Vbuzzer again - going nuts!!

syedamerali · #1 (**permalink**) January 13th, 2006, 09:52 AM

I have a Sipura working with vbuzzer without problems....
The vbuzzer softphone works ....
But I have been trying to get it to work with asterisk for the last two weeks and cannot .....

The following indicates that I have registered with vbuzzer (as does Asterisk info in *@home)

--- (10 headers 0 lines)---
Scheduling destruction of call '6b7dde0f4896a1b257f81d350b5c0f54@syedamerali.dynd ns.org' in 32000 ms
Destroying call '6b7dde0f4896a1b257f81d350b5c0f54@syedamerali.dynd ns.org'
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 209.47.41.48:80:
REGISTER sip:vbuzzer.com SIP/2.0
Via: SIP/2.0/UDP 203.81.196.33:5050;branch=z9hG4bK43e1f725;rport
From: <sip:syedamerali@vbuzzer.com>;tag=as07f0e11c
To: <sip:syedamerali@vbuzzer.com>
Call-ID: 6b7dde0f4896a1b257f81d350b5c0f54@sye...i.dyndn s.org
CSeq: 105 REGISTER
User-Agent: VBuzzer/1.1.0.9
Max-Forwards: 70
Authorization: Digest username="syedamerali", realm="vbuzzer.com", algorithm=MD5, uri="sip:vbuzzer.com", nonce="43c77981bd4c619d9495231973b30816f9b2e82c", response="50c09720a70c1e47b457e1cc81d41b55", opaque=""
Expires: 120
Contact: <sip:s@203.81.196.33:5050>
Event: registration
Content-Length: 0

---
asterisk1*CLI>
<-- SIP read from 209.47.41.48:80:
SIP/2.0 100 Stop retransmission
Via: SIP/2.0/UDP 203.81.196.33:5050;branch=z9hG4bK43e1f725;rport=50 50
From: <sip:syedamerali@vbuzzer.com>;tag=as07f0e11c
To: <sip:syedamerali@vbuzzer.com>
Call-ID: 6b7dde0f4896a1b257f81d350b5c0f54@sye...i.dyndn s.org
CSeq: 105 REGISTER
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.48:80 "Noisy feedback tells: pid=8642 req_src_ip=203.81.196.33 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 209.47.41.48:80:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.81.196.33:5050;branch=z9hG4bK43e1f725;rport=50 50
From: <sip:syedamerali@vbuzzer.com>;tag=as07f0e11c
To: <sip:syedamerali@vbuzzer.com>;tag=c3a7d3f7f58ee36d af11fd3295f362bc.9820
Call-ID: 6b7dde0f4896a1b257f81d350b5c0f54@sye...i.dyndn s.org
CSeq: 105 REGISTER
Contact: <sip:s@203.81.196.33:5050>;expires=120
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.48:80 "Noisy feedback tells: pid=8642 req_src_ip=203.81.196.33 req_src_port=5050 in_uri=sip:vbuzzer.com out_uri=sip:vbuzzer.com via_cnt==1"

But when I try and call ......:cry:

-- Executing SetVar("SIP/204-248b", "OUTNUM=18005558355") in new stack
-- Executing Cut("SIP/204-248b", "custom=OUT_1|:|1") in new stack
-- Executing GotoIf("SIP/204-248b", "0?16") in new stack
-- Executing Dial("SIP/204-248b", "SIP/vbuzzer/18005558355") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/204-248b", "s-CHANUNAVAIL|1") in new stack

Could somebody please help?

chandave · #2 (**permalink**) January 13th, 2006, 06:58 PM

What are the SIP INVITEs and replies look like between your system and vbuzzer? Also, did you include a "port=80" in your outgoing context definition for vbuzzer?

See ya...

d.c.

lensor · #3 (**permalink**) January 13th, 2006, 07:29 PM

Check to make sure your vBuzzer TRUNK dialplan is setup to accept the 1800 number you are calling. If not, that would cause the channel unavail/congestion message.

syedamerali · #4 (**permalink**) January 14th, 2006, 07:30 AM

The SIP communication is right on top - and indicates registration.
The dial plan is XX. - but surely it is irrelevant as the DIAL command is executing.
regards

lensor · #5 (**permalink**) January 14th, 2006, 10:29 AM

I've seen several posts about vbuzzer trunk outgoing 'peer details' requiring the following to get it to work (I had to add these to make it work):

type=friend
useragent=VBuzzer/1.0

Are you using these settings?

syedamerali · #6 (**permalink**) January 14th, 2006, 11:38 AM

the context is as under

[vbuzzer]
username=syedamerali
type=friend
secret=XXXXXX
restrictcid=yes
qualify=3000
port=80
nat=no
insecure=very
host=vbuzzer.com:80
hidecallerid=yes
fromuser=syedamerali
fromdomain=vbuzzer.com:80
dtmfmode=inband
disallow=all
context=outbound-allroutes-custom
canreinvite=no
authname=syedamerali
allow=ulaw

The useragent is defined in sip conf as:

port = 5050 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
bindport = 5050
useragent=VBuzzer/1.1.0.9
;
externip = syedamerali.dyndns.org:5050
fromdomain = syedamerali.dyndns.org:5050
localnet = 192.168.1.0/255.255.255.0
progressinband=yes

Does this help ...

