Cannot Xfer Calls Received from Unmanaged Trunks

[Chris Parker]

Kind of stumped with this one...

I have Vitelity as my SIP trunk, which is configured in Asterisk to answer with an IVR and perform some other functions.
If a call is passed from my AA in Asterisk or the DID is configured to call an extension that belongs to my sipX box (Polycom phones), I cannot transfer that call to any other internal extension or park that call. The same is true if a call comes from another PBX (CUCM) - both gateways are configured as "Unmanaged Gateway"

If I call out from sipX to the PSTN (there is also a trunk configured with Vitelity on sipX) I can place that outbound call in park orbit or transfer them to another internal extension.

I'm more familiar with troubleshooting Asterisk than sipX, so consider me a newbie in this regard. I will gladly post configs / dump log files upon request once I know what I'm looking for.

[Tony Graziano]

An unmanaged gateway is just that. Can I assume that the address for both
systems are on the same subnet? Unmanaged gateways would assume that the
other and knows how to handle the SIP REFER method.

Asterisk as a sip trunking system is not exactly compliant.

If REFER is not supported, then the media needs to be anchored by sipx once
it accepts the call and hold the REFER internally, at which point you would
setup a manual sip TRUNK not a gateway.

Post by Chris Parker
Kind of stumped with this one...
I have Vitelity as my SIP trunk, which is configured in Asterisk to answer
with an IVR and perform some other functions.
If a call is passed from my AA in Asterisk or the DID is configured to
call an extension that belongs to my sipX box (Polycom phones), I cannot
transfer that call to any other internal extension or park that call. The
same is true if a call comes from another PBX (CUCM) - both gateways are
configured as "Unmanaged Gateway"
If I call out from sipX to the PSTN (there is also a trunk configured with
Vitelity on sipX) I can place that outbound call in park orbit or transfer
them to another internal extension.
I'm more familiar with troubleshooting Asterisk than sipX, so consider me
a newbie in this regard. I will gladly post configs / dump log files upon
request once I know what I'm looking for.
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

[Josh Patten]

The biggest red flags are Asterisk and CUCM

The SIP stacks on these platforms aren't complete and REFER support is
usually lacking. Trunking with these platforms requires a session border
controller to interface with these platforms. How many active calls do you
expect between these systems? If it's a small amount you could probably get
away with the internal sipXbridge SBC. Otherwise you could look into using
a Patton or Ingate SBC, or you could roll your own with FreeSWITCH:
http://wiki.sipfoundry.org/display/sipXecs/FreeSWITCH+SIP+Trunking+Gateway

Post by Chris Parker
Kind of stumped with this one...
I have Vitelity as my SIP trunk, which is configured in Asterisk to answer
with an IVR and perform some other functions.
If a call is passed from my AA in Asterisk or the DID is configured to
call an extension that belongs to my sipX box (Polycom phones), I cannot
transfer that call to any other internal extension or park that call. The
same is true if a call comes from another PBX (CUCM) - both gateways are
configured as "Unmanaged Gateway"
If I call out from sipX to the PSTN (there is also a trunk configured with
Vitelity on sipX) I can place that outbound call in park orbit or transfer
them to another internal extension.
I'm more familiar with troubleshooting Asterisk than sipX, so consider me
a newbie in this regard. I will gladly post configs / dump log files upon
request once I know what I'm looking for.
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

[Chris Parker]

The call volume is going to be very low. If I understand this correctly, I would create a trunk under Gateways in sipX for my Asterisk system and create the other end in Asterisk accordingly, rather than calling it an Unmanaged Gateway.
And to answer another question, yes the sipX and Asterisk system are on the same subnet whereas CUCM is in a different subnet but has unrestricted access to that subnet.

An unmanaged gateway is just that. Can I assume that the address for both systems are on the same subnet? Unmanaged gateways would assume that the other and knows how to handle the SIP REFER method.
Asterisk as a sip trunking system is not exactly compliant.
If REFER is not supported, then the media needs to be anchored by sipx once it accepts the call and hold the REFER internally, at which point you would setup a manual sip TRUNK not a gateway.

[Richard Bruce]

I am currently working with a Digium gateway that runs on Asterisk with the
same symptoms. For call transfer, I was able to create a route on the
gateway to take any extension number in my dial plan and send it back out to
the Sipxecs system. I have not been able to overcome the Park issue.

Richard Bruce
Dimensional Communications
7915 S. Emerson Ave, Suite 131
Indianapolis, IN 46237
[phone]- office
[phone] - cell
-----Original Message-----
From: sipx-users-***@list.sipfoundry.org
[mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Chris Parker
Sent: Tuesday, November 06, 2012 2:40 PM
To: sipx-***@list.sipfoundry.org
Subject: Re: [sipx-users] Cannot Xfer Calls Received from Unmanaged Trunks

The call volume is going to be very low. If I understand this correctly, I
would create a trunk under Gateways in sipX for my Asterisk system and
create the other end in Asterisk accordingly, rather than calling it an
Unmanaged Gateway.
And to answer another question, yes the sipX and Asterisk system are on the
same subnet whereas CUCM is in a different subnet but has unrestricted
access to that subnet.

Post by Tony Graziano
An unmanaged gateway is just that. Can I assume that the address for both

systems are on the same subnet? Unmanaged gateways would assume that the
other and knows how to handle the SIP REFER method.

