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Shadow Roldan

2005-Mar-01 12:40 UTC

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

Hi All

First time poster, long time reader. 

I just ran into something really bizarre. I've enabled videosupport and
have been testing sip calls with Xten Eyebeam software, it works
(mostly)

However, when I have 

Videosupport=yes

In the [general] section of my sip.conf, broadvoice calls fail w/
"We're
sorry your call cannot be completed at this time"


So... I've commented it out and tried adding videosupport=yes to
specific extensions, now video doesn't work as eyebeam reports "remote
user does not support video" but broadvoice works.

Bizarre

I'm running CVS v1-0-02/15/05

Any ideas?


_____________________________________________________________________
 
Shadow Roldan
IT Manager
Zero G Software, Inc.
 
tel: +1.415.512.7771 x 306
fax: +1.415.723.7244
mailto:Shadow.Roldan@ZeroG.com
www.ZeroG.com
 
The leading provider of multiplatform software deployment solutions.
_____________________________________________________________________

Kevin P. Fleming

2005-Mar-01 12:47 UTC

head link

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

Shadow Roldan wrote:
> So... I've commented it out and tried adding videosupport=yes to
> specific extensions, now video doesn't work as eyebeam reports
"remote
> user does not support video" but broadvoice works.
Yes, this is because when the EyeBeam sends the first INVITE to 
Asterisk, Asterisk does not yet know who the caller is so the general 
section settings apply. With 'videosupport' off, Asterisk will reject 
the INVITE because it requests a video session.

You have two choices:

- go back to videosupport=yes in the general section, and put 
videosupport=no in your BroadVoice section (and hope it works)

- update to the latest CVS STABLE (or the 1.0.6 release), where an 
attempt has been made to cure this problem (not for this reason, but it 
will still help)

When you report problems and you are using a CVS pull, you should 
_always_ update to the latest CVS release before reporting, in case the 
problem has already been fixed.

Shadow Roldan

2005-Mar-01 14:03 UTC

head link

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

Thanks for the tips I'll report back on my findings.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, March 01, 2005 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!

Shadow Roldan wrote:
> So... I've commented it out and tried adding videosupport=yes to 
> specific extensions, now video doesn't work as eyebeam reports
"remote
> user does not support video" but broadvoice works.
Yes, this is because when the EyeBeam sends the first INVITE to
Asterisk, Asterisk does not yet know who the caller is so the general
section settings apply. With 'videosupport' off, Asterisk will reject
the INVITE because it requests a video session.

You have two choices:

- go back to videosupport=yes in the general section, and put
videosupport=no in your BroadVoice section (and hope it works)

- update to the latest CVS STABLE (or the 1.0.6 release), where an
attempt has been made to cure this problem (not for this reason, but it
will still help)

When you report problems and you are using a CVS pull, you should
_always_ update to the latest CVS release before reporting, in case the
problem has already been fixed.
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Shadow Roldan

2005-Mar-01 15:04 UTC

head link

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

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.

Shadow

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Shadow
Roldan
Sent: Tuesday, March 01, 2005 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!

Thanks for the tips I'll report back on my findings.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, March 01, 2005 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!

Shadow Roldan wrote:
> So... I've commented it out and tried adding videosupport=yes to 
> specific extensions, now video doesn't work as eyebeam reports
"remote
> user does not support video" but broadvoice works.
Yes, this is because when the EyeBeam sends the first INVITE to
Asterisk, Asterisk does not yet know who the caller is so the general
section settings apply. With 'videosupport' off, Asterisk will reject
the INVITE because it requests a video session.

You have two choices:

- go back to videosupport=yes in the general section, and put
videosupport=no in your BroadVoice section (and hope it works)

- update to the latest CVS STABLE (or the 1.0.6 release), where an
attempt has been made to cure this problem (not for this reason, but it
will still help)

When you report problems and you are using a CVS pull, you should
_always_ update to the latest CVS release before reporting, in case the
problem has already been fixed.
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Shadow Roldan

2005-Mar-01 16:37 UTC

head link

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

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@lists.digium.com
[mailto:asterisk-users-bounces@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.
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- 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@sip.broadvoice.com:MySecret:MyNumber@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@66.92.49.28>;tag=as7a72e3ab
To: <sip:147.135.8.128>
Contact: <sip:asterisk@66.92.49.28>
Call-ID: 503e367c1277cec80719bf400fb27dbb@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@66.92.49.28>;tag=as7a72e3ab
To: <sip:147.135.8.128>;tag=SD304ld99-
Call-ID: 503e367c1277cec80719bf400fb27dbb@66.92.49.28
CSeq: 102 OPTIONS
Content-Length: 0


