Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****Solved*****!

[MF Hulber]

I concur.  I rebuilt today and now I seem to be able to dial out.
MARK.
Chris Nibeck wrote:
thank you everyone!
It does not seen that it was configuration problems at all.
It appears it was the CVS that I was using from yesterday.
I decided to start over, downloaded the latest CVS, recompiled, and 
voila!  * started working

Indeed even a Cisco ATA that was never working before started working!
Thanks to everyone!
Chris

On Mar 9, 2005, at 10:28 AM, Chris Nibeck wrote:
there are two of us with the same problem so I will answer for me.  
Yes I tried the below instructions.

The current thinking by multiple people is * never tries 
authenticating so removing the FQDN will force * to go to the related 
section named by either a phone number or a non Fully Qualified 
Domain Name.

But I still don't have it working so who knows.
Anyone that wishes to call me via BV my number is 8475100139 and it 
is up.

Chris
On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote:
Have you tried this:
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
Zanzamar Majere wrote:
Thank you for the response.   I still have the errors mentioned 
below, sip response and Failed to authenticate on INVITE

[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
host=sip.broadvoice.com
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no
Does anyone else have any other suggestions?

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***SOLVED*** [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****Solved*****!

[Chris Nibeck]

thank you everyone!
It does not seen that it was configuration problems at all.
It appears it was the CVS that I was using from yesterday.
I decided to start over, downloaded the latest CVS, recompiled, and 
voila!  * started working

Indeed even a Cisco ATA that was never working before started working!
Thanks to everyone!
Chris

On Mar 9, 2005, at 10:28 AM, Chris Nibeck wrote:
there are two of us with the same problem so I will answer for me.  
Yes I tried the below instructions.

The current thinking by multiple people is * never tries 
authenticating so removing the FQDN will force * to go to the related 
section named by either a phone number or a non Fully Qualified Domain 
Name.

But I still don't have it working so who knows.
Anyone that wishes to call me via BV my number is 8475100139 and it is 
up.

Chris
On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote:
Have you tried this:
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
Zanzamar Majere wrote:
Thank you for the response.   I still have the errors mentioned 
below, sip response and Failed to authenticate on INVITE

[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
host=sip.broadvoice.com
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no
Does anyone else have any other suggestions?

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RE: [Asterisk-Users] Broadvoice latest changes and still not working-An

[Marios Andreou]

The problem that you have it was the one that I stabled 
across the very first time tried to setup BV.
The 404 not found that you are getting is because there is 
no such phone number [EMAIL PROTECTED]
But there is a [EMAIL PROTECTED].
 
This is like saying [EMAIL PROTECTED] (you 
are going to get a 404)
 
The chi worked because it was a test server (beta/debug) 
that I read somewhere in this list.
 
So the fix for you will be to change the 

 
host=proxy.lax.broadvoice.com
to 
host=sip.broadvoice.com
 
 
Now if you are getting better responses from lax then 
change your host file to
147.135.8.128  sip.broadvoice.com
 
This is because sip.broadvoice.com resolves to 
proxy.dca.broadvoice.com.
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  JoeSent: Wednesday, March 09, 2005 1:41 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Broadvoice 
  latest changes and still not working-An
  
  
   
  I’ve tried everything with the * box after this 
  weekend.  I have read every 
  document on the problems people are having with them after this weekend as 
  well, but none of them address my problem.
   
  I checked my settings in my sips which I have below as 
  well,  
  
   
  I have changed the host file a few times,  but this was new to me and I never had 
  modified it before.  I have and 
  the same results happened.
   
  I have always used the CHI proxy until this past 
  weekend.
   
  I get a 404 not found when the invite goes out.   
   
  Below is my debug for broadvoice,  which I think tells the whole 
  story,  but for the life of me, I 
  can not figure out where the 404 is coming from.
   
  I have listed my sip file below as 
  well.
   
  Inbound calls work and I am 
  registered.
   
  Before we go into the debug,  I get this message when I reload my 
  configs files.
  
  Mar  9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response: 
  Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling 
  reregistration in 1933000 ms)
   
   
  Below is the debug:
   
      
  -- Executing Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]|30") in 
  new stack
  We're at outsideIPaddress port 
  14842
  Answering with preferred capability 0x4 
  (ulaw)
  12 headers, 8 lines
  Reliably Transmitting:
  INVITE sip:[EMAIL PROTECTED] 
  SIP/2.0
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  
  Contact: 
  
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 INVITE
  User-Agent: Asterisk PBX
  Date: Wed, 09 Mar 2005 18:15:18 
  GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 
  REFER
  Content-Type: 
  application/sdp
  Content-Length: 164
   
  v=0
  o=root 17647 17647 IN IP4 
  outsideIPaddress
  s=session
  c=IN IP4 outsideIPaddress
  t=0 0
  m=audio 14842 RTP/AVP 0
  a=rtpmap:0 PCMU/8000
  a=silenceSupp:off - - - -
   (no NAT) 
  to 147.135.8.128:5060
      
  -- Called [EMAIL PROTECTED]
  asterisk1*CLI>
   
  Sip read:
  SIP/2.0 100 Trying
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 INVITE
   
   
  6 headers, 0 lines
  asterisk1*CLI>
   
  Sip read:
  SIP/2.0 404 
  Not Found
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  ;tag=SD4ou5a99-
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 
  INVITE
  Content-Length: 0
   
   
  7 headers, 0 lines
      
  -- Got SIP response 404 "Not Found" back from 
  147.135.8.128
  Transmitting:
  ACK sip:[EMAIL PROTECTED] 
  SIP/2.0
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  ;tag=SD4ou5a99-
  Contact: 
  
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 ACK
  User-Agent: Asterisk PBX
  Content-Length: 0
   
   (no NAT) 
  to 147.135.8.128:5060
      
  -- SIP/sip.broadvoice.com-2a2c is 
  circuit-busy
    == 
  Everyone is busy/congested at this time
      
  -- Executing Busy("OSS/dsp", "") in new 
  stack
  Destroying call 
  '[EMAIL PROTECTED]'
  asterisk1*CLI> hangup
    == Spawn 
  extension (default, 509, 102) exited non-zero on 
  'OSS/dsp'
   << 
  Hangup on console >>
   
   
  [sip.broadvoice.com]
  type=peer
  host=proxy.lax.broadvoice.com
  fromdomain=sip.broadvoice.com
  fromuser= BB
  username= BB
  ;authuser= BB
  secret= secret
  context=sip
  nat=no
  insecure=very
  dtmfmode=inband
   
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Re: [Asterisk-Users] Broadvoice latest changes and still not working-An

[Scott Wolfe]

Just wondering. How are you getting this debug. I 
am having problems to and I cant seem to track it down.

  - Original Message - 
  From: 
  Joe 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, March 09, 2005 10:41 
  AM
  Subject: [Asterisk-Users] Broadvoice 
  latest changes and still not working-An
  
  
   
  I’ve tried everything with the * box after this 
  weekend.  I have read every 
  document on the problems people are having with them after this weekend as 
  well, but none of them address my problem.
   
  I checked my settings in my sips which I have below as 
  well,  
  
   
  I have changed the host file a few times,  but this was new to me and I never had 
  modified it before.  I have and 
  the same results happened.
   
  I have always used the CHI proxy until this past 
  weekend.
   
  I get a 404 not found when the invite goes out.   
   
