vbuzzer - here we go again!!!

syedamerali · #1 (**permalink**) April 29th, 2006, 03:39 PM

I recently upgraded to AAH 2.8 and guess what? Vbuzzer incoming stopped working .....
The situation is as follows.
1. My IP is mapped to syedamerali.dyndns.org
2. I have a vbuzzer softphone on ip 192.168.1.100
3. My Asterisk box is on ip 192.168.1.110 which is in a DMZ.

Outgoing is fine
Incoming - nothing gets in to my asterisk box if the vbuzzer softphone has not registered - but if the softphone has registered asterisk incoming works fine.

The sip debug from CLI is:

--- (9 headers 0 lines)---
asterisk1*CLI>
<-- SIP read from 209.47.41.24:80:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK4b63f6e3;rport=50 50;received=203.81.220.13
From: <sip:syedamerali@vbuzzer.com>;tag=as7bed3895
To: <sip:syedamerali@vbuzzer.com>;tag=0137240595b7ec88 199a5a4d45fb6597.195a
Call-ID: 38849bd7103ce0bd1c90c1ee3d5b73d2@sye...i.dyndn s.org:5050
CSeq: 107 REGISTER
Contact: <sip:13473293686@203.81.220.13:5050>;expires=120
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.24:80 "Noisy feedback tells: pid=2334 req_src_ip=203.81.220.13 req_src_port=5050 in_uri=sip:vbuzzer.com out_uri=sip:vbuzzer.com via_cnt==1"

--- (10 headers 0 lines)---
Scheduling destruction of call '38849bd7103ce0bd1c90c1ee3d5b73d2@syedamerali.dynd ns.org:5050' in 32000 ms
Destroying call '38849bd7103ce0bd1c90c1ee3d5b73d2@syedamerali.dynd ns.org:5050'
12 headers, 0 lines
Reliably Transmitting (NAT) to 209.47.41.24:80:
OPTIONS sip:vbuzzer.com:80 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK03141923;rport
From: "syedamerali" <sip:syedamerali@syedamerali.dyndns.org:5050>;tag= as2f3aa8ba
To: <sip:vbuzzer.com:80>
Contact: <sip:syedamerali@192.168.1.110:5050>
Call-ID: 7d2a5c757917950a09233a2c24b71e23@sye...i.dyndn s.org:5050
CSeq: 102 OPTIONS
User-Agent: VBuzzer/1.0(v1.1.1.2-2006.03.29)
Max-Forwards: 70
Date: Sat, 29 Apr 2006 14:20:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

The sip debug is:
---
asterisk1*CLI>
<-- SIP read from 209.47.41.24:80:
SIP/2.0 200 Keep Alive Works
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK03141923;rport=50 50;received=203.81.220.13
From: "syedamerali" <sip:syedamerali@syedamerali.dyndns.org:5050>;tag= as2f3aa8ba
To: <sip:vbuzzer.com:80>;tag=0137240595b7ec88199a5a4d4 5fb6597.7542
Call-ID: 7d2a5c757917950a09233a2c24b71e23@sye...i.dyndn s.org:5050
CSeq: 102 OPTIONS
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.24:80 "Noisy feedback tells: pid=2332 req_src_ip=203.81.220.13 req_src_port=5050 in_uri=sip:vbuzzer.com:80 out_uri=sip:vbuzzer.com:80 via_cnt==1"

asterisk info is:

Verbosity is at least 3

Sip Registry

Name/username Host Dyn Nat ACL Port Status
vbuzzer/syedamerali 209.47.41.24 N 80 OK (354 ms)
7589522/asterisk 192.168.1.120 5204 OK (10 ms)
756 9612/asterisk 192.168.1.121 5206 OK (9 ms)
210/210 192.168.1.121 D 5210 Unmonitored
208/208 192.168.1.122 D 5208 Unmonitored
206/206 192.168.1.121 D 5206 Unmonitored
204/204 192.168.1.120 D 5204 Unmonitored
202/202 192.168.1.120 D 5202 Unmonitored
200/200 192.168.1.122 D 5200 Unmonitored
9 sip peers [9 online , 0 offline]
Verbosity is at least 3