7 headers, 0 lines
Destroying call '503e367c1277cec80719bf400fb27dbb@66.92.49.28'
    -- Executing Dial("SIP/115-7bd9",
"SIP/14155127771@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@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 66.92.49.28:5060;branch=z9hG4bK1ab151a6
From: "Shadow Roldan"
<sip:4152942073@sip.broadvoice.com>;tag=as6717b3c0
To: <sip:14155127771@sip.broadvoice.com>
Contact: <sip:4152942073@66.92.49.28>
Call-ID: 72d8b677496274c6088f5e7160d0d2b5@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@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@sip.broadvoice.com>;tag=as6717b3c0
To: <sip:14155127771@sip.broadvoice.com>
Call-ID: 72d8b677496274c6088f5e7160d0d2b5@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@sip.broadvoice.com>;tag=as6717b3c0
To:
<sip:14155127771@sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
Call-ID: 72d8b677496274c6088f5e7160d0d2b5@sip.broadvoice.com
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: 100rel,timer
Contact:
<sip:14155127771@147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
Remote-Party-ID:
<sip:14155127771@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@sip.broadvoice.com>;tag=as6717b3c0
To:
<sip:14155127771@sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
Call-ID: 72d8b677496274c6088f5e7160d0d2b5@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@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@147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
set_destination: Parsing
<sip:14155127771@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@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 66.92.49.28:5060;branch=z9hG4bK620438f8
From: "Shadow Roldan"
<sip:4152942073@sip.broadvoice.com>;tag=as6717b3c0
To:
<sip:14155127771@sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
Contact: <sip:4152942073@66.92.49.28>
Call-ID: 72d8b677496274c6088f5e7160d0d2b5@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@66.92.49.28 SIP/2.0
Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK229cr4001g11a9oqo2c0.1sr
From:
<sip:14155127771@sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
To: "Shadow Roldan"
<sip:4152942073@sip.broadvoice.com>;tag=as6717b3c0
Call-ID: 72d8b677496274c6088f5e7160d0d2b5@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@sip.broadvoice.com>;tag=SD5g9s399-1592780352-1109720086772
To: "Shadow Roldan"
<sip:4152942073@sip.broadvoice.com>;tag=as6717b3c0
Call-ID: 72d8b677496274c6088f5e7160d0d2b5@sip.broadvoice.com
CSeq: 809260878 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4152942073@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@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

Shadow Roldan

2005-Mar-01 17:00 UTC

head link

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

Yes I'm an idiot and gave ya'll my passwords

They've been changed so don't go getting any ideas

:)

Thanks Patrick for pointing out my stupidity 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Shadow
Roldan
Sent: Tuesday, March 01, 2005 3:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!

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@lists.digium.com
[mailto:asterisk-users-bounces@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.
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Nicolas FOURNIL

2005-Mar-02 09:30 UTC

head link

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

Same problems as you...

Eyebeam is not really fine in video... We have find some nasty bugs in it
(PC freeze, codecs issues...) and no feedback from Xten after sending back
reports (tcpdump and long descriptions). I think that EyeBeam works fine
with... eyebeam. The software seems to be beta because of each version of
Eyebeam I've download has differents bugs.

Try with our hard-videophone ( ;-) ), Asterisk video features works. Perhaps
a small problem in Intra Frame request (I've posted it in feature request
without success). We will work on it ASAP.

Nicolas
http://www.call.fr


-----Message d'origine-----
De : asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]De la part de Shadow
Roldan
Envoye : mardi 1 mars 2005 20:40
A : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!


Hi All

First time poster, long time reader.

I just ran into something really bizarre. I've enabled videosupport and
have been testing sip calls with Xten Eyebeam software, it works
(mostly)

However, when I have

Videosupport=yes

In the [general] section of my sip.conf, broadvoice calls fail w/
"We're
sorry your call cannot be completed at this time"


So... I've commented it out and tried adding videosupport=yes to
specific extensions, now video doesn't work as eyebeam reports "remote
user does not support video" but broadvoice works.

Bizarre

I'm running CVS v1-0-02/15/05

Any ideas?


_____________________________________________________________________

Shadow Roldan
IT Manager
Zero G Software, Inc.

tel: +1.415.512.7771 x 306
fax: +1.415.723.7244
mailto:Shadow.Roldan@ZeroG.com
www.ZeroG.com

The leading provider of multiplatform software deployment solutions.
_____________________________________________________________________

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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