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yeah m8 :)<br>
<br>
nice to hear you resolved it<br>
<br>
Kurt Marasco wrote:
<blockquote cite="mid:47DBB5C4.2090407@faithwork.org" type="cite">
<pre wrap="">We must have been typing at the same time. You are exactly right. Just
resolved it a bit ago.
Thank you.
Josip Djuricic wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Kurt if I see it correctly ain't you missing brackets?
<condition field="destination_number" expression="^(In-2061234567)$">
otherwise $1 has almost no meaning, cause it won't be filled with In-2061234567
Josip
Kurt Marasco wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Michael, you are absolutely correct that it is matching on the enum
extension in the default profile. When I comment out the enum extension
I get no match.
So I know that the transfer is being made from public.xml to
default.xml. The question is why am I not getting a match? And is there
any way to see what's contained in the $1 that is being passed from
public,xml to default.xml??? The debug shows $1
(Caller-Destination-Number: [$1]) and not the contents of the variable.
This field shows 1001 when I hard code the final extension.
I found an error in my extension in default xml relating to where the
condition is closed, but I fixed it and still do not get a match.
public.xml contains the following:
<!-- <a class="moz-txt-link-freetext" href="http://wiki.freeswitch.org/wiki/Dialplan_XML">http://wiki.freeswitch.org/wiki/Dialplan_XML</a> -->
<include>
<context name="public">
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<extension name="public_did2">
<condition field="destination_number" expression="^In-2061234567$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
</context>
</include>
The extension public_did2 differs from my working scenario only in that
I hard coded the destination extension 1001 in place of $1.
My extension to match in default.xml is listed prior to the enum
extension that is being matched as follows:
<extension name="In-2061234567">
<condition field="destination_number" expression="^In-2061234567$">
<!-- <action application="ringback" /> -->
<action application="set" data="call_timeout=20"/>
<action application="bridge" data="sofia/default/1001%$${domain}"/>
<action application="javascript"
data="/usr/local/freeswitch/scripts/answermachine.js"/>
</condition>
</extension>
My understanding is that $1 should contain whatever was matched in
public.xml, which is the string "In-2061234567". My condition expression
in default.xml is "^In-2061234567$", so I don't where else to look at
this point. My probable conclusions are that I have a syntax error or
that $1 is passing something other than expected. Extension
"In-2061234567" was not in the directory because it is not a physical
endpoint. I added it to the directory thinking that this might be an
authentication issue, but the result is the same even if "In-2061234567"
is in the directory (No sip phone is registering to "In-2061234567" as
my intent is to transfer/bridge to 1001.
Kurt
Michael Jerris wrote:
</pre>
<blockquote type="cite">
<pre wrap="">This is a different extension that is transfering to the enum dialplan
not XML like the extension you pasted. Chek your conditions to see
why its matching the wrong ext.
Mike
On Mar 14, 2008, at 6:31 PM, Kurt Marasco <a class="moz-txt-link-rfc2396E" href="mailto:kmarasco@faithwork.org"><kmarasco@faithwork.org></a>
wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Here's the FS debug from the console:
nta: received INVITE <a class="moz-txt-link-freetext" href="sip:In-2061234567@mydomain.com:5080">sip:In-2061234567@mydomain.com:5080</a> SIP/2.0
(CSeq 102)
nta: INVITE (102) going to a default leg
nua(0x81c22c0): adding session usage
nta: sent 100 Trying for INVITE (102)
nua(0x81c22c0): call state changed: init -> received, received offer
2008-03-14 18:18:40 [NOTICE] switch_channel.c:522
switch_channel_set_name() New Chan
<a class="moz-txt-link-abbreviated" href="mailto:sofia/outbound/5031234567@67.55.341.56:5060">sofia/outbound/5031234567@67.55.341.56:5060</a>
[b9608e95-8a80-49ba-b3c5-883432c4bcb2]
2008-03-14 18:18:40 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
Processing PORTLAND OR->In-2061234567!
