[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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