vicidial.org

Posted: **Wed Sep 29, 2010 7:24 am**

I have install vicibox server 2.0, on LAN server ip 192.168.0.40
No extra hardware| no extra software| firewall is not running

when i am dialing from sip client, call is not reaches the dial plan,

Error msg
chan_sip.c:15147 handle_request_invite: Call from 'cc101' to extension '999' rejected because extension not found.

sip user

[cc101]
disallow=all
allow=ulaw
type=friend
username=ccccc
secret=test
host=dynamic
dtmfmode=rfc2833; inband
context=default
Nat=yes
insecure=invite
externip = 192.168.0.164

here is sip debug

[Sep 29 16:57:00]
<--- SIP read from 192.168.0.164:47812 --->
INVITE sip:999@192.168.0.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.164:47812;branch=z9hG4bK-d8754z-0330d800f2025c31-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc101@192.168.0.164:47812>
To: "999"<sip:999@192.168.0.40>
From: "cc101"<sip:cc101@192.168.0.40>;tag=e34f2c7e
Call-ID: ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 265

v=0
o=- 1 2 IN IP4 192.168.0.164
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.164
t=0 0
m=audio 8922 RTP/AVP 107 0 8 101
a=alt:1 1 : kCcD3GZy 2MqGGLhe 192.168.0.164 8922
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 29 16:57:00] --- (12 headers 11 lines) ---
[Sep 29 16:57:00] Sending to 192.168.0.164 : 47812 (NAT)
[Sep 29 16:57:00] Using INVITE request as basis request - ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.
[Sep 29 16:57:00]
<--- Reliably Transmitting (NAT) to 192.168.0.164:47812 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.164:47812;branch=z9hG4bK-d8754z-0330d800f2025c31-1---d8754z-;received=192.168.0.164;rport=47812
From: "cc101"<sip:cc101@192.168.0.40>;tag=e34f2c7e
To: "999"<sip:999@192.168.0.40>;tag=as2afa30c3
Call-ID: ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="550e8581"
Content-Length: 0

<------------>
[Sep 29 16:57:00] Scheduling destruction of SIP dialog 'ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.' in 32000 ms (Method: INVITE)
[Sep 29 16:57:00] Found user 'cc101'
[Sep 29 16:57:00]
<--- SIP read from 192.168.0.164:47812 --->
ACK sip:999@192.168.0.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.164:47812;branch=z9hG4bK-d8754z-0330d800f2025c31-1---d8754z-;rport
To: "999"<sip:999@192.168.0.40>;tag=as2afa30c3
From: "cc101"<sip:cc101@192.168.0.40>;tag=e34f2c7e
Call-ID: ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.
CSeq: 1 ACK
Content-Length: 0

<------------->
[Sep 29 16:57:00] --- (7 headers 0 lines) ---
[Sep 29 16:57:00]
<--- SIP read from 192.168.0.164:47812 --->
INVITE sip:999@192.168.0.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.164:47812;branch=z9hG4bK-d8754z-d3180c29380abb4b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc101@192.168.0.164:47812>
To: "999"<sip:999@192.168.0.40>
From: "cc101"<sip:cc101@192.168.0.40>;tag=e34f2c7e
Call-ID: ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="cc101",realm="asterisk",nonce="550e8581",uri="sip:999@192.168.0.40",response="ce8f98190882a6923bb12bd2ceee8fec",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 265

v=0
o=- 1 2 IN IP4 192.168.0.164
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.164
t=0 0
m=audio 8922 RTP/AVP 107 0 8 101
a=alt:1 1 : kCcD3GZy 2MqGGLhe 192.168.0.164 8922
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 29 16:57:00] --- (13 headers 11 lines) ---
[Sep 29 16:57:00] Sending to 192.168.0.164 : 47812 (NAT)
[Sep 29 16:57:00] Using INVITE request as basis request - ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.
[Sep 29 16:57:00] Found user 'cc101'
[Sep 29 16:57:00] Found RTP audio format 107
[Sep 29 16:57:00] Found RTP audio format 0
[Sep 29 16:57:00] Found RTP audio format 8
[Sep 29 16:57:00] Found RTP audio format 101
[Sep 29 16:57:00] Found unknown media description format BV32 for ID 107
[Sep 29 16:57:00] Found audio description format telephone-event for ID 101
[Sep 29 16:57:00] Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Sep 29 16:57:00] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 29 16:57:00] Peer audio RTP is at port 192.168.0.164:8922
[Sep 29 16:57:00] Looking for 999 in default (domain 192.168.0.40)
[Sep 29 16:57:00]
<--- Reliably Transmitting (NAT) to 192.168.0.164:47812 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.164:47812;branch=z9hG4bK-d8754z-d3180c29380abb4b-1---d8754z-;received=192.168.0.164;rport=47812
From: "cc101"<sip:cc101@192.168.0.40>;tag=e34f2c7e
To: "999"<sip:999@192.168.0.40>;tag=as2afa30c3
Call-ID: ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
[Sep 29 16:57:00] NOTICE[3609]: chan_sip.c:15147 handle_request_invite: Call from 'cc101' to extension '999' rejected because extension not found.
[Sep 29 16:57:00] Scheduling destruction of SIP dialog 'ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.' in 32000 ms (Method: INVITE)
[Sep 29 16:57:00]
<--- SIP read from 192.168.0.164:47812 --->
ACK sip:999@192.168.0.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.164:47812;branch=z9hG4bK-d8754z-d3180c29380abb4b-1---d8754z-;rport
To: "999"<sip:999@192.168.0.40>;tag=as2afa30c3
From: "cc101"<sip:cc101@192.168.0.40>;tag=e34f2c7e
Call-ID: ZmRkNTAyYTI1ZTBhZDFjNTBkYTYzYzgyZGVlMjZiNDA.
CSeq: 2 ACK
Content-Length: 0

<------------->
[Sep 29 16:57:00] --- (7 headers 0 lines) ---
[Sep 29 16:57:01] == Parsing '/etc/asterisk/manager.conf': [Sep 29 16:57:01] Found
[Sep 29 16:57:01] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 29 16:57:01] == Parsing '/etc/asterisk/manager.conf': [Sep 29 16:57:01] Found
[Sep 29 16:57:01] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 29 16:57:01] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 29 16:57:01] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 29 16:57:02] Really destroying SIP dialog 'MGE4MmZlZTk5MmI0MWNiMWQ3Yzc3NzI3NDFjYzNhMWE.' Method: ACK
[Sep 29 16:57:02]
<--- SIP read from 192.168.0.164:47812 --->

plz guide

Posted: **Wed Sep 29, 2010 3:42 pm**

1) you showed your install information very well, but you left off the asterisk version and "cluster" information. kudos for the attempt, though!

2) nat=no when you are both on the same local subnet

NAT = network address translation, which is something that firewalls/routers do when there is a different local subnet from the network outside the firewall ... the router "translates" the packets and sends them to places not reachable from the other side of the firewall. since your sip phone and your sip server (asterisk/vicidial) are on the same local subnet (192.168.0 is the local subnet), there is no "nat" occurring, so nat=no.

Posted: **Fri Oct 01, 2010 1:48 am**

Thanks williamconley.

Asterisk 1.4.27.1-vici
vicidial VERSION: 2.4-264 BUILD: 100720-1332

I run it with nat=no but problem not solve, i also installed dahdi (sangoma analog FXO ) its too not receiving calls,

here is some new conf & status

extension.conf
[sipcalls]

exten => s,1,Noop(in s)
exten => s,2,echo()
exten => s,3,hangup

exten => 999,1,Noop(In ext 999)
exten => 999,2,Answer
exten => 999,3,wait(1)
exten => 999,4,Hangup

sip-vicidial.conf
[cc101]
username=cc101
secret=test
accountcode=cc101
callerid="cc101" <101>
mailbox=101
context=default
type=friend
host=dynamic
nat=no

sip result
======
[Oct 1 11:34:52] NOTICE[3519]: chan_sip.c:15147 handle_request_invite: Call from 'cc101' to extension '999' rejected because extension not found.

