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Jon Miron

2008-Mar-16 20:10 UTC

[asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18

Hi all,

I just upgraded to Asterisk 1.4.18 a few days ago and I don't use
Broadvoice TOO often, however I have a Vermont number with them and so
my mother in law calls it to talk to my wife once in a while, so
that's why it took me so long to notice it wasn't working.  Anyway,
when she calls she gets a busy signal (as I've tested when calling it
from my cell).

When I enable debugging I get the following:

SIP Debugging Enabled for IP: 147.135.0.128
net-xero*CLI>
<--- SIP read from UDP://147.135.0.128:5060 --->
INVITE sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0
Call-ID: 320190-32 at 147.135.0.128
CSeq: 1 INVITE
From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=prtu
To: "<my name>"<sip:s@<servers IP>>
Via: SIP/2.0/UDP 147.135.0.128:5060
Contact: <sip:<my cell #>@147.135.0.128:5060>
Supported: 100rel
Content-Length:  309
Content-Type: application/sdp

v=0
o=2475098871 10 10 IN IP4 147.135.2.247
s=-
c=IN IP4 147.135.2.250
t=0 0
m=audio 28274 RTP/AVP 0 8 18 96 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:97 t38/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (10 headers 14 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 147.135.0.128 : 5060 (no NAT)
Using INVITE request as basis request - 320190-32 at 147.135.0.128
No user '<my cell #>' in SIP users list
Found peer 'sip.broadvoice.com' for '<my cell #>' from
147.135.0.128:5060
net-xero*CLI>
<--- Reliably Transmitting (no NAT) to 147.135.0.128:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128
From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=prtu
To: "<my name>"<sip:s@<servers IP>>;tag=as77a74c13
Call-ID: 320190-32 at 147.135.0.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX SVN-trunk-r106946
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="06b61489"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '320190-32 at 147.135.0.128' in
32000 ms (Method: INVITE)
net-xero*CLI>
<--- SIP read from UDP://147.135.0.128:5060 --->
ACK sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0
Call-ID: 320190-32 at 147.135.0.128
CSeq: 1 ACK
From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=prtu
To: "<my name>"<sip:s@<servers IP>>;tag=as77a74c13
Via: SIP/2.0/UDP 147.135.0.128:5060
Content-Length:    0


<------------->
--- (7 headers 0 lines) ---
[Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:9020 sip_reregister:    --
Re-registration for  <my Broadvoice #>@sip.broadvoice.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.0.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP <servers IP>:5060;branch=z9hG4bK339bbe4e;rport
Max-Forwards: 70
From: <sip:<my Broadvoice #>@sip.broadvoice.com>;tag=as2a457f50
To: <sip:<my Broadvoice #>@sip.broadvoice.com>
Call-ID: 7c356bca35c9e62f6c90c3287466e70a at sip.broadvoice.com
CSeq: 104 REGISTER
User-Agent: Asterisk PBX SVN-trunk-r106946
Expires: 120
Contact: <sip:s@<servers IP>>
Event: registration
Content-Length: 0


---
net-xero*CLI>
<--- SIP read from UDP://147.135.0.128:5060 --->
SIP/2.0 200 OK
Call-ID: 7c356bca35c9e62f6c90c3287466e70a at sip.broadvoice.com
CSeq: 104 REGISTER
From: <sip:<my Broadvoice #>@sip.broadvoice.com>;tag=as2a457f50
To: <sip:<my Broadvoice #>@sip.broadvoice.com>
Via: SIP/2.0/UDP <servers IP>:5060;branch=z9hG4bK339bbe4e
Contact: <sip:s@<servers IP>>
Expires: 30
Event: registration
Content-Length:    0


