6 Replies Last post: Nov 29, 2008 9:35 AM by Guest  
Guest

Nov 18, 2008 12:04 PM

[OpenSBC] Fw: SipX<=>OpenSBC: inbound route doesn't work

Hi

I am using a inGate SBC and i am able to hairpin. I did run into a proble
at first but it turned out to be something else. But along the way i was
able to route inbound to the sipX by changing the port away form 5060,
just pick one like 6070 then set up a rule that looked for and took traffic
from port 6070 and set it Forward back to 5060 out to the SBC => ITSP. The
other and final solution was changing the remote phone from Grandstream to
Snom
because Grandstream fails in all phones to manage RE-Invites. Lost two
weeks screwing with the Grandstreams.

AT&T analog phone <==> Sip trunk ITSP <==> ((inGate <==> sipX(
registration ) back out <==> SiP Trunk ITSP)(high speed T1)) <==> and ring
a ip phone connected to an AT&T DSL router in another part of the office.

Don't do STUN. Do Outbound Proxy instead.

r


Original Message

From: "OpenSBC Forum"
To:
Sent: Tuesday, November 18, 2008 9:50 AM
Subject: Re: OpenSBC SipX<=>OpenSBC: inbound route doesn't work


I guess I can make two calls over sip-trunk at the same time.

I want to test incoming calls but I can't call from PSTN number. So
I'm
trying to call from my internal sipx number over sip-trunk to another
number
which must be routed to another internal number.


According to this sip-trunk config I dial 44555555555 from number
sip:100@sipx.mydomain and want to be routed to number
sip:200@sipx.mydomain


Is it correct?

&lt;xml&gt;
&lt;siptrunk trunk-name="provider.org" route-set="provider.org"
sip-domain="provider.org" expires="20"&gt;
&lt;trunk-accounts&gt;
&lt;account user-name="44444444444" auth-user-name="44444444444"
auth-password="mypass" inboundroute="sip:100@sipx.mydomain"
expires="3600"
/&gt;
&lt;account user-name="44555555555" auth-user-name="44555555555"
auth-password="mypass" inboundroute="sip:200@sipx.mydomain"
expires="3600"
/&gt;
&lt;/trunk-accounts&gt;
&lt;transient-accounts&gt;
&lt;account user-name="44444444444" auth-user-name="44444444444"
auth-password="mypass" inboundroute="sip:100@sipx.ebc.mydomain" /&gt;
&lt;account user-name="44555555555" auth-user-name="44555555555"
auth-password="mypass" inboundroute="sip:200@sipx.mydomain" /&gt;
&lt;/transient-accounts&gt;
&lt;/siptrunk&gt;
&lt;/xml&gt;

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Guest
1. Nov 18, 2008 11:48 PM in response to: Guest
Re: [OpenSBC] Fw: SipX<=>OpenSBC: inbound route doesn't work
Hi Robert,

OpenSBC can do hairpin if you do it correctly. The outbound call should
be sent through the main trunk

UA1 -> OSBC-Main-Trunk -> SIP-Server

When the SIP-Server hairpins/trombones the call back to OpenSBC, it must
either use one of two listeners based on which config applies

    • SIP-Server has static routing policies
SIP-Server -> OSBC-BackDoor -> OSBC-Main-Trunk -> UA2

Or

    • SIP-Server configured as a trunk provider
SIP-Server -> OSBC-SIP-Trunk -> OSBC-Main-Trunk -> UA2

In the case of the original poster, he is trying to trombone the call
back to the SIP-Trunk which would cause call-id collisions

Joegen

voice wrote:
Hi

I am using a inGate SBC and i am able to hairpin. I did run into a proble
at first but it turned out to be something else. But along the way i was
able to route inbound to the sipX by changing the port away form 5060,
just pick one like 6070 then set up a rule that looked for and took traffic
from port 6070 and set it Forward back to 5060 out to the SBC => ITSP. The
other and final solution was changing the remote phone from Grandstream to
Snom
because Grandstream fails in all phones to manage RE-Invites. Lost two
weeks screwing with the Grandstreams.

