[Asterisk-Users] Broadvoice + Videosupport=yes - Fails!

Tue Mar 1 16:37:19 MST 2005

Here we go:

Sip.conf attached and includes relevant configs in general + broadvoice
+ 1 extension.

This is with video enabled and the config in which broadvoice fails.
Again, changing to videosupport=no in general section and everything
works fine.

Call log with failed broadvoice call attached.

I really appreciate the help

Shadow

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, March 01, 2005 2:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!

Shadow Roldan wrote:
> Ok I've updated to CVS-v1-0-03/01/05-13 and unfortunately the problems

> exist in the same manner as before.
> 
> I also tried the videosupport=yes in general and videosupport=no in 
> broadvoice to no avail.

OK, then something else is going on. Please post the relevant portions
of your sip.conf, along with a "sip debug" trace of a call attempt from
the EyeBeam.
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-------------- next part --------------
[general]
videosupport=yes
disallow=all                        
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h261
port=5060                          
context=default           
maxexpirey=180                      
defaultexpirey=160                  
canreinvite=no                      
srvlookup=yes                       
dtmfmode=rfc2833                     ; preferred is "rfc2833" - out of band
nat=no                              ; enable nat

register => MyNumber at sip.broadvoice.com:MySecret:MyNumber at sip.broadvoice.com/115

[sip.broadvoice.com]
videosupport=no
type=peer
host = sip.broadvoice.com
fromdomain = sip.broadvoice.com
fromuser=4152942073
secret=76qv9yyxqq
context=from-broadvoice
canreinvite = no
dtmfmode = inband
insecure=very
permit=147.135.8.128/32
qualify=yes
nat=no

[115]
type=friend              
username=115             
nat=yes
secret=ZeroG123         
context=inside
videosupport=yes
disallow=all                        
allow=ulaw
allow=alaw
allow=h263
allow=h261
dtmfmode=inband
qualify=1000             
host=dynamic             
canreinvite=no   
-------------- next part --------------
sipper*CLI> sip debug peer sip.broadvoice.com
SIP Debugging Enabled for IP: 147.135.8.128:5060
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:147.135.8.128 SIP/2.0
Via: SIP/2.0/UDP 66.92.49.28:5060;branch=z9hG4bK6b7498d1
From: "asterisk" <sip:asterisk at 66.92.49.28>;tag=as7a72e3ab
To: <sip:147.135.8.128>
Contact: <sip:asterisk at 66.92.49.28>
Call-ID: 503e367c1277cec80719bf400fb27dbb at 66.92.49.28
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 01 Mar 2005 23:34:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 147.135.8.128:5060
sipper*CLI>

Sip read:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 66.92.49.28:5060;branch=z9hG4bK6b7498d1
From: "asterisk" <sip:asterisk at 66.92.49.28>;tag=as7a72e3ab
To: <sip:147.135.8.128>;tag=SD304ld99-
Call-ID: 503e367c1277cec80719bf400fb27dbb at 66.92.49.28
CSeq: 102 OPTIONS
Content-Length: 0

7 headers, 0 lines
Destroying call '503e367c1277cec80719bf400fb27dbb at 66.92.49.28'
    -- Executing Dial("SIP/115-7bd9", "SIP/14155127771 at sip.broadvoice.com|30") in new stack
We're at 66.92.49.28 port 16740
Video is at 66.92.49.28 port 18672
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x80000 (h263)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:14155127771 at sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 66.92.49.28:5060;branch=z9hG4bK1ab151a6
From: "Shadow Roldan" <sip:4152942073 at sip.broadvoice.com>;tag=as6717b3c0
To: <sip:14155127771 at sip.broadvoice.com>
Contact: <sip:4152942073 at 66.92.49.28>
Call-ID: 72d8b677496274c6088f5e7160d0d2b5 at sip.broadvoice.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 01 Mar 2005 23:34:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
ontent-Length: 255

v=0
o=root 22296 22296 IN IP4 66.92.49.28
s=session
c=IN IP4 66.92.49.28
t=0 0
m=audio 16740 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
m=video 18672 RTP/AVP 34
a=rtpmap:34 H263/90000
 (no NAT) to 147.135.8.128:5060
    -- Called 14155127771 at sip.broadvoice.com
sipper*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.92.49.28:5060;branch=z9hG4bK1ab151a6
From: "Shadow Roldan" <sip:4152942073 at sip.broadvoice.com>;tag=as6717b3c0
To: <sip:14155127771 at sip.broadvoice.com>
Call-ID: 72d8b677496274c6088f5e7160d0d2b5 at sip.broadvoice.com
CSeq: 102 INVITE

