Inbound call [SOLVED] and outbound [NOT SOLVED] doesn't work

General Help
1

Hello!
I’m new of FreePBX and I setup a VM with CentOS and Asterisk with FreePBX.
All seems to go well, I read a lot of documentation to configure in a right way all, but now I’m stopped in a strange problem:

My trunk is registered correctly:

localhost*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time                 
voip.xxx.it:5060                    N      xxx        105 Registered           Wed, 06 Aug 2014 20:19:47
1 SIP registrations.

My peers are registered correctly:

localhost*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      
1                         (Unspecified)                            D  No         No          A  0        UNKNOWN                                      
2/2                       192.168.7.20                             D  No         No          A  5060     OK (18 ms)                                   
3/3                       192.168.7.22                             D  Yes        Yes         A  5060     OK (77 ms)                                   
xx/xx xxx.xxx.xxx.xxx                               No         No             5060     OK (26 ms)                                   
4 sip peers [Monitored: 3 online, 1 offline Unmonitored: 0 online, 0 offline]

When I receive an Incoming call all my peers ring together as a group in a right way.
But when I try to answer the call it is with no sound and terminate after 8 seconds.
Extensions can call each other with no problem inside my network.

More when I try to generate an outgoing call suddenly I receive the message: “All circuits are busy”.

So I tried to create a DMZ to avoid no sound in coming calls but nothing happens, calls are still with no sound.

The only thing I’ve noticed strange is this one in debug mode:

[2014-08-06 20:29:01] WARNING[1576][C-00000005]: chan_sip.c:23167 handle_response_invite: Received response: "Forbidden" from '"Andres" <sip:[email protected]>;tag=as270445a2'

And I don’t understand why. Have you any suggestion to help a newbie like me? :blush:
Thanks in advance

Andrea

2

One small add, now I’ve fixed the problem of inbound calls.
There where g729 and g723 codec that doesn’t work properly so I disabled them and now incoming calls works good.

There is still the problem of outbound calls to solve…tok tok…I’m breaking my head…please help me!

3

  1. On asterisk CLI “sip set debug on”

  2. Make one call

  3. Goto freepbx module “Reports” -> “Asterisk Logfiles”

  4. Set value from 500 to 5000

  5. Copy and post output like this

    ÿ<— SIP read from UDP:192.168.1.241:5060 —>
    ÿSUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
    ÿVia: SIP/2.0/UDP 192.168.1.241:5060;branch=z9hG4bK-d8754z-b4bdd093f40fbb43-1—d8754z-
    ÿMax-Forwards: 70
    ÿContact: sip:[email protected]:5060;transport=UDP
    ÿTo: sip:[email protected];transport=UDP
    ÿFrom: sip:[email protected];transport=UDP;tag=6d043b5e
    ÿCall-ID: MjM4ZTdmMDMyNjY1YmU5ZjFhNDY1NThkNDAyYmJlM2I.
    ÿCSeq: 2 SUBSCRIBE
    ÿExpires: 600
    ÿAccept: application/watcherinfo+xml
    ÿAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
    ÿSupported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
    ÿUser-Agent: Z 3.2.21357 r21367
    ÿAuthorization: Digest username=“1000”,realm=“asterisk”,nonce=“647044e8”,uri="sip:[email protected];transport=UDP",response=“d96e673c3e5620f903ba8ae9c926ca55”,algorithm=MD5
    ÿEvent: presence.winfo
    ÿAllow-Events: presence, kpml
    ÿContent-Length: 0
    ÿ
    ÿ<------------->

4

Another way

  1. On Asterisk CLI “core set debug 1”
  2. Make some calls
  3. Goto freepbx module “Reports” -> “Asterisk Logfiles”
  4. Set value from 500 to 50000
  5. Copy and post some ERROR and WARNINGS
5

Here it is:

localhost*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:192.168.7.20:5060 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.7.20:5060;branch=z9hG4bKdcfe3ab2c67c28ea18ff40f1af80c03f;rport
From: "2" <sip:[email protected]>;tag=3071687493
To: <sip:[email protected];user=phone>
Call-ID: 3383719870@192_168_7_20
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: A510 IP/42.076.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 378

v=0
o=2 5018 70 IN IP4 192.168.7.20
s=Mapping
c=IN IP4 192.168.7.20
t=0 0
m=audio 5018 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (14 headers 17 lines) ---
Sending to 192.168.7.20:5060 (NAT)
Sending to 192.168.7.20:5060 (NAT)
Using INVITE request as basis request - 3383719870@192_168_7_20
Found peer '2' for '2' from 192.168.7.20:5060