Sip Peers

Host Username Refresh State
vbuzzer.com:80 syedamerali 105 Registered
Verbosity is at least 3

settings are:

[vbuzzer]
allow=g729
auth=md5
authname=syedamerali
canreinvite=no
context=from-internal-custom
disallow=all
dtmfmode=RFC2833
fromdomain=vbuzzer.com
fromuser=syedamerali
hidecallerid=yes
host=vbuzzer.com
insecure=very
nat=yes
port=80
qualify=3000
restrictcid=yes
secret=XXXXXX
type=friend
user=syedamerali
username=syedamerali

[from-internal-custom]
exten => _1347XX.,1,NoOp(Incoming call from vbuzzer 347 extension)
exten => _1347XX.,2,Answer()
exten => _1347XX.,3,Dial(SIP/210&SIP/200&SIP/202,60,Ttr)
exten => _1347XX.,4,Hangup

exten => s,1,Answer()
exten => s,2,Wait(1)
exten => s,3,Dial(SIP/210&SIP/200,60,Tt)
exten => s,5,Hangup
Register String
syedamerali:XXXXXX:syedamerali@vbuzzer.com:80/13473293686

Any ideas anybody?
kindest regards

syedamerali · #2 (**permalink**) May 1st, 2006, 03:29 PM

Footnote - Works with a Sipura

chandave · #3 (**permalink**) May 1st, 2006, 04:05 PM

Set the verbosity to 5 and call your DID. We are interested in what your systems does with the SIP INVITE.

See ya...

d.c.

syedamerali · #4 (**permalink**) May 2nd, 2006, 12:51 AM

Chandave
did as you suggested - nothing gets through to the asterisk box !
kindest regards

chandave · #5 (**permalink**) May 2nd, 2006, 05:58 AM

Interesting. According to your SIP 200 OK:
Contact: <sip:13473293686@203.81.220.13:5050>
agrees with what VBuzzer saw as your source info for SIP:
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK4b63f6e3;rport=50 50;received=203.81.220.13

If you don't see Asterisk's CLI move at all during an inbound call, then that means Asterisk isn't getting the SIP INVITE.

There could be 2 possibilities:
1) VBuzzer is not sending it to you
2) VBuzzer sends the INVITE to you but Asterisk isn't gettting

On the Asterisk box, start up a TCPDump session to capture the SIP packets. Run the following command as root at the Linux prompt:

/usr/sbin/tcpdump -l - eth0 -s 1500 -w /tmp/vbuzzer.cap host vbuzzer.com and udp port 80

This will capture all SIP packets to and from vbuzzer.com. While the tcpdump is running, dial into your DID. After you get your VM, hangup and terminate the tcpdump. Now, run:
/usr/sbin/tcpdump -l -s 1500 -r /tmp/vbuzzer.cap -X
and see if you see any SIP INVITEs from vbuzzer. If you do see the SIP INVITE, then Asterisk is ignoring it for some reason.

See ya...

d.c.

syedamerali · #6 (**permalink**) May 2nd, 2006, 03:04 PM

Chandev
1. I suddenly could not register - but I managed to register by changing my register string to the following:
syedamerali:XXXXXXX:syedamerali@vbuzzer:80/13473293686

2. I did what you suggested nothing coming into the asterisk box.
3. I fired up the vbuzzer softphone and made it register and the INVITES started coming through.
4. I have the feeling that although asterisk says I have registered vbuzzer does not recognise it. It is only when the softphone registers that vbuzzer recognises the registration and hence sends the INVITE to asterisk.

The sip debug with the softphone registered is:

--- (9 headers 0 lines)---
Destroying call '025beb001685f6535ea4f5793ff5ac6d@syedamerali.dynd ns.org:5050'
asterisk1*CLI>
<-- SIP read from 209.47.41.24:80:
INVITE sip:13473293686@203.81.224.12:5050 SIP/2.0
Record-Route: <sip:209.47.41.24:80;ftag=as2c012150;lr=on>
Record-Route: <sip:209.47.41.48:80;ftag=as2c012150;lr=on>
Via: SIP/2.0/UDP 209.47.41.24:80;branch=z9hG4bK8ea.18877405.0
Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK8ea.ef361107.0
Via: SIP/2.0/UDP 207.111.170.20:5060;branch=z9hG4bK4e16a60b;rport=5 060
From: "+923218415358" <sip:+923218415358@207.111.170.20>;tag=as2c01215 0
To: <sip:3473293686@209.47.41.48:80>
Contact: <sip:+923218415358@207.111.170.20:5060>
Call-ID: 12a557b129780bee0949007c2837b14c@207.111.170.20
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 15
Date: Tue, 02 May 2006 13:51:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 23158 23158 IN IP4 207.111.170.20
s=session
c=IN IP4 209.47.41.26
t=0 0
m=audio 51512 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=nortpproxy:yes

--- (17 headers 11 lines)---
Using INVITE request as basis request - 12a557b129780bee0949007c2837b14c@207.111.170.20
Sending to 209.47.41.24 : 80 (NAT)
Found peer 'vbuzzer'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.47.41.26:51512
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Transmitting (NAT) to 209.47.41.24:80:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 209.47.41.24:80;branch=z9hG4bK8ea.18877405.0;recei ved=209.47.41.24
Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK8ea.ef361107.0
Via: SIP/2.0/UDP 207.111.170.20:5060;branch=z9hG4bK4e16a60b;rport=5 060
From: "+923218415358" <sip:+923218415358@207.111.170.20>;tag=as2c01215 0
To: <sip:3473293686@209.47.41.48:80>;tag=as15a6209b
Call-ID: 12a557b129780bee0949007c2837b14c@207.111.170.20
CSeq: 102 INVITE
User-Agent: VBuzzer/1.0(v1.1.1.2-2006.03.29)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:13473293686@192.168.1.110:5050>
Content-Length: 0

---
Destroying call '12a557b129780bee0949007c2837b14c@207.111.170.20'
Destroying call '5020e17463aa951a63c4d57811715226@vbuzzer.com'
asterisk1*CLI>
<-- SIP read from 209.47.41.24:80:
INVITE sip:13473293686@203.81.224.12:5050 SIP/2.0
Record-Route: <sip:209.47.41.24:80;ftag=as2c012150;lr=on>
Record-Route: <sip:209.47.41.48:80;ftag=as2c012150;lr=on>
Via: SIP/2.0/UDP 209.47.41.24:80;branch=z9hG4bK8ea.18877405.0
Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK8ea.ef361107.0
Via: SIP/2.0/UDP 207.111.170.20:5060;branch=z9hG4bK4e16a60b;rport=5 060
From: "+923218415358" <sip:+923218415358@207.111.170.20>;tag=as2c01215 0
To: <sip:3473293686@209.47.41.48:80>
Contact: <sip:+923218415358@207.111.170.20:5060>
Call-ID: 12a557b129780bee0949007c2837b14c@207.111.170.20
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 15
Date: Tue, 02 May 2006 13:51:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 23158 23158 IN IP4 207.111.170.20
s=session
c=IN IP4 209.47.41.26
t=0 0
m=audio 51512 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=nortpproxy:yes

--- (17 headers 11 lines)---
Using INVITE request as basis request - 12a557b129780bee0949007c2837b14c@207.111.170.20
Sending to 209.47.41.24 : 80 (NAT)
Found peer 'vbuzzer'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.47.41.26:51512
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Transmitting (NAT) to 209.47.41.24:80:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 209.47.41.24:80;branch=z9hG4bK8ea.18877405.0;recei ved=209.47.41.24
Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK8ea.ef361107.0
Via: SIP/2.0/UDP 207.111.170.20:5060;branch=z9hG4bK4e16a60b;rport=5 060
From: "+923218415358" <sip:+923218415358@207.111.170.20>;tag=as2c01215 0
To: <sip:3473293686@209.47.41.48:80>;tag=as54aaa75b
Call-ID: 12a557b129780bee0949007c2837b14c@207.111.170.20
CSeq: 102 INVITE
User-Agent: VBuzzer/1.0(v1.1.1.2-2006.03.29)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:13473293686@192.168.1.110:5050>
Content-Length: 0