6 headers, 0 lines
sipper*CLI>

Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.92.49.28:5060;branch=z9hG4bK1ab151a6
From: "Shadow Roldan" <sip:4152942073 at sip.broadvoice.com>;tag=as6717b3c0
To: <sip:14155127771 at sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
Call-ID: 72d8b677496274c6088f5e7160d0d2b5 at sip.broadvoice.com
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: 100rel,timer
Contact: <sip:14155127771 at 147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
Remote-Party-ID: <sip:14155127771 at 147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;screen=yes;party=called;privacy=off;id-type=subscriber
Content-Length: 0

11 headers, 0 lines
    -- SIP/sip.broadvoice.com-3102 is ringing
sipper*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.92.49.28:5060;branch=z9hG4bK1ab151a6
From: "Shadow Roldan" <sip:4152942073 at sip.broadvoice.com>;tag=as6717b3c0
To: <sip:14155127771 at sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
Call-ID: 72d8b677496274c6088f5e7160d0d2b5 at sip.broadvoice.com
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: 100rel,timer
Accept: application/sdp,application/dtmf
Contact: <sip:14155127771 at 147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
Content-Type: application/sdp
Content-Length: 179

v=0
o=BroadWorks 2887472 1 IN IP4 147.135.8.128
s=-
c=IN IP4 192.168.8.4
t=0 0
m=audio 12844 RTP/AVP 0
c=IN IP4 147.135.8.128
m=video 0 RTP/AVP 34
a=rtpmap:34 H263/90000

12 headers, 9 lines
Found RTP audio format 0
Found video format unknown
Peer audio RTP is at port 192.168.8.4:12844
Found description format H263
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x80000 (h263), combined - 0x80004 (ulaw|h263)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:14155127771 at 147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
set_destination: Parsing <sip:14155127771 at 147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp> for address/port to send to
set_destination: set destination to 147.135.8.128, port 5060
Transmitting:
ACK sip:14155127771 at sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 66.92.49.28:5060;branch=z9hG4bK620438f8
From: "Shadow Roldan" <sip:4152942073 at sip.broadvoice.com>;tag=as6717b3c0
To: <sip:14155127771 at sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
Contact: <sip:4152942073 at 66.92.49.28>
Call-ID: 72d8b677496274c6088f5e7160d0d2b5 at sip.broadvoice.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 147.135.8.128:5060
    -- SIP/sip.broadvoice.com-3102 answered SIP/115-7bd9
    -- Attempting native bridge of SIP/115-7bd9 and SIP/sip.broadvoice.com-3102
Sip read:
BYE sip:4152942073 at 66.92.49.28 SIP/2.0
Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK229cr4001g11a9oqo2c0.1sr
From: <sip:14155127771 at sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
To: "Shadow Roldan" <sip:4152942073 at sip.broadvoice.com>;tag=as6717b3c0
Call-ID: 72d8b677496274c6088f5e7160d0d2b5 at sip.broadvoice.com
CSeq: 809260878 BYE
Max-Forwards: 69
Content-Length: 0

8 headers, 0 lines
Sending to 147.135.8.128 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
ia: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK229cr4001g11a9oqo2c0.1sr
From: <sip:14155127771 at sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
To: "Shadow Roldan" <sip:4152942073 at sip.broadvoice.com>;tag=as6717b3c0
Call-ID: 72d8b677496274c6088f5e7160d0d2b5 at sip.broadvoice.com
CSeq: 809260878 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4152942073 at 66.92.49.28>
Content-Length: 0

 to 147.135.8.128:5060
  == Spawn extension (inside, 14155127771, 1) exited non-zero on 'SIP/115-7bd9'
Destroying call '72d8b677496274c6088f5e7160d0d2b5 at sip.broadvoice.com'
    -- Saved useragent "Sipura/SPA1000-2.0.9(GWc)" for peer 1154
    -- Saved useragent "Sipura/SPA2000-2.0.7(f)" for peer 154