Dahdi result (2FSX-2FXO)
=====================

[Oct 1 11:35:25] -- Starting simple switch on 'DAHDI/4-1'
[Oct 1 11:35:25] == Starting DAHDI/4-1 at sipcalls,s,1 failed so falling back to exten 's'
[Oct 1 11:35:25] == Starting DAHDI/4-1 at sipcalls,s,1 still failed so falling back to context 'default'
[Oct 1 11:35:25] WARNING[5419]: pbx.c:2456 __ast_pbx_run: Channel 'DAHDI/4-1' sent into invalid extension 's' in context 'default', but no invalid handler
[Oct 1 11:35:25] -- Hungup 'DAHDI/4-1'
[Oct 1 11:35:30] -- Starting simple switch on 'DAHDI/4-1'
[Oct 1 11:35:33] NOTICE[5431]: chan_dahdi.c:6846 ss_thread: Got event 18 (Ring Begin)...
[Oct 1 11:35:35] NOTICE[5431]: chan_dahdi.c:6846 ss_thread: Got event 2 (Ring/Answered)...
[Oct 1 11:35:35] == Starting DAHDI/4-1 at sipcalls,s,1 failed so falling back to exten 's'
[Oct 1 11:35:35] == Starting DAHDI/4-1 at sipcalls,s,1 still failed so falling back to context 'default'
[Oct 1 11:35:35] WARNING[5431]: pbx.c:2456 __ast_pbx_run: Channel 'DAHDI/4-1' sent into invalid extension 's' in context 'default', but no invalid handler
[Oct 1 11:35:35] -- Hungup 'DAHDI/4-1'

sip debug
=========

ccserver*CLI>
ccserver*CLI>
[Oct 1 11:47:43]
<--- SIP read from 192.168.0.164:60396 --->
INVITE sip:999@192.168.0.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.164:60396;branch=z9hG4bK-d8754z-cd0058527d14537a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc101@192.168.0.164:60396>
To: "999"<sip:999@192.168.0.40>
From: "ccagent1"<sip:cc101@192.168.0.40>;tag=d9508f44
Call-ID: MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 267

v=0
o=- 9 2 IN IP4 192.168.0.164
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.164
t=0 0
m=audio 40970 RTP/AVP 107 0 8 101
a=alt:1 1 : +G6JQkK8 irWgG9sR 192.168.0.164 40970
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Oct 1 11:47:43] --- (12 headers 11 lines) ---
[Oct 1 11:47:43] Sending to 192.168.0.164 : 60396 (NAT)
[Oct 1 11:47:43] Using INVITE request as basis request - MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.
[Oct 1 11:47:43]
<--- Reliably Transmitting (no NAT) to 192.168.0.164:60396 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.164:60396;branch=z9hG4bK-d8754z-cd0058527d14537a-1---d8754z-;received=192.168.0.164;rport=60396
From: "ccagent1"<sip:cc101@192.168.0.40>;tag=d9508f44
To: "999"<sip:999@192.168.0.40>;tag=as0c890e28
Call-ID: MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="655880d0"
Content-Length: 0

<------------>
[Oct 1 11:47:43] Scheduling destruction of SIP dialog 'MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.' in 32000 ms (Method: INVITE)
[Oct 1 11:47:43] Found user 'cc101'
[Oct 1 11:47:43]
<--- SIP read from 192.168.0.164:60396 --->
ACK sip:999@192.168.0.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.164:60396;branch=z9hG4bK-d8754z-cd0058527d14537a-1---d8754z-;rport
To: "999"<sip:999@192.168.0.40>;tag=as0c890e28
From: "ccagent1"<sip:cc101@192.168.0.40>;tag=d9508f44
Call-ID: MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.
CSeq: 1 ACK
Content-Length: 0

<------------->
[Oct 1 11:47:43] --- (7 headers 0 lines) ---
[Oct 1 11:47:43]
<--- SIP read from 192.168.0.164:60396 --->
INVITE sip:999@192.168.0.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.164:60396;branch=z9hG4bK-d8754z-ef736a7118503544-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc101@192.168.0.164:60396>
To: "999"<sip:999@192.168.0.40>
From: "ccagent1"<sip:cc101@192.168.0.40>;tag=d9508f44
Call-ID: MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="cc101",realm="asterisk",nonce="655880d0",uri="sip:999@192.168.0.40",response="543895bd556173bd082ca0d612d8ae78",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 267