  Below is my debug for broadvoice,  which I think tells the whole 
  story,  but for the life of me, I 
  can not figure out where the 404 is coming from.
   
  I have listed my sip file below as 
  well.
   
  Inbound calls work and I am 
  registered.
   
  Before we go into the debug,  I get this message when I reload my 
  configs files.
  
  Mar  9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response: 
  Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling 
  reregistration in 1933000 ms)
   
   
  Below is the debug:
   
      
  -- Executing Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]|30") in 
  new stack
  We're at outsideIPaddress port 
  14842
  Answering with preferred capability 0x4 
  (ulaw)
  12 headers, 8 lines
  Reliably Transmitting:
  INVITE sip:[EMAIL PROTECTED] 
  SIP/2.0
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  
  Contact: 
  
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 INVITE
  User-Agent: Asterisk PBX
  Date: Wed, 09 Mar 2005 18:15:18 
  GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 
  REFER
  Content-Type: 
  application/sdp
  Content-Length: 164
   
  v=0
  o=root 17647 17647 IN IP4 
  outsideIPaddress
  s=session
  c=IN IP4 outsideIPaddress
  t=0 0
  m=audio 14842 RTP/AVP 0
  a=rtpmap:0 PCMU/8000
  a=silenceSupp:off - - - -
   (no NAT) 
  to 147.135.8.128:5060
      
  -- Called [EMAIL PROTECTED]
  asterisk1*CLI>
   
  Sip read:
  SIP/2.0 100 Trying
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 INVITE
   
   
  6 headers, 0 lines
  asterisk1*CLI>
   
  Sip read:
  SIP/2.0 404 
  Not Found
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  ;tag=SD4ou5a99-
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 
  INVITE
  Content-Length: 0
   
   
  7 headers, 0 lines
      
  -- Got SIP response 404 "Not Found" back from 
  147.135.8.128
  Transmitting:
  ACK sip:[EMAIL PROTECTED] 
  SIP/2.0
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  ;tag=SD4ou5a99-
  Contact: 
  
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 ACK
  User-Agent: Asterisk PBX
  Content-Length: 0
   
   (no NAT) 
  to 147.135.8.128:5060
      
  -- SIP/sip.broadvoice.com-2a2c is 
  circuit-busy
    == 
  Everyone is busy/congested at this time
      
  -- Executing Busy("OSS/dsp", "") in new 
  stack
  Destroying call 
  '[EMAIL PROTECTED]'
  asterisk1*CLI> hangup
    == Spawn 
  extension (default, 509, 102) exited non-zero on 
  'OSS/dsp'
   << 
  Hangup on console >>
   
   
  [sip.broadvoice.com]
  type=peer
  host=proxy.lax.broadvoice.com
  fromdomain=sip.broadvoice.com
  fromuser= BB
  username= BB
  ;authuser= BB
  secret= secret
  context=sip
  nat=no
  insecure=very
  dtmfmode=inband
   
  
  

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[Asterisk-Users] Broadvoice latest changes and still not working-An

[Joe]

 

I’ve tried everything with the * box after this weekend.  I have read every document on the
problems people are having with them after this weekend as well, but none of
them address my problem.

 

I checked my settings in my sips which I have below as well,  

 

I have changed the host file a few times,  but this was new to me and I never had
modified it before.  I have and the
same results happened.

 

I have always used the CHI proxy until this past weekend.

 

I get a 404 not found when the invite goes out.   

 

Below is my debug for broadvoice, 
which I think tells the whole story,  but for the life of me, I can not figure
out where the 404 is coming from.

 

I have listed my sip file below as well.

 

Inbound calls work and I am registered.

 

Before we go into the debug, 
I get this message when I reload my configs files.



Mar  9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response:
Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling
reregistration in 1933000 ms)



 

 

Below is the debug:

 

    -- Executing
Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]|30") in
new stack

We're at outsideIPaddress port 14842

Answering with preferred capability 0x4 (ulaw)

12 headers, 8 lines

Reliably Transmitting:

INVITE sip:[EMAIL PROTECTED] SIP/2.0

Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc

From: "asterisk" ;tag=as6ed673e9

To: 

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Wed, 09 Mar
 2005 18:15:18 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Content-Type: application/sdp

Content-Length: 164

 

v=0

o=root 17647 17647 IN IP4 outsideIPaddress

s=session

c=IN IP4 outsideIPaddress

t=0 0

m=audio 14842 RTP/AVP 0

a=rtpmap:0 PCMU/8000

a=silenceSupp:off - - - -

 (no NAT) to
147.135.8.128:5060

    -- Called
[EMAIL PROTECTED]

asterisk1*CLI>

 

Sip read:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc

From: "asterisk" ;tag=as6ed673e9

To: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 INVITE

 

 

6 headers, 0 lines

asterisk1*CLI>

 

Sip read:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc

From: "asterisk"
;tag=as6ed673e9

To: ;tag=SD4ou5a99-

Call-ID: [EMAIL PROTECTED]

CSeq: 102 INVITE

Content-Length: 0

 

 

7 headers, 0 lines

    -- Got SIP
response 404 "Not Found" back from 147.135.8.128

Transmitting:

ACK sip:[EMAIL PROTECTED] SIP/2.0

Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc

From: "asterisk" ;tag=as6ed673e9

To: ;tag=SD4ou5a99-

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 ACK

User-Agent: Asterisk PBX

Content-Length: 0

 

 (no NAT) to
147.135.8.128:5060

    --
SIP/sip.broadvoice.com-2a2c is circuit-busy

  == Everyone is
busy/congested at this time

    -- Executing
Busy("OSS/dsp", "") in new stack

Destroying call '[EMAIL PROTECTED]'

asterisk1*CLI> hangup

  == Spawn extension
(default, 509, 102) exited non-zero on 'OSS/dsp'

 << Hangup on console
>>

 

 

[sip.broadvoice.com]

type=peer

host=proxy.lax.broadvoice.com

fromdomain=sip.broadvoice.com

fromuser= BB

username= BB

;authuser= BB

secret= secret

context=sip

nat=no

insecure=very

dtmfmode=inband

 






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Fwd: Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****SOLVED****

[Zanzamar Majere]

This configuration solved my problem.  I could have sworn I tried this
 before. I guess not.  I did not need to apply the patch.  Also, I am using a
 regular Registration setup in my sip.conf not broadvoice's funky one...

The only thing I can surmise is that order of the variables matters.

This is what worked for me:


[PP]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=PP
secret=XX
username=PP
insecure=very
context=sip
authname=PP
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no


Thank you

On Wednesday 09 March 2005 08:23 am, Mike Matthews wrote:
> Have you tried this:
>
> http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
>
> Zanzamar Majere wrote:
> >Thank you for the response.   I still have the errors mentioned below, sip
> >response and Failed to authenticate on INVITE
> >
> >[PP]
> >type=peer
> >username=PP
> >fromuser=PP
> >authuser=PP
> >fromdomain=sip.broadvoice.com
> >secret=XX
> >host=sip.broadvoice.com
> >dtmfmode=inband
> >insecure=very
> >context=sip
> >qualify=yes
> >disallow=all
> >allow=ulaw
> >allow=gsm
> >;Disable canreinvite if you are behind a NAT
> >;canreinvite=no
> >nat=no
> >
> >Does anyone else have any other suggestions?
>
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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Chris Nibeck]

there are two of us with the same problem so I will answer for me.  Yes 
I tried the below instructions.