<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog
'7c356bca35c9e62f6c90c3287466e70a at sip.broadvoice.com' in 32000 ms
(Method: REGISTER)
[Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:14949
handle_response_register: Outbound Registration: Expiry for
sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
net-xero*CLI>
<--- SIP read from UDP://147.135.0.128:5060 --->
INVITE sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0
Call-ID: 660240-66 at 147.135.0.128
CSeq: 1 INVITE
From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=ikmn
To: "<my name>"<sip:s@<servers IP>>
Via: SIP/2.0/UDP 147.135.0.128:5060
Contact: <sip:<my cell #>@147.135.0.128:5060>
Supported: 100rel
Content-Length:  309
Content-Type: application/sdp

v=0
o=2475098871 10 10 IN IP4 147.135.2.247
s=-
c=IN IP4 147.135.2.250
t=0 0
m=audio 28276 RTP/AVP 0 8 18 96 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:97 t38/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (10 headers 14 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 147.135.0.128 : 5060 (no NAT)
Using INVITE request as basis request - 660240-66 at 147.135.0.128
No user '<my cell #>' in SIP users list
Found peer 'sip.broadvoice.com' for '<my cell #>' from
147.135.0.128:5060
net-xero*CLI>
<--- Reliably Transmitting (no NAT) to 147.135.0.128:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128
From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=ikmn
To: "<my name>"<sip:s@<servers IP>>;tag=as6ef11459
Call-ID: 660240-66 at 147.135.0.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX SVN-trunk-r106946
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="1a011874"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '660240-66 at 147.135.0.128' in
32000 ms (Method: INVITE)
net-xero*CLI>
<--- SIP read from UDP://147.135.0.128:5060 --->
ACK sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0
Call-ID: 660240-66 at 147.135.0.128
CSeq: 1 ACK
From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=ikmn
To: "<my name>"<sip:s@<servers IP>>;tag=as6ef11459
Via: SIP/2.0/UDP 147.135.0.128:5060
Content-Length:    0


<------------->
--- (7 headers 0 lines) ---



sip.conf:
register => <username>:<password>@sip.broadvoice.com

[sip.broadvoice.com]
type=peer
user=<username>
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=<username>
secret=<password>
username=<username>
insecure=very
context=from-bv
authname=<username>
dtmfmode=inband
dtmf=inband
canreinvite=yes

extensions.conf:

[from-bv]
exten => s,1,Answer()
exten => s,n,MusicOnHold

exten => <number>,Answer()
exten => <number>,n,MusicOnHold

I did these 2 lines for debugging purposes.  the dialplan is a little
more complex but because this didn't even work, there's no point in
posting.

Does anyone have any idea why this works fine when I was using 1.2 but
suddenly with 1.4.18 it isn't?  This is on a server connected directly
to the internet, no NAT.  Nothing else has changed on it, and
Link2Voip (SIP) and Vittelity (IAX) works flawlessly.  Any help would
be GREATLY appreciated.  Thanks in advance!

Raj Jain

2008-Mar-16 20:45 UTC

head link

[asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18

Looking at the trace, the entity sending you the INVITE is not
resubmitting INVITE with credentials after the initial INVITE was
challenged with a 401 response by Asterisk. The trace shows two
independent calls and both have the same problem.

-- 
Raj Jain

mailto:rj2807 at gmail dot com
sip:rjain at iptel dot org


On Sun, Mar 16, 2008 at 4:10 PM, Jon Miron <mironj at gmail.com>
wrote:> Hi all,
>
>  I just upgraded to Asterisk 1.4.18 a few days ago and I don't use
>  Broadvoice TOO often, however I have a Vermont number with them and so
>  my mother in law calls it to talk to my wife once in a while, so
>  that's why it took me so long to notice it wasn't working. 
Anyway,
>  when she calls she gets a busy signal (as I've tested when calling it
>  from my cell).
>
>  When I enable debugging I get the following:
>
>  SIP Debugging Enabled for IP: 147.135.0.128
>  net-xero*CLI>
>  <--- SIP read from UDP://147.135.0.128:5060 --->
>  INVITE sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0
>  Call-ID: 320190-32 at 147.135.0.128
>  CSeq: 1 INVITE
>  From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=prtu
>  To: "<my name>"<sip:s@<servers IP>>
>  Via: SIP/2.0/UDP 147.135.0.128:5060
>  Contact: <sip:<my cell #>@147.135.0.128:5060>
>  Supported: 100rel
>  Content-Length:  309
>  Content-Type: application/sdp
>
>  v=0
>  o=2475098871 10 10 IN IP4 147.135.2.247
>  s=-
>  c=IN IP4 147.135.2.250
>  t=0 0
>  m=audio 28274 RTP/AVP 0 8 18 96 97 101
>  a=rtpmap:0 PCMU/8000
>  a=rtpmap:8 PCMA/8000
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=no
>  a=rtpmap:96 iLBC/8000
>  a=fmtp:96 mode=30
>  a=rtpmap:97 t38/8000
>  a=rtpmap:101 telephone-event/8000
>
>  <------------->
>  --- (10 headers 14 lines) ---
>   == Using SIP RTP CoS mark 5
>  Sending to 147.135.0.128 : 5060 (no NAT)
>  Using INVITE request as basis request - 320190-32 at 147.135.0.128
>  No user '<my cell #>' in SIP users list
>  Found peer 'sip.broadvoice.com' for '<my cell #>'
from 147.135.0.128:5060
>  net-xero*CLI>
>  <--- Reliably Transmitting (no NAT) to 147.135.0.128:5060 --->
>  SIP/2.0 401 Unauthorized
>  Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128
>  From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=prtu
>  To: "<my name>"<sip:s@<servers
IP>>;tag=as77a74c13
>  Call-ID: 320190-32 at 147.135.0.128
>  CSeq: 1 INVITE
>  User-Agent: Asterisk PBX SVN-trunk-r106946
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces, timer
>  WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="06b61489"
>  Content-Length: 0
>
>
>  <------------>
>  Scheduling destruction of SIP dialog '320190-32 at 147.135.0.128'
in
>  32000 ms (Method: INVITE)
>  net-xero*CLI>
>  <--- SIP read from UDP://147.135.0.128:5060 --->
>  ACK sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0
>  Call-ID: 320190-32 at 147.135.0.128
>  CSeq: 1 ACK
>  From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=prtu
>  To: "<my name>"<sip:s@<servers
IP>>;tag=as77a74c13
>  Via: SIP/2.0/UDP 147.135.0.128:5060
>  Content-Length:    0
>
>
>  <------------->
>  --- (7 headers 0 lines) ---
>  [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:9020 sip_reregister:    --
>  Re-registration for  <my Broadvoice #>@sip.broadvoice.com
>  REGISTER 12 headers, 0 lines
>  Reliably Transmitting (no NAT) to 147.135.0.128:5060:
>  REGISTER sip:sip.broadvoice.com SIP/2.0
>  Via: SIP/2.0/UDP <servers IP>:5060;branch=z9hG4bK339bbe4e;rport
>  Max-Forwards: 70
>  From: <sip:<my Broadvoice
#>@sip.broadvoice.com>;tag=as2a457f50
>  To: <sip:<my Broadvoice #>@sip.broadvoice.com>
>  Call-ID: 7c356bca35c9e62f6c90c3287466e70a at sip.broadvoice.com
>  CSeq: 104 REGISTER
>  User-Agent: Asterisk PBX SVN-trunk-r106946
>  Expires: 120
>  Contact: <sip:s@<servers IP>>
>  Event: registration
>  Content-Length: 0
>
>
>  ---
>  net-xero*CLI>
>  <--- SIP read from UDP://147.135.0.128:5060 --->
>  SIP/2.0 200 OK
>  Call-ID: 7c356bca35c9e62f6c90c3287466e70a at sip.broadvoice.com
>  CSeq: 104 REGISTER
>  From: <sip:<my Broadvoice
#>@sip.broadvoice.com>;tag=as2a457f50
>  To: <sip:<my Broadvoice #>@sip.broadvoice.com>
>  Via: SIP/2.0/UDP <servers IP>:5060;branch=z9hG4bK339bbe4e
>  Contact: <sip:s@<servers IP>>