v=0
o=- 9 2 IN IP4 192.168.0.164
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.164
t=0 0
m=audio 40970 RTP/AVP 107 0 8 101
a=alt:1 1 : +G6JQkK8 irWgG9sR 192.168.0.164 40970
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Oct 1 11:47:43] --- (13 headers 11 lines) ---
[Oct 1 11:47:43] Sending to 192.168.0.164 : 60396 (NAT)
[Oct 1 11:47:43] Using INVITE request as basis request - MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.
[Oct 1 11:47:43] Found user 'cc101'
[Oct 1 11:47:43] Found RTP audio format 107
[Oct 1 11:47:43] Found RTP audio format 0
[Oct 1 11:47:43] Found RTP audio format 8
[Oct 1 11:47:43] Found RTP audio format 101
[Oct 1 11:47:43] Found unknown media description format BV32 for ID 107
[Oct 1 11:47:43] Found audio description format telephone-event for ID 101
[Oct 1 11:47:43] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Oct 1 11:47:43] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 1 11:47:43] Peer audio RTP is at port 192.168.0.164:40970
[Oct 1 11:47:43] Looking for 999 in default (domain 192.168.0.40)
[Oct 1 11:47:43]
<--- Reliably Transmitting (NAT) to 192.168.0.164:60396 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.164:60396;branch=z9hG4bK-d8754z-ef736a7118503544-1---d8754z-;received=192.168.0.164;rport=60396
From: "ccagent1"<sip:cc101@192.168.0.40>;tag=d9508f44
To: "999"<sip:999@192.168.0.40>;tag=as0c890e28
Call-ID: MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
[Oct 1 11:47:43] NOTICE[3519]: chan_sip.c:15147 handle_request_invite: Call from 'cc101' to extension '999' rejected because extension not found.
[Oct 1 11:47:43] Scheduling destruction of SIP dialog 'MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.' in 32000 ms (Method: INVITE)
[Oct 1 11:47:43]
<--- SIP read from 192.168.0.164:60396 --->
ACK sip:999@192.168.0.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.164:60396;branch=z9hG4bK-d8754z-ef736a7118503544-1---d8754z-;rport
To: "999"<sip:999@192.168.0.40>;tag=as0c890e28
From: "ccagent1"<sip:cc101@192.168.0.40>;tag=d9508f44
Call-ID: MWU1YTQ3MmVkZmUzNDAxYTVlMjA3NjFlMmJlMzlmNTE.
CSeq: 2 ACK
Content-Length: 0

<------------->
[Oct 1 11:47:43] --- (7 headers 0 lines) ---
ccserver*CLI>

Posted: **Fri Oct 01, 2010 9:46 am**

what did not work exactly? the agent trying to dial "999"?

Oct 1 11:47:43] NOTICE[3519]: chan_sip.c:15147 handle_request_invite: Call from 'cc101' to extension '999' rejected because extension not found.

Posted: **Sun Oct 03, 2010 1:04 pm**

williamconley wrote:what did not work exactly? the agent trying to dial "999"?

yes agent trying to dial 999, but any call is not connecting to dialplan extern

Posted: **Tue Oct 05, 2010 7:54 pm**

i think what we'd need to troubleshoot a call not working properly is a REAL number being dialed that should have succeeded.

showing us the output from a "not gonna work!" call does not help us to resolve why it didn't work. (It failed because there is NO 999).

can you attempt a real call and show us ONLY the output from that failure?

Posted: **Thu Oct 07, 2010 12:53 am**

Thanks williamconley for your reply, problem solve last night,
999 was exits in dialplan. Even calls (dahdi/sip) was not connecting to vicidial sample dialplan exten like 8307 0r 8304.

Step I take to solve

1)
problem was solve when i recompile dahdi, asterisk manually

I also install vicibox multiple time to got reason

2)
I attached sangoma analog FXS-FXO card, installed new vicibox 3.0.5, when setup ask "Do you have Telephony ISDN Card" is say "No" & install wanpipe manully at end of installation. Same result dialplan was failed to receive call of any type.

3)
With sangoma analog FXS-FXO card, I reinstall vicibox 3.0.5, say yes to query "Do you have Telephony ISDN Card", I know it was wrong i dont have isdn telephony card but i select y so vicibox install wanpipe it self, after installation i only configure wanpipe, its was taking calls sip/dahdi

yap strange but ture :-D

, may help someone else
[/b]

Posted: **Thu Oct 07, 2010 3:44 pm**

That question determined whether or not ViciBox installs the wanpipe drivers. Since we rarely see a FXO/FXS card being utilized with ViciDial, the word "T1" is used there.

I'll make a note of it and change it to something more generic in the next release.

Posted: **Fri Oct 08, 2010 6:07 am**

Kumba wrote:That question determined whether or not ViciBox installs the wanpipe drivers. Since we rarely see a FXO/FXS card being utilized with ViciDial, the word "T1" is used there.

Kumba wanpipe is a generic driver for sangoma cards either T1/E1 or analog, main thing is to configuration your installed card model after installation.

I am using FSO-FXS with vicidial, but with one core limitation i.e DNIS detection

vicidial.org

vicibox inbound Sip call not working

vicibox inbound Sip call not working