<--- Reliably Transmitting (no NAT) to 192.168.7.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.7.20:5060;branch=z9hG4bKdcfe3ab2c67c28ea18ff40f1af80c03f;received=192.168.7.20;rport=5060
From: "2" <sip:[email protected]>;tag=3071687493
To: <sip:[email protected];user=phone>;tag=as4b326bda
Call-ID: 3383719870@192_168_7_20
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="463da2ae"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3383719870@192_168_7_20' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.7.20:5060 --->
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.7.20:5060;branch=z9hG4bKdcfe3ab2c67c28ea18ff40f1af80c03f;rport
From: "2" <sip:[email protected]>;tag=3071687493
To: <sip:[email protected];user=phone>;tag=as4b326bda
Call-ID: 3383719870@192_168_7_20
CSeq: 2 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: A510 IP/42.076.00.000.000
Content-Length: 0

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

<--- SIP read from UDP:192.168.7.20:5060 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.7.20:5060;branch=z9hG4bKfcdf2b72265cd8f46173fa3c60901b9c;rport
From: "2" <sip:[email protected]>;tag=3071687493
To: <sip:[email protected];user=phone>
Call-ID: 3383719870@192_168_7_20
CSeq: 3 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="463da2ae", response="d62dee4aa408c3bb5b2c424454fe2b5b"
Max-Forwards: 70
User-Agent: A510 IP/42.076.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 378

v=0
o=2 5018 70 IN IP4 192.168.7.20
s=Mapping
c=IN IP4 192.168.7.20
t=0 0
m=audio 5018 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (15 headers 17 lines) ---
Sending to 192.168.7.20:5060 (no NAT)
Using INVITE request as basis request - 3383719870@192_168_7_20
Found peer '2' for '2' from 192.168.7.20:5060
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 96
Found audio description format AAL2-G726-32 for ID 97
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726|ilbc|g726aal2|g722), peer - audio=(ulaw|alaw|g726|g729|g726aal2|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726|g726aal2|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.7.20:5018
Looking for 4010 in from-internal (domain 192.168.7.101)
list_route: hop: <sip:[email protected]:5060>

<--- Transmitting (no NAT) to 192.168.7.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.20:5060;branch=z9hG4bKfcdf2b72265cd8f46173fa3c60901b9c;received=192.168.7.20;rport=5060
From: "2" <sip:[email protected]>;tag=3071687493
To: <sip:[email protected];user=phone>
Call-ID: 3383719870@192_168_7_20
CSeq: 3 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
We think we can do text
Audio is at 11240
Adding codec 100012 (g722) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to xx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK0942279f
Max-Forwards: 70
From: "Andres" <sip:[email protected]>;tag=as06db4891
To: <sip:[email protected]>
Contact: <sip:xxx@xxx:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.11.0)
Date: Thu, 07 Aug 2014 17:51:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 840

v=0
o=root 1714351031 1714351031 IN IP4 xxx
s=Asterisk PBX 11.11.0
c=IN IP4 xxx
t=0 0
m=audio 11240 RTP/AVP 9 111 112 0 8 3 97 4 5 7 18 110 102 115 116 117 10 118 101
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- Transmitting (no NAT) to 192.168.7.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.7.20:5060;branch=z9hG4bKfcdf2b72265cd8f46173fa3c60901b9c;received=192.168.7.20;rport=5060
From: "2" <sip:[email protected]>;tag=3071687493
To: <sip:[email protected];user=phone>;tag=as487c912a
Call-ID: 3383719870@192_168_7_20
CSeq: 3 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:xx:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK0942279f;rport=44292;received=xxxx
From: "Andres" <sip:[email protected]>;tag=as06db4891
To: <sip:[email protected]>;tag=c040a69dfc7733bdec8c921a7a9f2d3a.e482
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="voip.xxx.it", nonce="53e479f39eb819ac326363d7608da6c202dd602e", qop="auth"
Server: SPS CI RM GW 04
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to xx:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK0942279f
Max-Forwards: 70
From: "Andres" <sip:[email protected]>;tag=as06db4891
To: <sip:[email protected]>;tag=c040a69dfc7733bdec8c921a7a9f2d3a.e482
Contact: <sip:xxx@xxx:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.11.0)
Content-Length: 0


---
We think we can do text
Audio is at 11240
Adding codec 100012 (g722) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to xx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK05f334e5
Max-Forwards: 70
From: "Andres" <sip:[email protected]>;tag=as06db4891
To: <sip:[email protected]>
Contact: <sip:xxx@xxx:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.11.0)
Proxy-Authorization: Digest username="04321842118", realm="voip.xxx.it", algorithm=MD5, uri="sip:[email protected]", nonce="53e479f39eb819ac326363d7608da6c202dd602e", response="5bfcb90d7150f50707defc1b43afb64e", qop=auth, cnonce="777e6167", nc=00000001
Date: Thu, 07 Aug 2014 17:51:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 840

v=0
o=root 1714351031 1714351032 IN IP4 xxx
s=Asterisk PBX 11.11.0
c=IN IP4 xxx
t=0 0
m=audio 11240 RTP/AVP 9 111 112 0 8 3 97 4 5 7 18 110 102 115 116 117 10 118 101
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:xx:5060 --->
SIP/2.0 403 From user does not match authenticated user
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK05f334e5;rport=44292;received=xxxx
From: "Andres" <sip:[email protected]>;tag=as06db4891
To: <sip:[email protected]>;tag=c040a69dfc7733bdec8c921a7a9f2d3a.1333
Call-ID: [email protected]
CSeq: 103 INVITE
Server: SPS CI RM GW 04
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to xx:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK05f334e5
Max-Forwards: 70
From: "Andres" <sip:[email protected]>;tag=as06db4891
To: <sip:[email protected]>;tag=c040a69dfc7733bdec8c921a7a9f2d3a.1333
Contact: <sip:xxx@xxx:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.11.0)
Content-Length: 0


