SIP 2.0 404 not found error

[scopor]

HI all,

I have a vicibox 5 i686-5.0.3 |Vicidial VERSION:2.8.517a Build:131019-0849|Asterisk 1.8.23.1-Vici|Single Server|No Digium/Sangoma Hardware|No Extra Software After Installation|Intel Xeon 2.4

I have an xconnect sip trunk between my local and hosted vicidial. I forward calls coming to my local to my hosted vici and handle all calls from hosted. To dial out local calls my hosted would connect to my local vici via xconnect to access the trunks. However I am getting one way audio if I place a call from my hosted. The agent hear the called party but the called party does not hear anything. If I call from the local vicibox there is two way audio. Incoming call to local vicibox suppose to be routed to hosted but this is not happening. What could be wrong.?


What you see below is me placing a call from my cellphone 8684992727, to 8682235013. My vicibox suppose to receive the call and forward it to my hosted vicidial. This use to work before but I relocated to a new office and had to use the internet connection that was once on my vicibox as my main internet through my firewall. Please see below my sip debug while trying to place the call. Even without a call I am seeing 404 trying to reach my hosted vicidial.

asterisk -r
Asterisk 1.8.23.0-vici, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.23.0-vici currently running on vici1 (pid = 2561)
Verbosity is at least 21
[Feb 2 23:06:50] -- Executing [h@default:1] AGI("SIP/xconnect-0000000f", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----102-----94") in new stack
[Feb 2 23:06:50] -- <SIP/xconnect-0000000f>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... 102-----94 completed, returning 0
[Feb 2 23:06:51] == Spawn extension (default, 718684992727, 2) exited non-zero on 'SIP/xconnect-0000000f'
[Feb 2 23:06:51] Scheduling destruction of SIP dialog '4c57b97845b83bd35223438e73aa184c@173.241.198.169' in 6400 ms (Method: ACK)
[Feb 2 23:06:51] set_destination: Parsing <sip:8682231200@173.241.198.169> for address/port to send to
[Feb 2 23:06:51] set_destination: set destination to 173.241.198.169:5060
[Feb 2 23:06:51] Reliably Transmitting (no NAT) to 173.241.198.169:5060:
BYE sip:8682231200@173.241.198.169 SIP/2.0
Via: SIP/2.0/UDP 190.213.2.2:5060;branch=z9hG4bK75bbf402;rport
Max-Forwards: 70
From: <sip:718684992727@190.213.2.2;cpd=on>;tag=as1b857a2a
To: "M2022204320000581312" <sip:8682231200@173.241.198.169>;tag=as3a213b0c
Call-ID: 4c57b97845b83bd35223438e73aa184c@173.241.198.169
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.23.0-vici
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Feb 2 23:06:51]
<--- SIP read from UDP:173.241.198.169:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.213.2.2;branch=z9hG4bK75bbf402;received=190.213.2.2;rport=5060
From: <sip:718684992727@190.213.2.2;cpd=on>;tag=as1b857a2a
To: "M2022204320000581312" <sip:8682231200@173.241.198.169>;tag=as3a213b0c
Call-ID: 4c57b97845b83bd35223438e73aa184c@173.241.198.169
CSeq: 102 BYE
User-Agent: ViciDial
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
[Feb 2 23:06:51] --- (10 headers 0 lines) ---
[Feb 2 23:06:51] SIP Response message for INCOMING dialog BYE arrived
[Feb 2 23:06:51] Really destroying SIP dialog '4c57b97845b83bd35223438e73aa184c@173.241.198.169' Method: ACK
[Feb 2 23:06:52]
<--- SIP read from UDP:173.241.198.169:5060 --->
OPTIONS sip:192.168.0.6:5060;cpd=on SIP/2.0
From: "asterisk" <sip:asterisk@173.241.198.169>;tag=as66fc6fd8
To: <sip:190.213.2.2;cpd=on>
Contact: <sip:asterisk@173.241.198.169>
Call-ID: 7a2970e97a207f61059f4351163e4b57@173.241.198.169
CSeq: 102 OPTIONS
User-Agent: ViciDial
Max-Forwards: 70
Date: Mon, 03 Feb 2014 03:06:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
Via: SIP/2.0/UDP 173.241.198.169;rport;branch=z9hG4bK2679b3a7

<------------->
[Feb 2 23:06:52] --- (13 headers 0 lines) ---
[Feb 2 23:06:52] Looking for s in trunkinbound (domain 192.168.0.6)
[Feb 2 23:06:52]
<--- Transmitting (no NAT) to 173.241.198.169:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 173.241.198.169;rport;branch=z9hG4bK2679b3a7;received=173.241.198.169
From: "asterisk" <sip:asterisk@173.241.198.169>;tag=as66fc6fd8
To: <sip:190.213.2.2;cpd=on>;tag=as5e617432
Call-ID: 7a2970e97a207f61059f4351163e4b57@173.241.198.169
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.23.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Feb 2 23:06:52] Scheduling destruction of SIP dialog '7a2970e97a207f61059f4351163e4b57@173.241.198.169' in 32000 ms (Method: OPTIONS)
[Feb 2 23:07:02] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 2 23:07:02] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 2 23:07:02] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 2 23:07:04] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 2 23:07:07] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 2 23:07:07] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 2 23:07:24] Really destroying SIP dialog '7a2970e97a207f61059f4351163e4b57@173.241.198.169' Method: OPTIONS
[Feb 2 23:07:29] NOTICE[2634]: chan_sip.c:15122 check_auth: Correct auth, but based on stale nonce received from '"6000"<sip:6000@192.168.0.6>;tag=910e6638'
[Feb 2 23:07:42] Reliably Transmitting (no NAT) to 173.241.198.169:5060:
OPTIONS sip:dial121.vicihost.com SIP/2.0
Via: SIP/2.0/UDP 190.213.2.2:5060;branch=z9hG4bK287a3a0b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@190.213.2.2>;tag=as105894bc
To: <sip:dial121.vicihost.com>
Contact: <sip:asterisk@190.213.2.2:5060>
Call-ID: 537d864e3c7658f84853392775117c05@190.213.2.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-vici
Date: Mon, 03 Feb 2014 03:07:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[williamconley]

one way sound is always firewall.

if you have an inbound call from an external server through a firewall ... that firewall will often forward the control channel but NOT the audio channel. control is 5060, but audio is anywhere from 10000 to 25000 (UDP). If your "local" vicidial had its own external IP address, this would not be an issue.

Be sure your vicidial servers both have "externip" set in sip.conf (to the Public IP address that the server will use to communicate off the local network). Be sure you are not trying to "double NAT" with a router incapable of this. Note that SIP is NOT capable of double NATting. So if your router can't handle SIP, then having a router at the start and stop ends of the calls is impossible. some routers have a SIP algorithm specifically to allow this, some have "triggering" that can be configured (so UDP port 5060 is allowed to Trigger a port to be open between 10000 and 25000, thus sending that traffic to the same ip on the internal network).

Often it is possible to Register the local asterisk server to the remote/external asterisk server which will open a specific port (and convey the port number to the external asterisk server!).

[scopor]

Hi William,

Thanks for your quick reply. The one way audio is not the lan but on the far side. Ie. I as the agent on the lan side can hear the person on his/her cell. However they can't hear me. Is that still the firewall?

I had separate internet for vicibox but since relocating no longer have separate internet. But I previously had this working with one internet through firewall. Can't firgure out why it stop working. I have set udp port 10000-25000 on firewall for inbound stream. Prior to now I had it udp any port still to no avail.

[geoff3dmg]

Audio only working in one direction is a classic indication of a firewall/NAT issue.