--- (10 headers 0 lines)---
Transmitting (no NAT) to 209.47.41.48:80:
ACK sip:18005558355@vbuzzer.com:80 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK4994f3d5;rport
From: "device" <sip:204@syedamerali.dyndns.org:5050>;tag=as64b982 60
To: <sip:18005558355@vbuzzer.com:80>;tag=c3a7d3f7f58ee 36daf11fd3295f362bc.ad9e
Contact: <sip:204@192.168.1.110:5050>
Call-ID: 3f0003f27c5cb4c709d9b97c7b8be94e@sye...i.dyndn s.org:5050
CSeq: 102 ACK
User-Agent: VBuzzer/1.1.0.9
Max-Forwards: 70
Content-Length: 0

---
-- SIP/vbuzzer.com:80-01b3 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Macro("SIP/204-9943", "outisbusy") in new stack
-- Executing Playback("SIP/204-9943", "all-circuits-busy-now") in new stack

Thank you for the time you are devoting on my behalf ...
kindest regards

chandave · #7 (**permalink**) January 14th, 2006, 02:11 PM

Quote:

Originally Posted by syedamerali

The SIP communication is right on top - and indicates registration.
The dial plan is XX. - but surely it is irrelevant as the DIAL command is executing.
regards

What you provided was a SIP REGISTER. That only tells your VoIP Service Provider (VSP) how it can contact you.

A SIP INVITE initiates a call setup...the predecessor to a call session. At the Asterisk CLI command line issue the following commands:
set verbose 5
set debug 0
sip debug ip 209.47.41.48:80

Make the call from asterisk to a destination. Record the SIP conversation VERBATIM and post it.

See ya...

d.c.

syedamerali · #8 (**permalink**) January 14th, 2006, 02:29 PM

Chandave - here goes

--- (10 headers 0 lines)---
Scheduling destruction of call '0676830a3b4a9d4e2b31292a55859be3@syedamerali.dyn dns.org:5050' in 32000 ms
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- Executing Dial("SIP/204-2265", "SIP/18005558355@vbuzzer.com:80|999|TW") in new stack
We're at 192.168.1.110 port 11274
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to 209.47.41.48:80:
INVITE sip:18005558355@vbuzzer.com:80 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK674a8939;rport
From: "device" <sip:204@syedamerali.dyndns.org:5050>;tag=as32582a ce
To: <sip:18005558355@vbuzzer.com:80>
Contact: <sip:204@192.168.1.110:5050>
Call-ID: 6076b7b2074e5b7568d813c70f6c3c79@sye...i.dyndn s.org:5050
CSeq: 102 INVITE
User-Agent: VBuzzer/1.1.0.9
Max-Forwards: 70
Date: Sat, 14 Jan 2006 14:52:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 308

v=0
o=root 2503 2503 IN IP4 192.168.1.110
s=session
c=IN IP4 192.168.1.110
t=0 0
m=audio 11274 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=noa=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -

---
-- Called 18005558355@vbuzzer.com:80
asterisk1*CLI>
<-- SIP read from 209.47.41.48:80:
SIP/2.0 100 Stop retransmission
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK674a8939;rport=50 50;received=203.81.196.33
From: "device" <sip:204@syedamerali.dyndns.org:5050>;tag=as32582a ce
To: <sip:18005558355@vbuzzer.com:80>
Call-ID: 6076b7b2074e5b7568d813c70f6c3c79@sye...i.dyndn s.org:5050
CSeq: 102 INVITE
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.48:80 "Noisy feedback tells: pid=41600 req_src_ip=203.81.196.33 req_src_port=5050 in_uri=sip:18005558355@vbuzzer.com:80 out_uri=sip:18005558355@vbuzzer.com:80 via_cnt==1"

--- (9 headers 0 lines)---

<-- SIP read from 209.47.41.48:80:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK674a8939;rport=50 50;received=203.81.196.33
From: "device" <sip:204@syedamerali.dyndns.org:5050>;tag=as32582a ce
To: <sip:18005558355@vbuzzer.com:80>;tag=c3a7d3f7f58ee 36daf11fd3295f362bc.0ce5
Call-ID: 6076b7b2074e5b7568d813c70f6c3c79@sye...i.dyndn s.org:5050
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="syedamerali.dyndns.org", nonce="43c91160c3fe66b0099111d819186abe2f52743b"
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.48:80 "Noisy feedback tells: pid=41600 req_src_ip=203.81.196.33 req_src_port=5050 in_uri=sip:18005558355@vbuzzer.com:80 out_uri=sip:18005558355@vbuzzer.com:80 via_cnt==1"