The current thinking by multiple people is * never tries authenticating 
so removing the FQDN will force * to go to the related section named by 
either a phone number or a non Fully Qualified Domain Name.

But I still don't have it working so who knows.
Anyone that wishes to call me via BV my number is 8475100139 and it is 
up.

Chris
On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote:
Have you tried this:
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
Zanzamar Majere wrote:
Thank you for the response.   I still have the errors mentioned 
below, sip response and Failed to authenticate on INVITE

[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
host=sip.broadvoice.com
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no
Does anyone else have any other suggestions?

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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Chris Nibeck]

Thanks MF,
Yes that was me that sent my PW :-)   It is changed now.
Same error...
Mar  9 10:12:46 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
to authenticate on INVITE to '"Chris Nibeck"  
;tag=as0cefa74c'

Sip.conf...
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=x
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
extensions.conf...
exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice  
for 30 seconds
exten => _8X.,2, congestion() ; No answer, nothing
exten => _8X., 102, busy() ;

On Mar 9, 2005, at 7:56 AM, MF Hulber wrote:
Try changing the extension from Broadvoice1 to the actual phone number  
(and don't send your secret in a public email or maybe that's Chris'):

[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139

Zanzamar Majere wrote:
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this  
response
(on call out).  Any suggestions?  I don't think it is a problem with  
the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however.  So I know at least that
works.


-- Got SIP response 400 "Bad request" back from 147.135.0.128
Mar  9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to '"PP"
;tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
First off...  please cancel previous amplification request.  I have   
implemented your ideas with the same errored result.

I am not sure that we are not making it thru authentication.  From  
my  digging and comparing packet dumps comparing the soft phone to  
asterisk  they have identical transactions through  the ACK reply  
(the last one  on the debug below).  The softphone seems to be  
authenticated after the  ACK.  I am a newbie to debugging this  
stuff. I just want to get it  working.

Thanks everyone in advance for your help.  I am certainly very very   
happy to try anything.

Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=DELETED
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial  
Broadvoice  for 30 seconds
exten => _8X.,2, congestion() ; No answer, nothing
exten => _8X., 102, busy() ;

End result...
Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response:  
Failed  to authenticate on INVITE to '"6050"   
;tag=as545ccba3'

SIP debug...
-- Executing Dial("SIP/6050-132b",   
"SIP/[EMAIL PROTECTED]|30") in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
 (no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 ;tag=7e2776985d5a0891o0
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest   
username="6050",realm="asterisk",nonce="42d82e9b",uri="sip:  
[EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a 
10c 129dd4fb5f97ec47"
Contact: 6050 
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 ;tag=7e2776985d5a0891o0
To: ;tag=as2f065f18
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,

Re: [Asterisk-Users] Broadvoice latest changes and still not working-An Additional Server

[Chris Nibeck]

Jerry-
Thank you
I accidently sent my password on the LISTSERV last night so I just 
changed (pasted) the new one in.

Still the same problem...
Mar  9 09:51:13 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed 
to authenticate on INVITE to '"Chris Nibeck" 
;tag=as4b70f2e7'

Incoming works fine still.  Anyone can call me at that number.  Please 
do.

It is a free call from another BV account.
Chris
On Mar 9, 2005, at 7:42 AM, Jerry Geis wrote:
CHris,
I had the exact same problem with the exact same error.
My password was entered incorrectly in context section.
The register line had the correct password. That is why you get
incoming calls. and not outgoing.
Jerry
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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Mike Matthews]

Have you tried this:
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
Zanzamar Majere wrote:
Thank you for the response.   I still have the errors mentioned below, sip 
response and Failed to authenticate on INVITE

[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
host=sip.broadvoice.com
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no
Does anyone else have any other suggestions?
 

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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Zanzamar Majere]

Thank you for the response.   I still have the errors mentioned below, sip 
response and Failed to authenticate on INVITE

[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
host=sip.broadvoice.com
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no

Does anyone else have any other suggestions?


On Wednesday 09 March 2005 06:56 am, MF Hulber wrote:
> Try changing the extension from Broadvoice1 to the actual phone number
> (and don't send your secret in a public email or maybe that's Chris'):
>
> [*8475100139*]
> type=peer
> ;user=phone
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> fromuser=8475100139
> secret=XXX
> username=8475100139
>
> Zanzamar Majere wrote:
> >I have made all the changes to sip.conf for my broadvoice peer
> >friend(and I have tried it as peer) and I am still seeing this response
> >(on call out).  Any suggestions?  I don't think it is a problem with the
> >phones themselves authenticating, as Asterisk takes care of all the
> >authentication from my understanding.
> >
> >Free world does work for calling out however.  So I know at least that
> >works.
> >
> >
> >
> >-- Got SIP response 400 "Bad request" back from 147.135.0.128
> >Mar  9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
> >to authenticate on INVITE to '"PP"
> >;tag=as5b80cade'
> >
> >On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
> >>First off...  please cancel previous amplification request.  I have
> >>implemented your ideas with the same errored result.
> >>
> >>I am not sure that we are not making it thru authentication.  From my
> >>digging and comparing packet dumps comparing the soft phone to asterisk
> >>they have identical transactions through  the ACK reply (the last one
> >>on the debug below).  The softphone seems to be authenticated after the
> >>ACK.  I am a newbie to debugging this stuff. I just want to get it
> >>working.
> >>
> >>Thanks everyone in advance for your help.  I am certainly very very
> >>happy to try anything.
> >>
> >>Based on Luki's suggestions I...
> >>
> >>Changed sip.conf...
> >>
> >>[broadvoice1]
> >>type=peer
> >>;user=phone
> >>host=sip.broadvoice.com
> >>fromdomain=sip.broadvoice.com
> >>fromuser=8475100139
> >>secret=DELETED
> >>username=8475100139
> >>insecure=very
> >>context=default
> >>authname=8475100139
> >>dtmfmode=inband
> >>dtmf=inband
> >>;Disable canreinvite if you are behind a NAT
> >>canreinvite=no
> >>nat=no
> >>
> >>Changed extensions.conf...
> >>
> >>exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
> >>for 30 seconds
> >>exten => _8X.,2, congestion() ; No answer, nothing
> >>exten => _8X., 102, busy() ;
> >>
> >>End result...
> >>
> >>Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
> >>to authenticate on INVITE to '"6050"
> >>;tag=as545ccba3'
> >>
> >>
> >>SIP debug...
> >>
> >> -- Executing Dial("SIP/6050-132b",
> >>"SIP/[EMAIL PROTECTED]|30") in new stack
> >>We're at xxx.xxx.xxx.xxx port 18212
> >>Answering with capability 2
> >>Answering with capability 4
> >>Answering with capability 8
> >>12 headers, 10 lines
> >>Reliably Transmitting:
> >>INVITE sip:[EMAIL PROTECTED] SIP/2.0
> >>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> >>From: "6050" ;tag=as545ccba3
> >>To: 
> >>Contact: 
> >>Call-ID: [EMAIL PROTECTED]
> >>CSeq: 102 INVITE
> >>User-Agent: Asterisk PBX
> >>Date: Wed, 09 Mar 2005 07:30:41 GMT
> >>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >>Content-Type: application/sdp
> >>Content-Length: 205
> >>
> >>v=0
> >>o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
> >>s=session
> >>c=IN IP4 xxx.xxx.xxx.xxx
> >>t=0 0
> >>m=audio 18212 RTP/AVP 3 0 8
> >>a=rtpmap:3 GSM/8000
> >>a=rtpmap:0 PCMU/8000
> >>a=rtpmap:8 PCMA/8000
> >>a=silenceSupp:off - - - -
> >>  (no NAT) to 147.135.8.128:5060
> >> -- Called [EMAIL PROTECTED]
> >>com*CLI>
> >>
> >>Sip read:
> >>INVITE sip:[EMAIL PROTECTED] SIP/2.0
> >>Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
> >>From: 6050 ;tag=7e2776985d5a0891o0
> >>To: 
> >>Call-ID: [EMAIL PROTECTED]
> >>CSeq: 102 INVITE
> >>Max-Forwards: 70
> >>Proxy-Authorization: Digest
> >>username="6050",realm="asterisk",nonce="42d82e9b",uri="sip:
> >>[EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c
> >>129dd4fb5f97ec47"
> >>Contact: 6050 
> >>Expires: 240
> >>User-Agent: Sipura/SPA3000-2.0.10(GWf)
> >>Content-Length: 241
> >>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> >>Supported: x-sipura
> >>Content-Type: application/sdp
> >>
> >>v=0
> >>o=- 1138990026 1138990026 IN IP4 64.4.192.110
> >>s=-
> >>c=IN IP4 64.4.192.110
> >>t=0 0
> >>m=audio 16388 RTP/AVP 0 100 101
> >>a=rtpmap:0 PCMU/8000
> >>a=rtpmap:100 NSE/8000
> >>a=rtpmap:101 telephone-event/8000
> >>a=fmtp:101 0-15
> >>a=ptime:30
> >>a=sendrecv
> >>
> >>15 header

Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[MF Hulber]

Try changing the extension from Broadvoice1 to the actual phone number 
(and don't send your secret in a public email or maybe that's Chris'):

[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139

Zanzamar Majere wrote:
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out).  Any suggestions?  I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.  

Free world does work for calling out however.  So I know at least that
works.

-- Got SIP response 400 "Bad request" back from 147.135.0.128
Mar  9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to '"PP"
;tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
 

First off...  please cancel previous amplification request.  I have  
implemented your ideas with the same errored result.

I am not sure that we are not making it thru authentication.  From my  
digging and comparing packet dumps comparing the soft phone to asterisk  
they have identical transactions through  the ACK reply (the last one  
on the debug below).  The softphone seems to be authenticated after the  
ACK.  I am a newbie to debugging this stuff. I just want to get it  
working.

Thanks everyone in advance for your help.  I am certainly very very  
happy to try anything.

Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=DELETED
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice  
for 30 seconds
exten => _8X.,2, congestion() ; No answer, nothing
exten => _8X., 102, busy() ;

End result...
Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
to authenticate on INVITE to '"6050"  
;tag=as545ccba3'

SIP debug...
-- Executing Dial("SIP/6050-132b",  
"SIP/[EMAIL PROTECTED]|30") in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
 (no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 ;tag=7e2776985d5a0891o0
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest  
username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: 
[EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c 
129dd4fb5f97ec47"
Contact: 6050 
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 ;tag=7e2776985d5a0891o0
To: ;tag=as2f065f18
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0
 to 64.4.192.110:5060
com*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI>
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: ;tag=SD38rq699-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST  
realm="BroadWorks",algorithm=MD5,nonce="1110353299563"
Content-Length: 0

8 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: ;tag=SD38rq699-
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq:

[Asterisk-Users] Broadvoice latest changes and still not working-An Additional Server

[Jerry Geis]

CHris,
I had the exact same problem with the exact same error.
My password was entered incorrectly in context section.
The register line had the correct password. That is why you get
incoming calls. and not outgoing.
Jerry
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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Zanzamar Majere]

I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out).  Any suggestions?  I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.  

Free world does work for calling out however.  So I know at least that
works.



-- Got SIP response 400 "Bad request" back from 147.135.0.128
Mar  9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to '"PP"
;tag=as5b80cade'

On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
> First off...  please cancel previous amplification request.  I have  
> implemented your ideas with the same errored result.
> 
> I am not sure that we are not making it thru authentication.  From my  
> digging and comparing packet dumps comparing the soft phone to asterisk  
> they have identical transactions through  the ACK reply (the last one  
> on the debug below).  The softphone seems to be authenticated after the  
> ACK.  I am a newbie to debugging this stuff. I just want to get it  
> working.
> 
> Thanks everyone in advance for your help.  I am certainly very very  
> happy to try anything.
> 
> Based on Luki's suggestions I...
> 
> Changed sip.conf...
> 
> [broadvoice1]
> type=peer
> ;user=phone
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> fromuser=8475100139
> secret=zjh018g8f8
> username=8475100139
> insecure=very
> context=default
> authname=8475100139
> dtmfmode=inband
> dtmf=inband
> ;Disable canreinvite if you are behind a NAT
> canreinvite=no
> nat=no
> 
> Changed extensions.conf...
> 
> exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice  
> for 30 seconds
> exten => _8X.,2, congestion() ; No answer, nothing
> exten => _8X., 102, busy() ;
> 
> End result...
> 
> Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
> to authenticate on INVITE to '"6050"  
> ;tag=as545ccba3'
> 
> 
> SIP debug...
> 
>  -- Executing Dial("SIP/6050-132b",  
> "SIP/[EMAIL PROTECTED]|30") in new stack
> We're at xxx.xxx.xxx.xxx port 18212
> Answering with capability 2
> Answering with capability 4
> Answering with capability 8
> 12 headers, 10 lines
> Reliably Transmitting:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" ;tag=as545ccba3
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Wed, 09 Mar 2005 07:30:41 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 205
> 
> v=0
> o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=audio 18212 RTP/AVP 3 0 8
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
>   (no NAT) to 147.135.8.128:5060
>  -- Called [EMAIL PROTECTED]
> com*CLI>
> 
> Sip read:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
> From: 6050 ;tag=7e2776985d5a0891o0
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest  
> username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: 
> [EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c 
> 129dd4fb5f97ec47"
> Contact: 6050 
> Expires: 240
> User-Agent: Sipura/SPA3000-2.0.10(GWf)
> Content-Length: 241
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
> 
> v=0
> o=- 1138990026 1138990026 IN IP4 64.4.192.110
> s=-
> c=IN IP4 64.4.192.110
> t=0 0
> m=audio 16388 RTP/AVP 0 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
> 
> 15 headers, 12 lines
> Ignoring this request
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
> From: 6050 ;tag=7e2776985d5a0891o0
> To: ;tag=as2f065f18
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: 
> Content-Length: 0
> 
> 
>   to 64.4.192.110:5060
> com*CLI>
> 
> Sip read:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" ;tag=as545ccba3
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> 
> 
> 6 headers, 0 lines
> com*CLI>
> 
> Sip read:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" ;tag=as545ccba3
> To: ;tag=SD38rq699-
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> WWW-Authenticate: DIGEST  
> realm="BroadWorks",algorithm=MD5,nonce="1110353299563"
> Content-Length: 0
> 
> 
> 8 headers, 0 lines
> Transmitting:
> ACK sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" ;tag=as545ccba3
> To: ;tag=SD38rq699-
> Con

Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Chris Nibeck]

First off...  please cancel previous amplification request.  I have  
implemented your ideas with the same errored result.