Post by Tony Graziano
Asterisk as a sip trunking system is not exactly compliant.
If REFER is not supported, then the media needs to be anchored by sipx

once it accepts the call and hold the REFER internally, at which point you
would setup a manual sip TRUNK not a gateway.

[Chris Parker]

Some development on this topic...
Watching the SIP debug in Asterisk, I see that when I try to transfer the call it actually asks Asterisk to dial the target extension and Asterisk has no clue how to deal with that since it owns the 1xxx group while sipX owns 2xxx. If I put in a line that says to send _20xx back to sipX it works! Although due to how hacky this is, calls sent to Park are lost forever and all transfers are blind. At least this is progress in a way.
I tried to create a SIP trunk between Asterisk and sipX but it was sort of wonky; it couldn't call back into Asterisk to reach the 1xxx and the calls sent over from Asterisk still exhibited the same broken transfer - long story short I tried and failed somehow at SIP trunk, so it's back to an unmanaged gateway.

Post by Chris Parker
The call volume is going to be very low. If I understand this correctly, I would create a trunk under Gateways in sipX for my Asterisk system and create the other end in Asterisk accordingly, rather than calling it an Unmanaged Gateway.
And to answer another question, yes the sipX and Asterisk system are on the same subnet whereas CUCM is in a different subnet but has unrestricted access to that subnet.

An unmanaged gateway is just that. Can I assume that the address for both systems are on the same subnet? Unmanaged gateways would assume that the other and knows how to handle the SIP REFER method.
Asterisk as a sip trunking system is not exactly compliant.
If REFER is not supported, then the media needs to be anchored by sipx once it accepts the call and hold the REFER internally, at which point you would setup a manual sip TRUNK not a gateway.

[Josh Patten]

When using a SIP trunk you will need to have Asterisk point to port 5080.

Post by Chris Parker
Some development on this topic...
Watching the SIP debug in Asterisk, I see that when I try to transfer the
call it actually asks Asterisk to dial the target extension and Asterisk
has no clue how to deal with that since it owns the 1xxx group while sipX
owns 2xxx. If I put in a line that says to send _20xx back to sipX it
works! Although due to how hacky this is, calls sent to Park are lost
forever and all transfers are blind. At least this is progress in a way.
I tried to create a SIP trunk between Asterisk and sipX but it was sort of
wonky; it couldn't call back into Asterisk to reach the 1xxx and the calls
sent over from Asterisk still exhibited the same broken transfer - long
story short I tried and failed somehow at SIP trunk, so it's back to an
unmanaged gateway.

Post by Chris Parker
The call volume is going to be very low. If I understand this correctly,

I would create a trunk under Gateways in sipX for my Asterisk system and
create the other end in Asterisk accordingly, rather than calling it an
Unmanaged Gateway.

Post by Chris Parker
And to answer another question, yes the sipX and Asterisk system are on

the same subnet whereas CUCM is in a different subnet but has unrestricted
access to that subnet.

Post by Chris Parker

Post by Tony Graziano
An unmanaged gateway is just that. Can I assume that the address for

both systems are on the same subnet? Unmanaged gateways would assume that
the other and knows how to handle the SIP REFER method.

Post by Chris Parker

Post by Tony Graziano
Asterisk as a sip trunking system is not exactly compliant.
If REFER is not supported, then the media needs to be anchored by sipx

once it accepts the call and hold the REFER internally, at which point you
would setup a manual sip TRUNK not a gateway.
_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

[Chris Parker]

That part seemed to work, but I kept getting sipx not being found as a peer, even though my context in sip.conf was [sipx]

Post by Josh Patten
When using a SIP trunk you will need to have Asterisk point to port 5080.

Post by Chris Parker
Some development on this topic...
Watching the SIP debug in Asterisk, I see that when I try to transfer the call it actually asks Asterisk to dial the target extension and Asterisk has no clue how to deal with that since it owns the 1xxx group while sipX owns 2xxx. If I put in a line that says to send _20xx back to sipX it works! Although due to how hacky this is, calls sent to Park are lost forever and all transfers are blind. At least this is progress in a way.
I tried to create a SIP trunk between Asterisk and sipX but it was sort of wonky; it couldn't call back into Asterisk to reach the 1xxx and the calls sent over from Asterisk still exhibited the same broken transfer - long story short I tried and failed somehow at SIP trunk, so it's back to an unmanaged gateway.

Post by Chris Parker
The call volume is going to be very low. If I understand this correctly, I would create a trunk under Gateways in sipX for my Asterisk system and create the other end in Asterisk accordingly, rather than calling it an Unmanaged Gateway.
And to answer another question, yes the sipX and Asterisk system are on the same subnet whereas CUCM is in a different subnet but has unrestricted access to that subnet.

An unmanaged gateway is just that. Can I assume that the address for both systems are on the same subnet? Unmanaged gateways would assume that the other and knows how to handle the SIP REFER method.
Asterisk as a sip trunking system is not exactly compliant.
If REFER is not supported, then the media needs to be anchored by sipx once it accepts the call and hold the REFER internally, at which point you would setup a manual sip TRUNK not a gateway.

_______________________________________________
sipx-users mailing list
List Archive: http://list.sipfoundry.org/archive/sipx-users/

--
Josh Patten
eZuce
Solutions Architect
O.978-296-1005 X2050
M.979-574-5699
http://www.ezuce.com
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