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

Raj Jain rj2807 at gmail.com
Sun Mar 16 15:45:24 CDT 2008


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!
>
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