---
Destroying call '12a557b129780bee0949007c2837b14c@207.111.170.20'
asterisk1*CLI>
<-- SIP read from 209.47.41.24:80:
ACK sip:13473293686@203.81.224.12:5050 SIP/2.0
Via: SIP/2.0/UDP 209.47.41.24:80;branch=z9hG4bK8ea.18877405.0
From: "+923218415358" <sip:+923218415358@207.111.170.20>;tag=as2c01215 0
Call-ID: 12a557b129780bee0949007c2837b14c@207.111.170.20
To: <sip:3473293686@209.47.41.48:80>;tag=as15a6209b
CSeq: 102 ACK
User-Agent: Phone Server 2
Content-Length: 0

--- (8 headers 0 lines)---
Destroying call '12a557b129780bee0949007c2837b14c@207.111.170.20'
asterisk1*CLI>
<-- SIP read from 209.47.41.24:80:
ACK sip:13473293686@203.81.224.12:5050 SIP/2.0
Via: SIP/2.0/UDP 209.47.41.24:80;branch=z9hG4bK8ea.18877405.0
From: "+923218415358" <sip:+923218415358@207.111.170.20>;tag=as2c01215 0
Call-ID: 12a557b129780bee0949007c2837b14c@207.111.170.20
To: <sip:3473293686@209.47.41.48:80>;tag=as54aaa75b
CSeq: 102 ACK
User-Agent: Phone Server 2
Content-Length: 0

--- (8 headers 0 lines)---
Destroying call '12a557b129780bee0949007c2837b14c@207.111.170.20'
asterisk1*CLI> exit
[root@asterisk1 ~]#
[root@asterisk1 ~]#

The +92321.... number is the number I am calling from.

Cannot help feeling that there is a bug in asterisk - any ideas?
kindest regards

AbNormal · #7 (**permalink**) May 2nd, 2006, 04:28 PM

It would appear that your asterisk PBX was still running while you captured the above.

If this was the case, then note that your softphone was sending User-Agent: VBuzzer/1.0(v1.1.1.2-2006.03.29), while your PBX was sending Phone Server 2 and/or Asterisk PBX. I'm pretty sure vbuzzer will reject those user agent decriptions.

Just my 2 cents.

syedamerali · #8 (**permalink**) May 2nd, 2006, 05:07 PM

And a very valueable two cents - but still no joy - no incoming.
the new sip dump:
12 headers, 0 lines
Reliably Transmitting (NAT) to 209.47.41.24:80:
OPTIONS sip:vbuzzer.com:80 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK495b5e0e;rport
From: "Unknown" <sip:Unknown@syedamerali.dyndns.org:5050>;tag=as0c 0dfbbf
To: <sip:vbuzzer.com:80>
Contact: <sip:Unknown@192.168.1.110:5050>
Call-ID: 01dae931062b655e02fd7f8465863d47@sye...i.dyndn s.org:5050
CSeq: 102 OPTIONS
User-Agent: vbuzzer/1.0
Max-Forwards: 70
Date: Tue, 02 May 2006 16:00:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
asterisk1*CLI>
<-- SIP read from 209.47.41.24:80:
SIP/2.0 200 Keep Alive Works
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK495b5e0e;rport=50 50;received=203.81.224.12
From: "Unknown" <sip:Unknown@syedamerali.dyndns.org:5050>;tag=as0c 0dfbbf
To: <sip:vbuzzer.com:80>;tag=0137240595b7ec88199a5a4d4 5fb6597.725d
Call-ID: 01dae931062b655e02fd7f8465863d47@sye...i.dyndn s.org:5050
CSeq: 102 OPTIONS
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.24:80 "Noisy feedback tells: pid=78610 req_src_ip=203.81.224.12 req_src_port=5050 in_uri=sip:vbuzzer.com:80 out_uri=sip:vbuzzer.com:80 via_cnt==1"