I am not sure that we are not making it thru authentication.  From my  
digging and comparing packet dumps comparing the soft phone to asterisk  
they have identical transactions through  the ACK reply (the last one  
on the debug below).  The softphone seems to be authenticated after the  
ACK.  I am a newbie to debugging this stuff. I just want to get it  
working.

Thanks everyone in advance for your help.  I am certainly very very  
happy to try anything.

Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=zjh018g8f8
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice  
for 30 seconds
exten => _8X.,2, congestion() ; No answer, nothing
exten => _8X., 102, busy() ;

End result...
Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
to authenticate on INVITE to '"6050"  
;tag=as545ccba3'

SIP debug...
-- Executing Dial("SIP/6050-132b",  
"SIP/[EMAIL PROTECTED]|30") in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
 (no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 ;tag=7e2776985d5a0891o0
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest  
username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: 
[EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c 
129dd4fb5f97ec47"
Contact: 6050 
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 ;tag=7e2776985d5a0891o0
To: ;tag=as2f065f18
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0
 to 64.4.192.110:5060
com*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI>
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: ;tag=SD38rq699-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST  
realm="BroadWorks",algorithm=MD5,nonce="1110353299563"
Content-Length: 0

8 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: ;tag=SD38rq699-
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
 (no NAT) to 147.135.8.128:5060
Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
to authenticate on INVITE to '"6050"  
;tag=as545ccba3'


On Mar 9, 2005, at 12:08 AM, Luki wrote:
Chris,
first of all, if your server has been up for 200 days, I suggest you
update the kernel -- you don't say if it's Linux, but chances are that
yes... and there have been some security bugs patched recently.
That aside. I'm not sure, but it's possible that since you are using a
valid host name ('sip.broadvoice.com') in your dial statement, perhaps
* tried to talk to it directly and does not consider the section in
sip.conf. Just a guess. You will notice from the the sip debug output
that * does not even try to authenticate, as if it didn't know about
the user/secret.
I use the BV number as the section name, so the dial statement
essentially looks like: Dial([EMAIL PROTECTED])
Try changing

Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Chris Nibeck]

First, thanks for your help.
I have been changing these to different values but not getting it. 
Could you further amplify your statement...

Try changing yours to say "broadvoice" and then the corresponding
section in sip.conf.
Thanks!
Chris
On Mar 9, 2005, at 12:08 AM, Luki wrote:
Chris,
first of all, if your server has been up for 200 days, I suggest you
update the kernel -- you don't say if it's Linux, but chances are that
yes... and there have been some security bugs patched recently.
That aside. I'm not sure, but it's possible that since you are using a
valid host name ('sip.broadvoice.com') in your dial statement, perhaps
* tried to talk to it directly and does not consider the section in
sip.conf. Just a guess. You will notice from the the sip debug output
that * does not even try to authenticate, as if it didn't know about
the user/secret.
I use the BV number as the section name, so the dial statement
essentially looks like: Dial([EMAIL PROTECTED])
Try changing yours to say "broadvoice" and then the corresponding
section in sip.conf. I'm using the DCA server, and didn't have an
issue at all when they introduced INVITE authentication on the
weekend. This is how my section looks like:
[360350]
type=peer
dtmfmode=inband
username=360350
fromuser=360350
secret=XX
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
canreinvite=no
nat=no
insecure=very
context=incoming
outgoinglimit=2
In /etc/hosts I have:
147.135.0.128   sip.broadvoice.com
It's the proxy.dca.broadvoice.com server. Hope this helps...
--Luki
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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Luki]

Chris,

first of all, if your server has been up for 200 days, I suggest you
update the kernel -- you don't say if it's Linux, but chances are that
yes... and there have been some security bugs patched recently.

That aside. I'm not sure, but it's possible that since you are using a
valid host name ('sip.broadvoice.com') in your dial statement, perhaps
* tried to talk to it directly and does not consider the section in
sip.conf. Just a guess. You will notice from the the sip debug output
that * does not even try to authenticate, as if it didn't know about
the user/secret.

I use the BV number as the section name, so the dial statement
essentially looks like: Dial([EMAIL PROTECTED])

Try changing yours to say "broadvoice" and then the corresponding
section in sip.conf. I'm using the DCA server, and didn't have an
issue at all when they introduced INVITE authentication on the
weekend. This is how my section looks like:

[360350]
type=peer
dtmfmode=inband
username=360350
fromuser=360350
secret=XX
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
canreinvite=no
nat=no
insecure=very
context=incoming
outgoinglimit=2

In /etc/hosts I have:
147.135.0.128   sip.broadvoice.com

It's the proxy.dca.broadvoice.com server. Hope this helps...

--Luki
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[Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Chris Nibeck]

I have been going crazy with this also since Sat.
Our server was working perfectly with BV but will now not place calls 
to BV.

Incoming from BV works fine.
I felt sad rebooting it today, it had been running for almost 200 days!
Here is my error message from the console...
Notice I am running today's CVS
Connected to Asterisk CVS-03/08/05-14:32:39 currently running on com 
(pid = 1624)
-- Executing Dial("SIP/6050-5bc9", 
"SIP/[EMAIL PROTECTED]|30") in new stack
-- Called [EMAIL PROTECTED]
Mar  8 23:11:55 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed 
to authenticate on INVITE to '"6050" 
;tag=as20911f6e'

I have tried many versions of sip.conf, here is the current...
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=blah
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
I have tried the different proxies proxy.dca.broadvoice.com, lax, mia, 
and was originally using chi when the system worked.

BV told me Mon that chi is considered a test server that should not be 
used for production, it is expected to go up and down.

My hosts file points to one of the working ones.
I verified my account through a softphone. It works fine to BV.
There is something wrong with the authentication.
Here is the SIP debug...
-- Executing Dial("SIP/6050-019c", 
"SIP/[EMAIL PROTECTED]|30") in new stack
We're at xxx.xxx.xxx.xxx port 16776
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c
From: "6050" ;tag=as292b9469
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 05:35:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 3501 3501 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 16776 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
 (no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c
From: "6050" ;tag=as292b9469
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI>
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c
From: "6050" ;tag=as292b9469
To: ;tag=SD38ad399-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST 
realm="BroadWorks",algorithm=MD5,nonce="1110346372627"
Content-Length: 0

8 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c
From: "6050" ;tag=as292b9469
To: ;tag=SD38ad399-
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
 (no NAT) to 147.135.8.128:5060
Mar  8 23:35:15 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed 
to authenticate on INVITE to '"6050" 
;tag=as292b9469'

TIA
Chris
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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Rich Adamson]

Since he can "call into the box", he is registering with BV. Otherwise he
would not be able to call in. So, his outgoing calls are messed up one
way or another. How about doing a "sip debug" while placing a call via
BV and post the results?