2008-03-14 18:18:40 [NOTICE] switch_ivr.c:924
switch_ivr_session_transfer() Transfer
<a class="moz-txt-link-abbreviated" href="mailto:sofia/outbound/5031234567@67.55.341.56:5060">sofia/outbound/5031234567@67.55.341.56:5060</a> to XML[$1@default]
2008-03-14 18:18:40 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
Processing PORTLAND OR->$1!
2008-03-14 18:18:40 [INFO] mod_dptools.c:601 info_function()
CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [<a class="moz-txt-link-abbreviated" href="mailto:sofia/outbound/5031234567@67.55.341.56:5060">sofia/outbound/5031234567@67.55.341.56:5060</a>]
Unique-ID: [b9608e95-8a80-49ba-b3c5-883432c4bcb2]
Call-Direction: [inbound]
Answer-State: [ringing]
Channel-Read-Codec-Name: [PCMU]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [PCMU]
Channel-Write-Codec-Rate: [8000]
Caller-Username: [5031234567]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [PORTLAND OR]
Caller-Caller-ID-Number: [5031234567]
Caller-Network-Addr: [67.55.341.56]
Caller-Destination-Number: [$1]
Caller-Unique-ID: [b9608e95-8a80-49ba-b3c5-883432c4bcb2]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [In-2061234567]
Caller-Channel-Name: [<a class="moz-txt-link-abbreviated" href="mailto:sofia/outbound/5031234567@67.55.341.56:5060">sofia/outbound/5031234567@67.55.341.56:5060</a>]
Caller-Channel-Created-Time: [1205543920601430]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [yes]
Caller-Privacy-Hide-Name: [no]
Caller-Privacy-Hide-Number: [no]
variable_sip_from_user: [5031234567]
variable_sip_from_port: [5060]
variable_sip_from_uri: [<a class="moz-txt-link-abbreviated" href="mailto:5031234567@67.55.341.56:5060">5031234567@67.55.341.56:5060</a>]
variable_sip_from_host: [67.55.341.56]
variable_sip_from_user_stripped: [5031234567]
variable_sip_from_tag: [as6a5e4b5c]
variable_sofia_profile_name: [outbound]
variable_sofia_profile_domain_name: [outbound]
variable_sip_req_user: [In-2061234567]
variable_sip_req_port: [5080]
variable_sip_req_uri: [<a class="moz-txt-link-abbreviated" href="mailto:In-2061234567@mydomain.com:5080">In-2061234567@mydomain.com:5080</a>]
variable_sip_req_host: [mydomain.com]
variable_sip_to_user: [In-2061234567]
variable_sip_to_port: [5080]
variable_sip_to_uri: [<a class="moz-txt-link-abbreviated" href="mailto:In-2061234567@mydomain.com:5080">In-2061234567@mydomain.com:5080</a>]
variable_sip_to_host: [mydomain.com]
variable_sip_contact_user: [5031234567]
variable_sip_contact_port: [5060]
variable_sip_contact_uri: [<a class="moz-txt-link-abbreviated" href="mailto:5031234567@67.55.341.56:5060">5031234567@67.55.341.56:5060</a>]
variable_sip_contact_host: [67.55.341.56]
variable_channel_name: [<a class="moz-txt-link-abbreviated" href="mailto:sofia/outbound/5031234567@67.55.341.56:5060">sofia/outbound/5031234567@67.55.341.56:5060</a>]
variable_sip_call_id: [<a class="moz-txt-link-abbreviated" href="mailto:637c71185629eca135630a1d3c215d08@67.55.341.56">637c71185629eca135630a1d3c215d08@67.55.341.56</a>]
variable_sip_user_agent: [Asterisk PBX]
variable_sip_via_host: [67.55.341.56]
variable_sip_via_port: [5060]
variable_sip_via_rport: [5060]
variable_max_forwards: [70]
variable_switch_r_sdp: [v=0
o=root 25187 25187 IN IP4 67.55.341.56
s=session
c=IN IP4 67.55.341.56
t=0 0
m=audio 12740 RTP/AVP 0 8 3 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
]
variable_remote_media_ip: [67.55.341.56]
variable_remote_media_port: [12740]
variable_read_codec: [PCMU]
variable_read_rate: [8000]
variable_write_codec: [PCMU]
variable_write_rate: [8000]
variable_endpoint_disposition: [RECEIVED]
variable_use_profile: [default]
variable_numbering_plan: [US]
variable_default_gateway: [192.168.2.1]
variable_default_area_code: [509]
variable_user_name: [default]
variable_domain_name: [192.168.2.102]
2008-03-14 18:18:41 [NOTICE] switch_ivr.c:924
switch_ivr_session_transfer() Transfer
<a class="moz-txt-link-abbreviated" href="mailto:sofia/outbound/5031234567@67.55.341.56:5060">sofia/outbound/5031234567@67.55.341.56:5060</a> to enum[$1@default]
2008-03-14 18:18:41 [INFO] switch_core_state_machine.c:112
switch_core_standard_on_ring() No Route, Aborting
2008-03-14 18:18:41 [NOTICE] switch_core_state_machine.c:113
switch_core_standard_on_ring() Hangup
<a class="moz-txt-link-abbreviated" href="mailto:sofia/outbound/5031234567@67.55.341.56:5060">sofia/outbound/5031234567@67.55.341.56:5060</a> [CS_RING]
[NO_ROUTE_DESTINATION]
nta: sent 404 Not Found for INVITE (102)
nua(0x81c22c0): removing session usage
nua(0x81c22c0): call state changed: init -> terminated
2008-03-14 18:18:41 [NOTICE] switch_core_session.c:717
switch_core_session_thread() Session 2
(<a class="moz-txt-link-abbreviated" href="mailto:sofia/outbound/5031234567@67.55.341.56:5060">sofia/outbound/5031234567@67.55.341.56:5060</a>) Ended
2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719
switch_core_session_thread() Close Channel
<a class="moz-txt-link-abbreviated" href="mailto:sofia/outbound/5031234567@67.55.341.56:5060">sofia/outbound/5031234567@67.55.341.56:5060</a> [CS_HANGUP]
Thanks
Josip Djuricic wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Could you please post a FS debug also with sip debug enabled on these
event?
Kurt Marasco wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Thanks,
That's what I tried, but I ended up I hardcoding the extension.
When I
left it as $1, the console showed it being passed literally as $1
(But
maybe it was actually passing the contents of the variable). Since I
don't have a registered endpoint that matched my incoming DID, I was
trying to have the default dial plan handle the passed DID with no
luck.
It seemed to only want to transfer or bridge to a registered
endpoint
and not into the dial plan.
Perhaps my syntax in the default dial plan was wrong. I tried this
in
the public.xml:
<extension name="public_did2">
<condition field="destination_number" expression="^
(In-2061234567)$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
And this in the default dial plan:
<extension name="In-2061234567">
<condition field="destination_number"
expression="^In-2061234567$"/>
<action application="ringback" />
<action application="set" data="call_timeout=20"/>
<action application="bridge" data="sofia/default/1001%$$
{domain}"/>
<action application="javascript"
data="/usr/local/freeswitch/scripts/answermachine.js"/>
</extension>
The above fails, but below worked by itself in public.xml:
<extension name="public_did">
<condition field="destination_number" expression="^In-2061234567
$">
<action application="transfer" data="1001 XML default"/>
</condition>
</extension>
Thanks,
Kurt
Josip Djuricic wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hi there,
if I'm not mistaking (if I am Brian or someone else will tell), you
can do it from the public.xml
Example:
<extension name="name_of_incoming_extension">
<condition field="destination_number"
expression="^(incoming_extension_number_match)$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
If I'm not mistaking with transfer to XML default you do exactly
what
you wanna do.
Josip
Kurt Marasco wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Thanks Brian and Josip for your responses,
Brian's suggestion did the trick for me. I can both transfer and
bridge the call to a registered extension in the default dial
plan.