---
[2014-08-07 19:51:09] WARNING[1577][C-00000001]: chan_sip.c:23167 handle_response_invite: Received response: "Forbidden" from '"Andres" <sip:[email protected]>;tag=as06db4891'
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
Audio is at 10122
Adding codec 100012 (g722) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.7.20:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.7.20:5060;branch=z9hG4bKfcdf2b72265cd8f46173fa3c60901b9c;received=192.168.7.20;rport=5060
From: "2" <sip:[email protected]>;tag=3071687493
To: <sip:[email protected];user=phone>;tag=as487c912a
Call-ID: 3383719870@192_168_7_20
CSeq: 3 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 347

v=0
o=root 2143877144 2143877144 IN IP4 192.168.7.101
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.7.101
t=0 0
m=audio 10122 RTP/AVP 9 2 97 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '3383719870@192_168_7_20' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 192.168.7.20:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.7.20:5060;branch=z9hG4bKfcdf2b72265cd8f46173fa3c60901b9c;received=192.168.7.20;rport=5060
From: "2" <sip:[email protected]>;tag=3071687493
To: <sip:[email protected];user=phone>;tag=as487c912a
Call-ID: 3383719870@192_168_7_20
CSeq: 3 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.7.20:5060 --->
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.7.20:5060;branch=z9hG4bKfcdf2b72265cd8f46173fa3c60901b9c;rport
From: "2" <sip:[email protected]>;tag=3071687493
To: <sip:[email protected];user=phone>;tag=as487c912a
Call-ID: 3383719870@192_168_7_20
CSeq: 3 ACK
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="463da2ae", response="d62dee4aa408c3bb5b2c424454fe2b5b"
Max-Forwards: 70
User-Agent: A510 IP/42.076.00.000.000
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
6

In the second way the only WARNING I get is this:

[2014-08-07 19:58:02] WARNING[1577][C-00000002]: chan_sip.c:23167 handle_response_invite: Received response: "Forbidden" from '"Andres" <sip:[email protected]>;tag=as037d2253'
7

@dicko look at this SIP debug please

8

A quick scan reveals that there is a probably a badly set up trunk/account with voip.xxx.it

9

But I receive normally incoming calls, here are setup:

Outgoing settings:

host=voip.yyy.it
username=nnn
secret=xxx
type=peer
qualify=yes
nat=auto
insecure=very
fromuser=nnn
fromdomain=voip.yyy.it
dtmfmode=rfc2833
allow=all

Incoming settings:

username=nnn
type=user
secret=xxx
nat=auto
insecure=very
host=voip.yyy.it
fromdomain=voip.yyy.it
dtmfmode=rfc2833
context=from-trunk
allow=all

Register string:

nnn:[email protected]/nnn

Need something else to understand better the problem, or if you have some specific indication for me would be very useful.
Thanks for your help!

Andrea

10

There are many variables to get right for a functional trunk, you should really check with yyy.it anything else would be a guess.

11

Ok, I’ve yet wrote to my Voip provider, I’m waiting for a reply, I will update you on this post.
Thanks for your kind help

Bye

Andrea

12

Hello Andrea,
Successfully solved the problem reported above?

I have a similar problem, but my one and follow me. When you enter an external call in the DDR and FreePBX tries to follow me in this same mistake. I have done internal testing by connecting an extension, it works like this. Only external call the failure occurs.

Help me please!

13

Bonjour,
OVH bloque les appels si le numéro de l’appelant ne corespond pas.

Pour savoir si le probleme viens de la, il suffit d’appeler en masquer.

hey,
OVH block calls if the number of the caller does not corespond .

To see if the problem come from , it Just to hide in Call.

14

Did you ever solve this problem?

I’m getting these periodically for outbound calls with Asterisk 13,
Received response: “Forbidden” from …

15

Logs from the time around the error would help. Otherwise, you’re asking a question that’s far too general for anyone to give you a reasonable answer. By the way, have you asked your provider to troubleshoot it with you?

16

I understand, but there was never a conclusion. I’ve been working with the provider for quite some time. We recently changed the registration timeout setting to 195 seconds, which may have helped. This might be an issue on the provider side where the connection retries too soon, and gets blocked by the provider.

17

Your provider is the one that’s keeping you from registering.

Are you using actual registrations or are you IP-bound to a connection? If you are using IP authentication, your registration attempts should fail. If not, the provider should be able to tell you what failed on their end and stopped you from registering.