> Can you call anywhere or is this problem just with broadvoice? Is there
> any type of firewall like a netscreen or iptables configured in your
> setup, which may be blocking outbound UDP? Do you have a packet capture
> of the traffic that is leaving your network for broadvoice? You should
> try to inspect the signaling exchange with their proxy.
> 
> -Original Message-
> From: Jerry Geis [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, March 08, 2005 2:53 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Broadvoice latest changes and still not
> working
> 
> I tried removing the permit and that made no difference.
> 
> I can still call in to the box but no calls out.
> 
> Jerry
> 
> --
> 
> Yes it is working just fine for me with the same sip.conf that you have.
> ??
> Except the "permit=sip.broadvoice.com"
> You can see my config at
> http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
> 
> Also what is your extensions.conf ?
> 
> -Original Message-
> From: asterisk-users-bounces at lists.digium.com
> <http://lists.digium.com/mailman/listinfo/asterisk-users>
> [mailto:asterisk-users-bounces at lists.digium.com
> <http://lists.digium.com/mailman/listinfo/asterisk-users>] On Behalf Of
> James Taylor
> Sent: Tuesday, March 08, 2005 2:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not
> working
> 
> Does anybody have Broadvoice outbound working?
> 
> On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis  <http://lists.digium.com/mailman/listinfo/asterisk-users>>  
> wrote:
> 
> >/ Here is my configs. from a previous post...
> />/
> />/ Jerry
> />/
> />/ --
> />
> 
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---End of Original Message-


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Re: [Asterisk-Users] Broadvoice latest changes and still not working

[John Millican]

On Tuesday March 08 2005 2:58 pm, James Taylor wrote:
> Ok, used your sip.conf inbound works.  Outbound gets:
> "SIP/2.0 604 Does not exist anywhere"
>
> Any ideas?
> James
>
> On Tue, 8 Mar 2005 14:18:11 -0500, Marios Andreou <[EMAIL PROTECTED]>
>
> wrote:
> > Yes it is working just fine for me with the same sip.conf that you have.
> > ??
> > Except the "permit=sip.broadvoice.com"
> > You can see my config at
> > http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
> >
> > Also what is your extensions.conf ?
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of James
> > Taylor
> > Sent: Tuesday, March 08, 2005 2:15 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not
> > working
> >
> > Does anybody have Broadvoice outbound working?
> >
> > On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis <[EMAIL PROTECTED]>
> >

I have a same sip.conf and out and in are working well.  I have 
sip.broadvoice.com mapped to proxy.lax.broadvoice.com in my hosts file.  this 
is nice for me as i can use sip.broadvoice.com in all .conf and if i need to 
change the proxy i do so in the hosts file. I do not use ${EXTEN:1} in my 
outbound dial and i always dial 10 digits.
John Millican
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[Asterisk-Users] Broadvoice latest changes and still not working - solved HEYYY

[Jerry Geis]

Looks like I had mistyped that long password.
so the register statement was correct but the context was NOT correct 
off by 1 character in the middle.

I never susspected the password as it worked before the weekend changes. 
Thought it was OK.

Thanks to everyone..
Jerry
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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Marios Andreou]

What is the output of the "show version"  ?

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 3:19 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not working

Marios,

You are correct. Every phone on the system (8 or so) has a context=smvoice-sip
in the config for every phone.

This config was all working uptil last saturday when broadvoice made the 
changes.
It has not worked for outgoing calls since then. Incoming is still working.

This is one of my extensions.

[405]
type=friend
dtmfmode=rfc2833
username=405
secret=SECRET
disallow=all
allow=ulaw
allow=alaw
host=dynamic
context=smvoice-sip
insecure=very
callerid="Fred Smith" <405>


Jerry

--

Hmm!!

OK I'm missing some variables (like what is ${OPERATOR}) but that is OK because 
incoming you said is working.

But what I don't know is the context for the SIP/ 's
It should be [smvoice-sip] so they can dial the _91X extension.
OR their context should "include => smvoice-sip"

 

-Original Message-
From: asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>
[mailto:asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>] On Behalf Of Jerry
Geis
Sent: Tuesday, March 08, 2005 2:45 PM
To: asterisk-users at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>
Subject: [Asterisk-Users] Broadvoice latest changes and still not working

My extension.conf is below.

Jerry


-



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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Giudice, Salvatore]

Can you call anywhere or is this problem just with broadvoice? Is there
any type of firewall like a netscreen or iptables configured in your
setup, which may be blocking outbound UDP? Do you have a packet capture
of the traffic that is leaving your network for broadvoice? You should
try to inspect the signaling exchange with their proxy.

-Original Message-
From: Jerry Geis [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, March 08, 2005 2:53 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not
working

I tried removing the permit and that made no difference.

I can still call in to the box but no calls out.

Jerry

--

Yes it is working just fine for me with the same sip.conf that you have.
??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html

Also what is your extensions.conf ?

-Original Message-
From: asterisk-users-bounces at lists.digium.com
<http://lists.digium.com/mailman/listinfo/asterisk-users>
[mailto:asterisk-users-bounces at lists.digium.com
<http://lists.digium.com/mailman/listinfo/asterisk-users>] On Behalf Of
James Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not
working

Does anybody have Broadvoice outbound working?

On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>>  
wrote:

>/ Here is my configs. from a previous post...
/>/
/>/ Jerry
/>/
/>/ --
/>

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[Asterisk-Users] Broadvoice latest changes and still not working

[Jerry Geis]

Marios,
You are correct. Every phone on the system (8 or so) has a context=smvoice-sip
in the config for every phone.
This config was all working uptil last saturday when broadvoice made the 
changes.
It has not worked for outgoing calls since then. Incoming is still working.
This is one of my extensions.
[405]
type=friend
dtmfmode=rfc2833
username=405
secret=SECRET
disallow=all
allow=ulaw
allow=alaw
host=dynamic
context=smvoice-sip
insecure=very
callerid="Fred Smith" <405>
Jerry
--
Hmm!!
OK I'm missing some variables (like what is ${OPERATOR}) but that is OK because 
incoming you said is working.
But what I don't know is the context for the SIP/ 's
It should be [smvoice-sip] so they can dial the _91X extension.
OR their context should "include => smvoice-sip"

-Original Message-
From: asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users> 
[mailto:asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 2:45 PM
To: asterisk-users at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>
Subject: [Asterisk-Users] Broadvoice latest changes and still not working
My extension.conf is below.
Jerry
-

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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Marios Andreou]

Hmm!!

OK I'm missing some variables (like what is ${OPERATOR}) but that is OK because 
incoming you said is working.

But what I don't know is the context for the SIP/ 's
It should be [smvoice-sip] so they can dial the _91X extension.
OR their context should "include => smvoice-sip"

 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 2:45 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not working

My extension.conf is below.

Jerry


-


[default]
exten => s,1,Wait,1 ; Wait before speaking
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,2 ; Set Digit Timeout to 
5 seconds
exten => s,4,ResponseTimeout,20 ; Set Response Timeout 
to 10 seconds
exten => s,5,ChanIsAvail(SIP/201&SIP/202&SIP/203&SIP/204&SIP/205&SIP/206)
exten => s,6,Cut(thechannel=AVAILCHAN,,1)
exten => s,7,Dial(${thechannel},${DIAL_TIMEOUT},tT)
exten => s,8,background(SM_ATTENDANT)
exten => s,9,noop("background done")
exten => s,10,SetVar(SMVOICE_EXTEN=${OPERATOR})
exten => s,11,Goto(default,operator,1)

exten => PHONE,1,Goto(default,s,1)


[smvoice-sip]
exten => 11,1,playback(demo-congrats)
exten => 11,2,hangup

exten => 
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

exten => 
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)






Yes it is working just fine for me with the same sip.conf that you have. ??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html

Also what is your extensions.conf ?