Not sure if If it makes sense to do this, but is there a way to
pass
the call into the default dial plan and have the default dial plan
process the sip invite. I'm able to send the incoming did to a
registered endpoint from (in the directory) but can't pass it
through
to the default and match on the original incoming did.
I'm still confused about what the nat profile does, because I'm
behind nat and am not using the nat profile, yet freeswitch
seems to
be working.
Brian West wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Kurt,
First off let me fill in a few blanks here.
Correct me if i'm wrong this looks like an inbound invite to port
5070 right? If so then you're not using the default config as it
was designed. (I did the bulk of the config)
Here is what you do. Have your IPKALL did hit your IP on port
5080
instead.. aka the outbound profile.
Then open up dialplan/public.xml and install an extension that
can
route to a registered endpoing. their is a 5551212 example in
there.
/b
On Mar 11, 2008, at 5:02 AM, Kurt Marasco wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hi I am testing FS and am currently working with the xml
dialplan.
I have FS behind a NAT router and have 2 soft phones
functioning on
another PC behind the router. I currently have working
conversations when dialing between the extensions set up on each
phone.
I am now trying to call one of the softphones via an IpKall
DID. I
have no problem making this work if I use wikipbx, but can't
make
it work using the xml dialplan, so clearly FS is working and my
configuration is the issue. I am currently sending the ipkall
sip
invite to port 5070, but have tried 5060 as well.
Here is the console output from FS when I dial my IpKall DID
from
my land line.
</pre>
<blockquote type="cite">
<pre wrap="">nta: received INVITE <a class="moz-txt-link-freetext" href="sip:In-2061234567@mydomain.com:5070">sip:In-2061234567@mydomain.com:5070</a> SIP/
2.0
(CSeq 102)
nta: INVITE (102) going to a default leg
nua(0x8117508): adding session usage
nta: sent 100 Trying for INVITE (102)
nua(0x8117508): call state changed: init -> received,
received offer
2008-03-11 02:25:31 [NOTICE] switch_channel.c:522
switch_channel_set_name() New Chan
<a class="moz-txt-link-abbreviated" href="mailto:sofia/nat/5035557777@69.64.180.77:5060">sofia/nat/5035557777@69.64.180.77:5060</a>
[53bb0a56-f059-483e-9e08-d583a9566255]
2008-03-11 02:25:32 [INFO] mod_dialplan_xml.c:222
dialplan_hunt()
Processing PORTLAND OR->In-2061234567!
*2008-03-11 02:25:32 [INFO] switch_core_state_machine.c:112
switch_core_standard_on_ring() No Route, Aborting*
2008-03-11 02:25:32 [NOTICE] switch_core_state_machine.c:113
switch_core_standard_on_ring() Hangup
<a class="moz-txt-link-abbreviated" href="mailto:sofia/nat/5035557777@69.64.180.77:5060">sofia/nat/5035557777@69.64.180.77:5060</a> [CS_RING]
[NO_ROUTE_DESTINATION]
nta: sent 404 Not Found for INVITE (102)
nua(0x8117508): removing session usage
nua(0x8117508): call state changed: init -> terminated
nta: received ACK <a class="moz-txt-link-freetext" href="sip:In-2061234567@mydomain.com:5070">sip:In-2061234567@mydomain.com:5070</a> SIP/2.0
(CSeq 102)
nta: ACK (102) is going to INVITE (102)
2008-03-11 02:25:32 [NOTICE] switch_core_session.c:717
switch_core_session_thread() Session 1
(<a class="moz-txt-link-abbreviated" href="mailto:sofia/nat/5035557777@69.64.180.77:5060">sofia/nat/5035557777@69.64.180.77:5060</a>) Ended
2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719
switch_core_session_thread() Close Channel
<a class="moz-txt-link-abbreviated" href="mailto:sofia/nat/5035557777@69.64.180.77:5060">sofia/nat/5035557777@69.64.180.77:5060</a> [CS_HANGUP]
</pre>
</blockquote>
<pre wrap="">Any thoughts on what I'm doing wrong would be appreciated.
Kurt
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<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
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<pre wrap=""><!---->
_______________________________________________
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