>  Expires: 30
>  Event: registration
>  Content-Length:    0
>
>
>  <------------->
>  --- (10 headers 0 lines) ---
>  Scheduling destruction of SIP dialog
>  '7c356bca35c9e62f6c90c3287466e70a at sip.broadvoice.com' in 32000
ms
>  (Method: REGISTER)
>  [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:14949
>  handle_response_register: Outbound Registration: Expiry for
>  sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
>  net-xero*CLI>
>  <--- SIP read from UDP://147.135.0.128:5060 --->
>  INVITE sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0
>  Call-ID: 660240-66 at 147.135.0.128
>  CSeq: 1 INVITE
>  From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=ikmn
>  To: "<my name>"<sip:s@<servers IP>>
>  Via: SIP/2.0/UDP 147.135.0.128:5060
>  Contact: <sip:<my cell #>@147.135.0.128:5060>
>  Supported: 100rel
>  Content-Length:  309
>  Content-Type: application/sdp
>
>  v=0
>  o=2475098871 10 10 IN IP4 147.135.2.247
>  s=-
>  c=IN IP4 147.135.2.250
>  t=0 0
>  m=audio 28276 RTP/AVP 0 8 18 96 97 101
>  a=rtpmap:0 PCMU/8000
>  a=rtpmap:8 PCMA/8000
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=no
>  a=rtpmap:96 iLBC/8000
>  a=fmtp:96 mode=30
>  a=rtpmap:97 t38/8000
>  a=rtpmap:101 telephone-event/8000
>
>  <------------->
>  --- (10 headers 14 lines) ---
>   == Using SIP RTP CoS mark 5
>  Sending to 147.135.0.128 : 5060 (no NAT)
>  Using INVITE request as basis request - 660240-66 at 147.135.0.128
>  No user '<my cell #>' in SIP users list
>  Found peer 'sip.broadvoice.com' for '<my cell #>'
from 147.135.0.128:5060
>  net-xero*CLI>
>  <--- Reliably Transmitting (no NAT) to 147.135.0.128:5060 --->
>  SIP/2.0 401 Unauthorized
>  Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128
>  From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=ikmn
>  To: "<my name>"<sip:s@<servers
IP>>;tag=as6ef11459
>  Call-ID: 660240-66 at 147.135.0.128
>  CSeq: 1 INVITE
>  User-Agent: Asterisk PBX SVN-trunk-r106946
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces, timer
>  WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="1a011874"
>  Content-Length: 0
>
>
>  <------------>
>  Scheduling destruction of SIP dialog '660240-66 at 147.135.0.128'
in
>  32000 ms (Method: INVITE)
>  net-xero*CLI>
>  <--- SIP read from UDP://147.135.0.128:5060 --->
>  ACK sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0
>  Call-ID: 660240-66 at 147.135.0.128
>  CSeq: 1 ACK
>  From: "Toronto ON"<sip:<my cell
#>@147.135.0.128;user=phone>;tag=ikmn
>  To: "<my name>"<sip:s@<servers
IP>>;tag=as6ef11459
>  Via: SIP/2.0/UDP 147.135.0.128:5060
>  Content-Length:    0
>
>
>  <------------->
>  --- (7 headers 0 lines) ---
>
>
>
>  sip.conf:
>  register => <username>:<password>@sip.broadvoice.com
>
>  [sip.broadvoice.com]
>  type=peer
>  user=<username>
>  host=sip.broadvoice.com
>  fromdomain=sip.broadvoice.com
>  fromuser=<username>
>  secret=<password>
>  username=<username>
>  insecure=very
>  context=from-bv
>  authname=<username>
>  dtmfmode=inband
>  dtmf=inband
>  canreinvite=yes
>
>  extensions.conf:
>
>  [from-bv]
>  exten => s,1,Answer()
>  exten => s,n,MusicOnHold
>
>  exten => <number>,Answer()
>  exten => <number>,n,MusicOnHold
>
>  I did these 2 lines for debugging purposes.  the dialplan is a little
>  more complex but because this didn't even work, there's no point
in
>  posting.
>
>  Does anyone have any idea why this works fine when I was using 1.2 but
>  suddenly with 1.4.18 it isn't?  This is on a server connected directly
>  to the internet, no NAT.  Nothing else has changed on it, and
>  Link2Voip (SIP) and Vittelity (IAX) works flawlessly.  Any help would
>  be GREATLY appreciated.  Thanks in advance!
>
>  _______________________________________________
>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>  asterisk-users mailing list
>  To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

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