-Original Message-
From: asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>
[mailto:asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>] On Behalf Of James
Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working

Does anybody have Broadvoice outbound working?

On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>>

wrote:

>/ Here is my configs. from a previous post...
/>/
/>/ Jerry
/>/
/>/ --
/>/
/>/ ; Broadvoice
/>/ register => PHONE at sip.broadvoice.com  
/>/ <http://lists.digium.com/mailman/listinfo/asterisk-users>:SECRET:PHONE  
/>/ at sip.broadvoice.com  
/>/ <http://lists.digium.com/mailman/listinfo/asterisk-users>/PHONE
/>/
/>/ [Broadvoice]
/>/ type=friend
/>/ username=PHONE
/>/ authuser=PHONE
/>/ fromuser=PHONE
/>/ secret=secret
/>/ host=sip.broadvoice.com
/>/ port=5060
/>/ context=default
/>/ fromdomain=sip.broadvoice.com
/>/ canreinvite=no
/>/ dtmfmode=inband
/>/ insecure=very
/>/ permit=sip.broadvoice.com
/>/ qualify=yes
/>/ disallow=all
/>/ allow=ulaw
/>/ maxexpirey=180
/>/ defaultexpirey=160
/>/ videosupport=no
/>/
/>/
/>/ exten =>  
/>/ _9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
/>/
/>/ exten =>  
/>/ 
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
/>/
/>/ ___
/>/ Asterisk-Users mailing list
/>/ Asterisk-Users at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>
/>/ http://lists.digium.com/mailman/listinfo/asterisk-users
/>/ To UNSUBSCRIBE or update options visit:
/>/http://lists.digium.com/mailman/listinfo/asterisk-users
/>/
/


-- 
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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Re: [Asterisk-Users] Broadvoice latest changes and still not working

[Roger Hanson]

My broadvoice works perfectly.  I am using a standard registration 
string,  however.  Not the funky one broadvoice says to use.  I can make 
outbound and receive inbound calls over broadvoice.

I'm using AMP also.
register=phonenumber:[EMAIL PROTECTED]
sip.conf:
[952XX]
username=952XX
type=friend
secret=password
regexten=952XXX
insecure=very
host=sip.broadvoice.com
fromuser=952XX
fromdomain=sip.broadvoice.com
dtmfmode=inband
context=from-pstn
canreinvite=yes
authuser=952
[sdfdsf]
- Original Message - 
From: "Marios Andreou" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 

Sent: Tuesday, March 08, 2005 1:18 PM
Subject: RE: [Asterisk-Users] Broadvoice latest changes and still not 
working


Yes it is working just fine for me with the same sip.conf that you 
have. ??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html

Also what is your extensions.conf ?
-Original Message-
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[Asterisk-Users] Broadvoice latest changes and still not working

[Jerry Geis]

I tried removing the permit and that made no difference.
I can still call in to the box but no calls out.
Jerry
--
Yes it is working just fine for me with the same sip.conf that you have. ??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
Also what is your extensions.conf ?
-Original Message-
From: asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users> 
[mailto:asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>] On Behalf Of James Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>>  
wrote:

/ Here is my configs. from a previous post...
/>/
/>/ Jerry
/>/
/>/ --
/>
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[Asterisk-Users] Broadvoice latest changes and still not working

[Jerry Geis]

My extension.conf is below.
Jerry
-
[default]
exten => s,1,Wait,1 ; Wait before speaking
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,2 ; Set Digit Timeout to 
5 seconds
exten => s,4,ResponseTimeout,20 ; Set Response Timeout 
to 10 seconds
exten => s,5,ChanIsAvail(SIP/201&SIP/202&SIP/203&SIP/204&SIP/205&SIP/206)
exten => s,6,Cut(thechannel=AVAILCHAN,,1)
exten => s,7,Dial(${thechannel},${DIAL_TIMEOUT},tT)
exten => s,8,background(SM_ATTENDANT)
exten => s,9,noop("background done")
exten => s,10,SetVar(SMVOICE_EXTEN=${OPERATOR})
exten => s,11,Goto(default,operator,1)
exten => PHONE,1,Goto(default,s,1)
[smvoice-sip]
exten => 11,1,playback(demo-congrats)
exten => 11,2,hangup
exten => 
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
exten => 
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)


Yes it is working just fine for me with the same sip.conf that you have. ??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
Also what is your extensions.conf ?
-Original Message-
From: asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users> 
[mailto:asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>] On Behalf Of James Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>>  
wrote:

/ Here is my configs. from a previous post...
/>/
/>/ Jerry
/>/
/>/ --
/>/
/>/ ; Broadvoice
/>/ register => PHONE at sip.broadvoice.com  
/>/ <http://lists.digium.com/mailman/listinfo/asterisk-users>:SECRET:PHONE  
/>/ at sip.broadvoice.com  
/>/ <http://lists.digium.com/mailman/listinfo/asterisk-users>/PHONE
/>/
/>/ [Broadvoice]
/>/ type=friend
/>/ username=PHONE
/>/ authuser=PHONE
/>/ fromuser=PHONE
/>/ secret=secret
/>/ host=sip.broadvoice.com
/>/ port=5060
/>/ context=default
/>/ fromdomain=sip.broadvoice.com
/>/ canreinvite=no
/>/ dtmfmode=inband
/>/ insecure=very
/>/ permit=sip.broadvoice.com
/>/ qualify=yes
/>/ disallow=all
/>/ allow=ulaw
/>/ maxexpirey=180
/>/ defaultexpirey=160
/>/ videosupport=no
/>/
/>/
/>/ exten =>  
/>/ _9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
/>/
/>/ exten =>  
/>/ _91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
/>/
/>/ ___
/>/ Asterisk-Users mailing list
/>/ Asterisk-Users at lists.digium.com <http://lists.digium.com/mailman/listinfo/asterisk-users>
/>/ http://lists.digium.com/mailman/listinfo/asterisk-users
/>/ To UNSUBSCRIBE or update options visit:
/>/http://lists.digium.com/mailman/listinfo/asterisk-users
/>/
/

--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
___
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Re: [Asterisk-Users] Broadvoice latest changes and still not working

[James Taylor]

Ok, used your sip.conf inbound works.  Outbound gets:
"SIP/2.0 604 Does not exist anywhere"
Any ideas?
James
On Tue, 8 Mar 2005 14:18:11 -0500, Marios Andreou <[EMAIL PROTECTED]>  
wrote:

Yes it is working just fine for me with the same sip.conf that you have.  
??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html

Also what is your extensions.conf ?
-Original Message-
From: [EMAIL PROTECTED]  
[mailto:[EMAIL PROTECTED] On Behalf Of James  
Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not  
working

Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis <[EMAIL PROTECTED]>
wrote:
Here is my configs. from a previous post...
Jerry
--
; Broadvoice
register => PHONE at sip.broadvoice.com
<http://lists.digium.com/mailman/listinfo/asterisk-users>:SECRET:PHONE
at sip.broadvoice.com
<http://lists.digium.com/mailman/listinfo/asterisk-users>/PHONE
[Broadvoice]
type=friend
username=PHONE
authuser=PHONE
fromuser=PHONE
secret=secret
host=sip.broadvoice.com
port=5060
context=default
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
permit=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw
maxexpirey=180
defaultexpirey=160
videosupport=no
exten =>
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
exten =>
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
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--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Marios Andreou]

Yes it is working just fine for me with the same sip.conf that you have. ??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html

Also what is your extensions.conf ?