AT&T analog phone <==> Sip trunk ITSP <==> ((inGate <==> sipX(
registration ) back out <==> SiP Trunk ITSP)(high speed T1)) <==> and ring
a ip phone connected to an AT&T DSL router in another part of the office.

Don't do STUN. Do Outbound Proxy instead.

r


This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
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Guest
2. Nov 20, 2008 5:51 PM in response to: Guest
Re: [OpenSBC] Fw: SipX<=>OpenSBC: inbound route doesn't work
Hi Joegen

Thanks.

Is there a CVS way to get openSBC? I can get the opensipstack aok

r


Original Message

From: "Joegen E. Baclor"
To: "voice" ;

Sent: Tuesday, November 18, 2008 10:48 PM
Subject: Re: OpenSBC Fw: SipX<=>OpenSBC: inbound route doesn't work

Hi Robert,

OpenSBC can do hairpin if you do it correctly. The outbound call should
be sent through the main trunk

UA1 -> OSBC-Main-Trunk -> SIP-Server

When the SIP-Server hairpins/trombones the call back to OpenSBC, it must
either use one of two listeners based on which config applies

    • SIP-Server has static routing policies
SIP-Server -> OSBC-BackDoor -> OSBC-Main-Trunk -> UA2

Or

    • SIP-Server configured as a trunk provider
SIP-Server -> OSBC-SIP-Trunk -> OSBC-Main-Trunk -> UA2

In the case of the original poster, he is trying to trombone the call
back to the SIP-Trunk which would cause call-id collisions

Joegen

voice wrote:
Hi

I am using a inGate SBC and i am able to hairpin. I did run into a
proble
at first but it turned out to be something else. But along the way i
was
able to route inbound to the sipX by changing the port away form 5060,
just pick one like 6070 then set up a rule that looked for and took
traffic
from port 6070 and set it Forward back to 5060 out to the SBC => ITSP.
The
other and final solution was changing the remote phone from Grandstream
to
Snom
because Grandstream fails in all phones to manage RE-Invites. Lost two
weeks screwing with the Grandstreams.

AT&T analog phone <==> Sip trunk ITSP <==> ((inGate <==> sipX(
registration ) back out <==> SiP Trunk ITSP)(high speed T1)) <==> and
ring
a ip phone connected to an AT&T DSL router in another part of the
office.

Don't do STUN. Do Outbound Proxy instead.

r




This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
Opensipstack-osbcdevel mailing list
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https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
Guest
3. Nov 24, 2008 3:49 PM in response to: Guest
Re: [OpenSBC] Fw: SipX<=>OpenSBC: Sip Trunk config username/password not required
Hi Joegen

I am using Broadvox for SIP Trunk ITSP orgination and termination. I don't
need to send them a username and passwd. Can i still us Sip Trunk
configuration. It seems that the OSBC config requires this setup. Is there
a way to turn off registration?

r


Original Message

From: "Joegen E. Baclor"
To: "voice" ;

Sent: Tuesday, November 18, 2008 10:48 PM
Subject: Re: OpenSBC Fw: SipX<=>OpenSBC: inbound route doesn't work

Hi Robert,

OpenSBC can do hairpin if you do it correctly. The outbound call should
be sent through the main trunk

UA1 -> OSBC-Main-Trunk -> SIP-Server

When the SIP-Server hairpins/trombones the call back to OpenSBC, it must
either use one of two listeners based on which config applies

    • SIP-Server has static routing policies
SIP-Server -> OSBC-BackDoor -> OSBC-Main-Trunk -> UA2

Or

    • SIP-Server configured as a trunk provider
SIP-Server -> OSBC-SIP-Trunk -> OSBC-Main-Trunk -> UA2

In the case of the original poster, he is trying to trombone the call
back to the SIP-Trunk which would cause call-id collisions

Joegen

voice wrote:
Hi

I am using a inGate SBC and i am able to hairpin. I did run into a
proble
at first but it turned out to be something else. But along the way i
was
able to route inbound to the sipX by changing the port away form 5060,
just pick one like 6070 then set up a rule that looked for and took
traffic
from port 6070 and set it Forward back to 5060 out to the SBC => ITSP.
The
other and final solution was changing the remote phone from Grandstream
to
Snom
because Grandstream fails in all phones to manage RE-Invites. Lost two
weeks screwing with the Grandstreams.