--- (9 headers 0 lines)---
Destroying call '01dae931062b655e02fd7f8465863d47@syedamerali.dynd ns.org:5050'
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 209.47.41.24:80:
REGISTER sip:vbuzzer.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK3fc538a5;rport
From: <sip:syedamerali@vbuzzer.com>;tag=as7a9007c1
To: <sip:syedamerali@vbuzzer.com>
Call-ID: 220756e1388e35335538fa471b34e5ce@vbuzzer.com
CSeq: 104 REGISTER
User-Agent: vbuzzer/1.0
Max-Forwards: 70
Authorization: Digest username="syedamerali", realm="vbuzzer.com", algorithm=MD5, uri="sip:vbuzzer.com", nonce="4457831caeef08f156293e50ae2a2bb357b3a93c", response="05c6461ebdd632854b0eaae1605ff12d", opaque=""
Expires: 120
Contact: <sip:13473293686@192.168.1.110:5050>
Event: registration
Content-Length: 0

---
asterisk1*CLI>
<-- SIP read from 209.47.41.24:80:
SIP/2.0 100 Stop retransmission
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK3fc538a5;rport=50 50;received=203.81.224.12
From: <sip:syedamerali@vbuzzer.com>;tag=as7a9007c1
To: <sip:syedamerali@vbuzzer.com>
Call-ID: 220756e1388e35335538fa471b34e5ce@vbuzzer.com
CSeq: 104 REGISTER
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.24:80 "Noisy feedback tells: pid=78613 req_src_ip=203.81.224.12 req_src_port=5050 in_uri=sip:vbuzzer.com out_uri=sip:vbuzzer.com via_cnt==1"

--- (9 headers 0 lines)---
asterisk1*CLI>
<-- SIP read from 209.47.41.24:80:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK3fc538a5;rport=50 50;received=203.81.224.12
From: <sip:syedamerali@vbuzzer.com>;tag=as7a9007c1
To: <sip:syedamerali@vbuzzer.com>;tag=0137240595b7ec88 199a5a4d45fb6597.b909
Call-ID: 220756e1388e35335538fa471b34e5ce@vbuzzer.com
CSeq: 104 REGISTER
Contact: <sip:13473293686@203.81.224.12:5050>;expires=120
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.24:80 "Noisy feedback tells: pid=78613 req_src_ip=203.81.224.12 req_src_port=5050 in_uri=sip:vbuzzer.com out_uri=sip:vbuzzer.com via_cnt==1"

--- (10 headers 0 lines)---
Scheduling destruction of call '220756e1388e35335538fa471b34e5ce@vbuzzer.com' in 32000 ms
asterisk1*CLI>

kindest regards

chandave · #9 (**permalink**) May 2nd, 2006, 07:24 PM

With Asterisk v1.2.x, make sure you are using externhost instead of externip. externhost will lookup the IP address associate with the hostname you provide on a periodic basis. externip will only lookup the IP address of the hostname during evaluation of the sip.conf.
Your Asterisk box is sending the wrong contact information to Vbuzzer. It is sending your Asterisk box's private IP address instead of your Public IP address. Do you have syedamerali.dyndns.org added into your /etc/hosts as 192.168.1.110? If so, then that's breaking Asterisk's ability to query for the Public IP address associated with syedamerali.dyndns.org.
When a SIP INVITE does come into your Asterisk box, your Asterisk box is rejecting it because no audio codec is common to both of you. The SIP URI=> sip:+923218415358@207.111.170.20 is offer G.711u and you only support G.729a.

See ya...

d.c.

gremln007 · #10 (**permalink**) May 2nd, 2006, 08:18 PM

Quote:

Originally Posted by chandave

With Asterisk v1.2.x, make sure you are using externhost instead of externip. externhost will lookup the IP address associate with the hostname you provide on a periodic basis. externip will only lookup the IP address of the hostname during evaluation of the sip.conf.
Your Asterisk box is sending the wrong contact information to Vbuzzer. It is sending your Asterisk box's private IP address instead of your Public IP address. Do you have syedamerali.dyndns.org added into your /etc/hosts as 192.168.1.110? If so, then that's breaking Asterisk's ability to query for the Public IP address associated with syedamerali.dyndns.org.
When a SIP INVITE does come into your Asterisk box, your Asterisk box is rejecting it because no audio codec is common to both of you. The SIP URI=> sip:+923218415358@207.111.170.20 is offer G.711u and you only support G.729a.

See ya...

d.c.

On #1, does that mean we no longer need to use the sip_reload script you created?

Thank you!