--- (10 headers 0 lines)---
Transmitting (no NAT) to 209.47.41.48:80:
ACK sip:18005558355@vbuzzer.com:80 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK674a8939;rport
From: "device" <sip:204@syedamerali.dyndns.org:5050>;tag=as32582a ce
To: <sip:18005558355@vbuzzer.com:80>;tag=c3a7d3f7f58ee 36daf11fd3295f362bc.0ce5
Contact: <sip:204@192.168.1.110:5050>
Call-ID: 6076b7b2074e5b7568d813c70f6c3c79@sye...i.dyndn s.org:5050
CSeq: 102 ACK
User-Agent: VBuzzer/1.1.0.9
Max-Forwards: 70
Content-Length: 0

---
-- SIP/vbuzzer.com:80-e86a is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Macro("SIP/204-2265", "outisbusy") in new stack
-- Executing Playback("SIP/204-2265", "all-circuits-busy-now") in new stack

makes any sense?
kindest regards

chandave · #9 (**permalink**) January 14th, 2006, 04:35 PM

Quote:

Originally Posted by syedamerali

-- Executing Dial("SIP/204-2265", "SIP/18005558355@vbuzzer.com:80|999|TW") in new stack

The above line indicates that you have changed your configuration to use SIP/18005558355@vbuzzer.com:80 instead of SIP/vbuzzer/18005558355. Since you only have the context [vbuzzer] defined in your sip.conf, Asterisk can no longer find the authentication credentials related to your vbuzzer account. This is why your Asterisk stops after receiving the"401 Unauthorized" response from vbuzzer.

Normally, receiving a 401 or 407 (Proxy authentication required) will force Asterisk to use the realm and nonce found in the WWW-Authenticate or Proxy-Authenticate header for generating a response. In your current case, it terminates because it does not have any authentication info from you.

Go back to specifying the channel as SIP/vbuzzer/phonenum and give us another Asterisk dump if you still have problems.

See ya...

d.c.

syedamerali · #10 (**permalink**) January 15th, 2006, 03:20 AM

Chandav
when I do it the way you suggested i.e. SIP/vbuzzer/18005558355
asterisk does not issue an SIP INVITE and starts saying all circuits are busy. I have confirmed this by alternating between what you suggested and what I was doing previously. The dump is below:

asterisk1*CLI>
<-- SIP read from 209.47.41.48:80:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:5050;branch=z9hG4bK70d89620;rport=50 50;received=203.81.196.33
From: <sip:syedamerali@vbuzzer.com>;tag=as3e5a823b
To: <sip:syedamerali@vbuzzer.com>;tag=c3a7d3f7f58ee36d af11fd3295f362bc.56ea
Call-ID: 0fb2042e0560dcf744dab1e677185a68@sye...i.dyndn s.org:5050
CSeq: 104 REGISTER
Contact: <sip:204@203.81.196.33:5050>;expires=120
Server: Phone Server 1
Content-Length: 0
Warning: 392 209.47.41.48:80 "Noisy feedback tells: pid=58378 req_src_ip=203.81.196.33 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 '0fb2042e0560dcf744dab1e677185a68@syedamerali.dynd ns.org:5050' in 32000 ms
-- Executing Dial("SIP/204-ebce", "SIP/vbuzzer/18005558355|999|TW") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Macro("SIP/204-ebce", "outisbusy") in new stack
-- Executing Playback("SIP/204-ebce", "all-circuits-busy-now") in new stack
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/204-ebce", "pls-try-call-later") in new stack
-- Playing 'pls-try-call-later' (language 'en')
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/204-ebce' in macro 'outisbusy'
== Spawn extension (from-internal, 0018005558355, 2) exited non-zero on 'SIP/204-ebce'
-- Executing Macro("SIP/204-ebce", "hangupcall") in new stack
-- Executing ResetCDR("SIP/204-ebce", "w") in new stack
-- Executing NoCDR("SIP/204-ebce", "") in new stack
-- Executing Wait("SIP/204-ebce", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/204-ebce' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/204-ebce'

I am including the context vbuzzer for refrence:

[vbuzzer]
username=syedamerali
type=friend
secret=XXXXXX
restrictcid=yes
qualify=3000
port=80
nat=yes
insecure=very
host=vbuzzer.com:80
hidecallerid=yes
fromuser=syedamerali
fromdomain=vbuzzer.com:80
dtmfmode=inband
disallow=all
context=outbound-allroutes-custom
canreinvite=no
authname=syedamerali
allow=ulaw

makes any sense ?
I am getting incoming calls from vbuzzer without too much difficulty ....
kindest regards