AT&T analog phone <==> Sip trunk ITSP <==> ((inGate <==> sipX(
registration ) back out <==> SiP Trunk ITSP)(high speed T1)) <==> and
ring
a ip phone connected to an AT&T DSL router in another part of the
office.

Don't do STUN. Do Outbound Proxy instead.

r




This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
Opensipstack-osbcdevel mailing list
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https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
Click to view joegen's profile   136 posts since
Apr 28, 2007
4. Nov 24, 2008 8:50 PM in response to: Guest
Re: [OpenSBC] Fw: SipX<=>OpenSBC: Sip Trunk config username/password not required
Guest wrote:
Hi Joegen

I am using Broadvox for SIP Trunk ITSP orgination and termination. I don't
need to send them a username and passwd. Can i still us Sip Trunk
configuration. It seems that the OSBC config requires this setup. Is there
a way to turn off registration?

r


Robert,

There is now. I have introduced a new attribute for sip-trunk named send-reg. Just set it to "no" in you trunk account config.

send-reg="no"
auth-user-name="1003"
auth-password="1003"
inbound-route="sip:9003@win32.opensipstack.org"

Please upgrade using CVS head and let me know if the patch works for you.

Joegen
Guest
5. Nov 25, 2008 2:23 PM in response to: Guest
Re: [OpenSBC] Fw: SipX<=>OpenSBC: Inbound and outbound Sip Trunks
Hi Joegen

Origination and Termination sip trunks use 2 different ip-addrs. Can i set
up the Sip Trunk-config to allow both inbound and out bound sip traffic? If
yes are there any examples?

What/why are there transcient-account?

r


Original Message

From: "Joegen E. Baclor"
To: "voice" ;

Sent: Tuesday, November 18, 2008 10:48 PM
Subject: Re: OpenSBC Fw: SipX<=>OpenSBC: inbound route doesn't work

Hi Robert,

OpenSBC can do hairpin if you do it correctly. The outbound call should
be sent through the main trunk

UA1 -> OSBC-Main-Trunk -> SIP-Server

When the SIP-Server hairpins/trombones the call back to OpenSBC, it must
either use one of two listeners based on which config applies

    • SIP-Server has static routing policies
SIP-Server -> OSBC-BackDoor -> OSBC-Main-Trunk -> UA2

Or

    • SIP-Server configured as a trunk provider
SIP-Server -> OSBC-SIP-Trunk -> OSBC-Main-Trunk -> UA2

In the case of the original poster, he is trying to trombone the call
back to the SIP-Trunk which would cause call-id collisions

Joegen

voice wrote:
Hi

I am using a inGate SBC and i am able to hairpin. I did run into a
proble
at first but it turned out to be something else. But along the way i
was
able to route inbound to the sipX by changing the port away form 5060,
just pick one like 6070 then set up a rule that looked for and took
traffic
from port 6070 and set it Forward back to 5060 out to the SBC => ITSP.
The
other and final solution was changing the remote phone from Grandstream
to
Snom
because Grandstream fails in all phones to manage RE-Invites. Lost two
weeks screwing with the Grandstreams.

AT&T analog phone <==> Sip trunk ITSP <==> ((inGate <==> sipX(
registration ) back out <==> SiP Trunk ITSP)(high speed T1)) <==> and
ring
a ip phone connected to an AT&T DSL router in another part of the
office.

Don't do STUN. Do Outbound Proxy instead.

r




This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
Opensipstack-osbcdevel mailing list
Opensipstack-osbcdevel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
Guest
6. Nov 29, 2008 9:35 AM in response to: Guest
[OpenSBC] Listener ***Failure***
Hi Joegen

Happy Thanksgiving

In a 2 dual NIC box I have set Listeners to the WAN ipaddr only, whether I
leave at default or i set both or just one each tells me error adresses
already exists. ***Failure***.

Why does this happen and how do I begin to fix it?

r



This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
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Opensipstack-osbcdevel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel