Thread: [OpenSIPStack] incoming routes
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From: OpenSIPStack F. <ope...@op...> - 2008-06-19 07:35:45
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Hello. I finally got my trunks to register. This issue I currently have is when a incoming call is received opensbc replies back with Internal Server Error 500. Here is the invite: U 64.34.181.47:5060 -> 204.13.XX.XX:5066 INVITE sip:78621XXX@204.13.XX.XX:5066 SIP/2.0 Via: SIP/2.0/UDP 64.34.181.47:5060;branch=z9hG4bK68d77e9c;rport From: "NOT FOUND" <sip:13052XXXXX@64.34.181.47>;tag=as1941af79 To: <sip:78621XXXX@204.13.XX.XX:5066> Contact: <sip:130529XXX@64.34.181.47> Call-ID: 046e8d7b24d7315c5074495a46fb29ba@64.34.181.47 CSeq: 102 INVITE User-Agent: LES.NET.VoIP Max-Forwards: 70 Date: Thu, 19 Jun 2008 07:30:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 317 Then opensbc responds back with: U 204.13.XX.XX:5066 -> 64.34.181.47:5060 SIP/2.0 500 Internal Server Error From: "NOT FOUND" <sip:1305XXXXXX@64.34.181.47>;tag=as1941af79 To: <sip:7862XXXXX@204.13.XX.XX:5066>;tag=c298d0d21d3cdd119638a704fbd83836 Via: SIP/2.0/UDP 64.34.181.47:5060;branch=z9hG4bK68d77e9c;rport=5060;received=64.34.181.47 CSeq: 102 INVITE Call-ID: 046e8d7b24d7315c5074495a46fb29ba@64.34.181.47 Content-Length: 0 In my trunk configuration I have inbound route=sip:XX...@si... (which is a registered UA on the same server) The URI the provider sends is the exact URI in my configuration. I have restarted multiple times to no avail :( Any help is appreciated! Thanks |
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From: OpenSIPStack F. <ope...@op...> - 2008-06-20 01:11:46
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It won't work for this INVITE. SIPTrunks are identified through their domains. In your previous post you indicated that the domain is sip-domain="did.les.net". This should be the same domain that your provider should use in the TO-URI for inbound calls. Instead they sent sip:78621XXXX@204.13.XX.XX:5066 which is obviously the contact address. Joegen > {quote:title=illizit wrote:}{quote} > > Hello. > > > I finally got my trunks to register. This issue I currently have is when a incoming call is received opensbc replies back with Internal Server Error 500. > > > Here is the invite: > > > > U 64.34.181.47:5060 -> 204.13.XX.XX:5066 > INVITE sip:78621XXX@204.13.XX.XX:5066 SIP/2.0 > Via: SIP/2.0/UDP 64.34.181.47:5060;branch=z9hG4bK68d77e9c;rport > From: "NOT FOUND" <sip:13052XXXXX@64.34.181.47>;tag=as1941af79 > To: <sip:78621XXXX@204.13.XX.XX:5066> > Contact: <sip:130529XXX@64.34.181.47> > Call-ID: 046e8d7b24d7315c5074495a46fb29ba@64.34.181.47 > CSeq: 102 INVITE > User-Agent: LES.NET.VoIP > Max-Forwards: 70 > Date: Thu, 19 Jun 2008 07:30:16 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 317 > > > > > > > > > Then opensbc responds back with: > > > > U 204.13.XX.XX:5066 -> 64.34.181.47:5060 > SIP/2.0 500 Internal Server Error > From: "NOT FOUND" <sip:1305XXXXXX@64.34.181.47>;tag=as1941af79 > To: <sip:7862XXXXX@204.13.XX.XX:5066>;tag=c298d0d21d3cdd119638a704fbd83836 > Via: SIP/2.0/UDP 64.34.181.47:5060;branch=z9hG4bK68d77e9c;rport=5060;received=64.34.181.47 > CSeq: 102 INVITE > Call-ID: 046e8d7b24d7315c5074495a46fb29ba@64.34.181.47 > Content-Length: 0 > > > > > > In my trunk configuration I have inbound route=sip:XX...@si... (which is a registered UA on the same server) > > > The URI the provider sends is the exact URI in my configuration. I have restarted multiple times to no avail :( > > > Any help is appreciated! > > > Thanks > > |
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From: OpenSIPStack F. <ope...@op...> - 2008-06-20 02:03:04
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All three providers (including gafachi) use this standard. Are you telling me it's not compatible? The TO field, I thought, was supposed to be used for the number that was dialed. For instance, if I call from my cell phone 55555 to the number 88888, then TO should be: TO:[88...@op...] and FROM should be: FROM:[55...@yo...] |
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From: OpenSIPStack F. <ope...@op...> - 2008-06-20 02:57:50
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Let us say your REGISTER to your provider looks like this
REGISTER sip:11...@pr... SIP/2.0
To: sip:11...@yo...;tag=12345
From: sip:11...@yo...
Contact: sip:1111@10.0.0.1:5066
then your provider should send the INVITE as follows
INVITE sip:1111@10.0.0.1:5066 SIP/2.0
To: sip:11...@yo...;tag=6789
From: sip:com...@yo...
This is simply how it works now in OSBC SIP Trunking. OpenSBC will take the domain from the To-URI and match it to the trunk you specified in the XML config.
> {quote:title=illizit wrote:}{quote}
>
> All three providers (including gafachi) use this standard. Are you telling me it's not compatible?
>
>
> The TO field, I thought, was supposed to be used for the number that was dialed.
>
>
> For instance, if I call from my cell phone 55555 to the number 88888, then TO should be: TO:[88...@op...] and FROM should be: FROM:[55...@yo...]
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From: OpenSIPStack F. <ope...@op...> - 2008-06-20 03:43:23
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Is there a specific reason why it works like this? If not, do you know which part of the code this sip trunking is in so I can modify it? According to this website, the from tag in a sip packet should say where the call originated from and the to tag should be the uri of where it is going. [http://www.tech-invite.com/Ti-sip-dialog.html#inv] |
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From: OpenSIPStack F. <ope...@op...> - 2008-06-20 04:03:03
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> {quote:title=illizit wrote:}{quote}
> Is there a specific reason why it works like this?
>
RFC 3261
10.3 Processing REGISTER Requests
5. The registrar extracts the address-of-record from the To header
field of the request. If the address-of-record is not valid
for the domain in the Request-URI, the registrar MUST send a
404 (Not Found) response and skip the remaining steps. The URI
MUST then be converted to a canonical form. To do that, all
URI parameters MUST be removed (including the user-param), and
any escaped characters MUST be converted to their unescaped
form. The result serves as an index into the list of bindings
I simply mean your provider is not sending the correct AOR it got from the REGISTER request. And that correct AOR processing is vital for OSBC SIP Trunking operation.
> If not, do you know which part of the code this sip trunking is in so I can modify it? According to this website, the from tag in a sip packet should say where the call originated from and the to tag should be the uri of where it is going.
>
>
See:
B2BUAConnection * SBCSIPTrunkEndPoint::OnCreateB2BUA(
const SIPMessage & request,
const OString & sessionId,
B2BUACall * /*call*/
)
Feel free to send in patches.
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From: OpenSIPStack F. <ope...@op...> - 2008-06-20 04:37:03
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Hmm. ok. Thanks for the help! Sorry to keep bothering however, what exactly should the provider be sending? Here is my SIP Register packet to the provider: 204.13.X.XX:5066 -> 64.34.181.47:5060 REGISTER sip:did.voip.les.net SIP/2.0.. From: <sip:151...@di...>;tag=920ef16ccc3cdd119562d496dbc76295.. To: sip:151...@di.....Via: SIP/2.0/UDP 204.13.7.33:5066;iid=9578;branch=z9hG4bKa6d1f36fcc3cdd119562d496dbc76295;uas-addr=64.34.181.47;rport..CSeq: 2 REGISTER.. Call-ID:[1c1...@di...].. Contact: <sip:1518421590@204.13.7.33:5066;transport=udp>.. User-Agent: ..Expires: 3600..Max-Forwards: 70..Authorization: Digest username="1518421590", realm="did.voip.les.net", nonce="3b131f7f", uri="sip:did.voip.les.net", response="e62e92397699 10e84475b05d72ac35e6", opaque="", algorithm=MD5..Allow: INVITE, BYE, ACK, R EFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK..Content-Length: 0.... # According to my register packet, what should the provider be sending me? Thanks |
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From: OpenSIPStack F. <ope...@op...> - 2008-06-20 05:08:44
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It should be something like below:
INVITE sip:1518421590@204.13.7.33:5066 SIP/2.0 <----------------- CONTACT you sent in your REGISTER request (Binding)
From: <sip:121...@di...>;tag=920ef16ccc3cdd119562d496dbc76295
To: sip:151...@di... <------------------------ or To-URI you sent in REGISTER (Address of Record)
> {quote:title=illizit wrote:}{quote}
>
> Hmm. ok.
>
>
> Thanks for the help! Sorry to keep bothering however, what exactly should the provider be sending?
>
>
> Here is my SIP Register packet to the provider:
>
>
> 204.13.X.XX:5066 -> 64.34.181.47:5060
> REGISTER sip:did.voip.les.net SIP/2.0..
> From: <sip:151...@di...>;tag=920ef16ccc3cdd119562d496dbc76295..
> To: sip:151...@di.....Via: SIP/2.0/UDP 204.13.7.33:5066;iid=9578;branch=z9hG4bKa6d1f36fcc3cdd119562d496dbc76295;uas-addr=64.34.181.47;rport..CSeq: 2 REGISTER..
> Call-ID:[1c1...@di...]..
> Contact: <sip:1518421590@204.13.7.33:5066;transport=udp>..
> User-Agent: ..Expires: 3600..Max-Forwards: 70..Authorization: Digest username="1518421590", realm="did.voip.les.net", nonce="3b131f7f", uri="sip:did.voip.les.net", response="e62e92397699
> 10e84475b05d72ac35e6", opaque="", algorithm=MD5..Allow: INVITE, BYE, ACK, R
> EFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK..Content-Length: 0....
> #
>
>
>
>
>
> According to my register packet, what should the provider be sending me?
>
>
> Thanks
>
>
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From: OpenSIPStack F. <ope...@op...> - 2008-06-20 05:16:58
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Thanks for the help! |
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