:46:15]
[Dec  1 12:46:15] <--- Reliably Transmitting (NAT) to 190.80.12.215:5060 --->
[Dec  1 12:46:15] SIP/2.0 488 Not acceptable here
[Dec  1 12:46:15] Via: SIP/2.0/UDP 190.80.12.215:5060;branch=z9hG4bK9a84c3e6f217e1c07944be0da2d822ce;received=190.80.12.215;rport=5060
[Dec  1 12:46:15] From: <sip:anonymous@anonymous.invalid>;tag=3689599575-658531
[Dec  1 12:46:15] To: "Sambora Comm" <sip:592231638900@190.108.212.250:5060>;tag=as3d6a6e88
[Dec  1 12:46:15] Call-ID: 1525729-3689599575-658526@GYGTS30.cellinkgy.com
[Dec  1 12:46:15] CSeq: 1 INVITE
[Dec  1 12:46:15] Server: Asterisk PBX 11.22.0-vici
[Dec  1 12:46:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Dec  1 12:46:15] Supported: replaces, timer
[Dec  1 12:46:15] Content-Length: 0
[Dec  1 12:46:15]
[Dec  1 12:46:15]
[Dec  1 12:46:15] <------------>
[Dec  1 12:46:15] Scheduling destruction of SIP dialog '1525729-3689599575-658526@GYGTS30.cellinkgy.com' in 32000 ms (Method: INVITE)
[Dec  1 12:46:15]
[Dec  1 12:46:15] <--- SIP read from UDP:190.80.12.215:5060 --->
[Dec  1 12:46:15] ACK sip:592231638900@190.108.212.250:5060 SIP/2.0
[Dec  1 12:46:15] Max-Forwards: 70
[Dec  1 12:46:15] To: "Sambora Comm" <sip:592231638900@190.108.212.250:5060>;tag=as3d6a6e88
[Dec  1 12:46:15] From: <sip:anonymous@anonymous.invalid>;tag=3689599575-658531
[Dec  1 12:46:15] Call-ID: 1525729-3689599575-658526@GYGTS30.cellinkgy.com
[Dec  1 12:46:15] CSeq: 1 ACK
[Dec  1 12:46:15] Via: SIP/2.0/UDP 190.80.12.215:5060;branch=z9hG4bK9a84c3e6f217e1c07944be0da2d822ce
[Dec  1 12:46:15] Contact: <sip:anonymous@190.80.12.215:5060>
[Dec  1 12:46:15] Content-Length: 0
shanepearce
 
Posts: 23
Joined: Mon Nov 28, 2016 3:50 pm
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Re: Unable to receive Inbound calls

Postby shanepearce » Thu Dec 01, 2016 12:10 pm

UPDATE:

i managed to get it fixed.

I added allow=alaw to muy sip.conf file and it worked.

I had it defined in my carrier settings via the GUI but it turns out that i need to out it in [general]

thank you for all the help.
shanepearce
 
Posts: 23
Joined: Mon Nov 28, 2016 3:50 pm
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