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working

Does anybody have Broadvoice outbound working?

On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis <[EMAIL PROTECTED]>  
wrote:

> Here is my configs. from a previous post...
>
> Jerry
>
> --
>
> ; Broadvoice
> register => PHONE at sip.broadvoice.com  
> <http://lists.digium.com/mailman/listinfo/asterisk-users>:SECRET:PHONE  
> at sip.broadvoice.com  
> <http://lists.digium.com/mailman/listinfo/asterisk-users>/PHONE
>
> [Broadvoice]
> type=friend
> username=PHONE
> authuser=PHONE
> fromuser=PHONE
> secret=secret
> host=sip.broadvoice.com
> port=5060
> context=default
> fromdomain=sip.broadvoice.com
> canreinvite=no
> dtmfmode=inband
> insecure=very
> permit=sip.broadvoice.com
> qualify=yes
> disallow=all
> allow=ulaw
> maxexpirey=180
> defaultexpirey=160
> videosupport=no
>
>
> exten =>  
> _9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
>
> exten =>  
> _91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
>
> ___
> 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
>



-- 
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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Re: [Asterisk-Users] Broadvoice latest changes and still not working

[James Taylor]

Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis <[EMAIL PROTECTED]>  
wrote:

Here is my configs. from a previous post...
Jerry
--
; Broadvoice
register => PHONE at sip.broadvoice.com  
:SECRET:PHONE  
at sip.broadvoice.com  
/PHONE

[Broadvoice]
type=friend
username=PHONE
authuser=PHONE
fromuser=PHONE
secret=secret
host=sip.broadvoice.com
port=5060
context=default
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
permit=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw
maxexpirey=180
defaultexpirey=160
videosupport=no
exten =>  
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

exten =>  
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

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--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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[Asterisk-Users] Broadvoice latest changes and still not working

[Jerry Geis]

Here is my configs. from a previous post...
Jerry
--
; Broadvoice
register => PHONE at sip.broadvoice.com 
:SECRET:PHONE at 
sip.broadvoice.com /PHONE
[Broadvoice]
type=friend
username=PHONE
authuser=PHONE
fromuser=PHONE
secret=secret
host=sip.broadvoice.com
port=5060
context=default
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
permit=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw
maxexpirey=180
defaultexpirey=160
videosupport=no
exten => 
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

exten => 
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Marios Andreou]

Can you can post the relevant information from your sip.conf and 
extensions.conf ?
Don't forget to hide password/phone/...





From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry 
Geis
Sent: Tuesday, March 08, 2005 12:49 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not 
working


The call is a local call so that should be fine.
This worked in the past and I tried it the other way also with the 
same error message about "Failed to authenticate on INVITE".

Thanks

Jerry



Don't you need 1 in front of the number?
Attempting call on SIP/Broadvoice/5068012

It should be "Attempting call on SIP/Broadvoice/1(area code)5068012"
Try it and see if you can place outgoing.

-Original Message-
From: asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users>
[mailto:asterisk-users-bounces at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users> ] On Behalf Of Jerry
Geis
Sent: Tuesday, March 08, 2005 9:11 AM
To: asterisk-users at lists.digium.com 
<http://lists.digium.com/mailman/listinfo/asterisk-users> 
    Subject: [Asterisk-Users] Broadvoice latest changes and still not 
working


I have added the three lines to the sip.conf file based on the latest 
changes
from broadvoice. I can receive incoming calls but cannot place any 
outgoing calls.

The error I get is:

*CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569
-- Attempting call on SIP/Broadvoice/5068012 for application 
Playback(demo-congrats) (Retry 1)
Mar  8 08:35:21 NOTICE[29290]: chan_sip.c:6814 handle_response: Failed 
to authenticate on INVITE to '"asterisk" 
http://lists.digium.com/mailman/listinfo/asterisk-users> >;tag=as1304fa68'

Any ideas on why I cannot place calls?

THanks very much.




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[Asterisk-Users] Broadvoice latest changes and still not working

[Jerry Geis]

The call is a local call so that should be fine.
This worked in the past and I tried it the other way also with the 
same error message about "Failed to authenticate on INVITE".

Thanks

Jerry



Don't you need 1 in front of the number?
Attempting call on SIP/Broadvoice/5068012

It should be "Attempting call on SIP/Broadvoice/1(area code)5068012"
Try it and see if you can place outgoing.

-Original Message-
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 9:11 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not working


I have added the three lines to the sip.conf file based on the latest 
changes
from broadvoice. I can receive incoming calls but cannot place any 
outgoing calls.

The error I get is:

*CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569
-- Attempting call on SIP/Broadvoice/5068012 for application 
Playback(demo-congrats) (Retry 1)
Mar  8 08:35:21 NOTICE[29290]: chan_sip.c:6814 handle_response: Failed 
to authenticate on INVITE to '"asterisk" 
PHONE at sip.broadvoice.com>;tag=as1304fa68'

Any ideas on why I cannot place calls?

THanks very much.




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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Marios Andreou]

Don't you need 1 in front of the number?
Attempting call on SIP/Broadvoice/5068012

It should be "Attempting call on SIP/Broadvoice/1(area code)5068012"
Try it and see if you can place outgoing.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 9:11 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not working


I have added the three lines to the sip.conf file based on the latest 
changes
from broadvoice. I can receive incoming calls but cannot place any 
outgoing calls.

The error I get is:

*CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569
-- Attempting call on SIP/Broadvoice/5068012 for application 
Playback(demo-congrats) (Retry 1)
Mar  8 08:35:21 NOTICE[29290]: chan_sip.c:6814 handle_response: Failed 
to authenticate on INVITE to '"asterisk" 
;tag=as1304fa68'

Any ideas on why I cannot place calls?

THanks very much.

Jerry

--

; Broadvoice
register => [EMAIL PROTECTED]:SECRET:[EMAIL PROTECTED]/PHONE

[Broadvoice]
type=friend
username=PHONE
authuser=PHONE
fromuser=PHONE
secret=secret
host=sip.broadvoice.com
port=5060
context=default
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
permit=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw
maxexpirey=180
defaultexpirey=160
videosupport=no


exten => 
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

exten => 
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

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[Asterisk-Users] Broadvoice latest changes and still not working

[Jerry Geis]

I have added the three lines to the sip.conf file based on the latest 
changes
from broadvoice. I can receive incoming calls but cannot place any 
outgoing calls.

The error I get is:
*CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569
   -- Attempting call on SIP/Broadvoice/5068012 for application 
Playback(demo-congrats) (Retry 1)
Mar  8 08:35:21 NOTICE[29290]: chan_sip.c:6814 handle_response: Failed 
to authenticate on INVITE to '"asterisk" 
;tag=as1304fa68'

Any ideas on why I cannot place calls?
THanks very much.
Jerry
--
; Broadvoice
register => [EMAIL PROTECTED]:SECRET:[EMAIL PROTECTED]/PHONE
[Broadvoice]
type=friend
username=PHONE
authuser=PHONE
fromuser=PHONE
secret=secret
host=sip.broadvoice.com
port=5060
context=default
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
permit=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw
maxexpirey=180
defaultexpirey=160
videosupport=no
exten => 
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

exten => 
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

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