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:
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.
nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 sofia/nat/ 5035557777@69.64.180.77:5060 [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 sofia/nat/5035557777@69.64.180.77 :5060 [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 sip:In-2061234567@mydomain.com:5070 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 (sofia/nat/5035557777@69.64.180.77 :5060) Ended 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/nat/5035557777@69.64.180.77 :5060 [CS_HANGUP] Any thoughts on what I'm doing wrong would be appreciated.
Kurt _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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:
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:
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:
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.
nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 sofia/nat/5035557777@69.64.180.77:5060 [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 sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 (sofia/nat/5035557777@69.64.180.77:5060) Ended 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] Any thoughts on what I'm doing wrong would be appreciated.
Kurt _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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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:
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:
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:
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:
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.
nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 sofia/nat/5035557777@69.64.180.77:5060 [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 sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 (sofia/nat/5035557777@69.64.180.77:5060) Ended 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] Any thoughts on what I'm doing wrong would be appreciated.
Kurt _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Could you please post a FS debug also with sip debug enabled on these event? Kurt Marasco wrote:
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$"> <action application="transfer" data="1001 XML default"/> </condition> </extension>
Thanks, Kurt
Josip Djuricic wrote:
Hi there,
if I'm not mistaking (if I am Brian or someone else will tell), you can do it from the public.xml
Example: <extension name="name_of_incoming_extension"> <condition field="destination_number" expression="^(incoming_extension_number_match)$"> <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:
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:
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:
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.
nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 sofia/nat/5035557777@69.64.180.77:5060 [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 sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 (sofia/nat/5035557777@69.64.180.77:5060) Ended 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP]
Any thoughts on what I'm doing wrong would be appreciated.
Kurt _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Here's the FS debug from the console: nta: received INVITE sip:In-2061234567@mydomain.com:5080 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 sofia/outbound/5031234567@67.55.341.56:5060 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [5031234567@67.55.341.56:5060] 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: [In-2061234567@mydomain.com:5080] variable_sip_req_host: [mydomain.com] variable_sip_to_user: [In-2061234567] variable_sip_to_port: [5080] variable_sip_to_uri: [In-2061234567@mydomain.com:5080] variable_sip_to_host: [mydomain.com] variable_sip_contact_user: [5031234567] variable_sip_contact_port: [5060] variable_sip_contact_uri: [5031234567@67.55.341.56:5060] variable_sip_contact_host: [67.55.341.56] variable_channel_name: [sofia/outbound/5031234567@67.55.341.56:5060] variable_sip_call_id: [637c71185629eca135630a1d3c215d08@67.55.341.56] 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 sofia/outbound/5031234567@67.55.341.56:5060 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 (sofia/outbound/5031234567@67.55.341.56:5060) Ended 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/outbound/5031234567@67.55.341.56:5060 [CS_HANGUP] Thanks Josip Djuricic wrote:
Could you please post a FS debug also with sip debug enabled on these event?
Kurt Marasco wrote:
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:
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:
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:
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:
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.
> nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 > sofia/nat/5035557777@69.64.180.77:5060 > [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 > sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 > (sofia/nat/5035557777@69.64.180.77:5060) Ended > 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 > switch_core_session_thread() Close Channel > sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] > Any thoughts on what I'm doing wrong would be appreciated.
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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 <kmarasco@faithwork.org> wrote:
Here's the FS debug from the console:
nta: received INVITE sip:In-2061234567@mydomain.com:5080 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 sofia/outbound/5031234567@67.55.341.56:5060 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [5031234567@67.55.341.56:5060] 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: [In-2061234567@mydomain.com:5080] variable_sip_req_host: [mydomain.com] variable_sip_to_user: [In-2061234567] variable_sip_to_port: [5080] variable_sip_to_uri: [In-2061234567@mydomain.com:5080] variable_sip_to_host: [mydomain.com] variable_sip_contact_user: [5031234567] variable_sip_contact_port: [5060] variable_sip_contact_uri: [5031234567@67.55.341.56:5060] variable_sip_contact_host: [67.55.341.56] variable_channel_name: [sofia/outbound/5031234567@67.55.341.56:5060] variable_sip_call_id: [637c71185629eca135630a1d3c215d08@67.55.341.56] 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 sofia/outbound/5031234567@67.55.341.56:5060 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 (sofia/outbound/5031234567@67.55.341.56:5060) Ended 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/outbound/5031234567@67.55.341.56:5060 [CS_HANGUP]
Thanks
Josip Djuricic wrote:
Could you please post a FS debug also with sip debug enabled on these event?
Kurt Marasco wrote:
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 $"> <action application="transfer" data="1001 XML default"/> </condition> </extension>
Thanks, Kurt
Josip Djuricic wrote:
Hi there,
if I'm not mistaking (if I am Brian or someone else will tell), you can do it from the public.xml
Example: <extension name="name_of_incoming_extension"> <condition field="destination_number" expression="^(incoming_extension_number_match)$"> <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:
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:
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:
> 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. > > >> nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 >> sofia/nat/5035557777@69.64.180.77:5060 >> [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 >> sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 >> (sofia/nat/5035557777@69.64.180.77:5060) Ended >> 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 >> switch_core_session_thread() Close Channel >> sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] >> > Any thoughts on what I'm doing wrong would be appreciated. > > Kurt > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > <mailto:Freeswitch-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
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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: [$1]) and not the contents of the variable. This field shows 1001 when I hard code the final extension. I found an error in my extension in default xml relating to where the condition is closed, but I fixed it and still do not get a match. public.xml contains the following: <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="public"> <extension name="unloop"> <condition field="${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:
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 <kmarasco@faithwork.org> wrote:
Here's the FS debug from the console:
nta: received INVITE sip:In-2061234567@mydomain.com:5080 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 sofia/outbound/5031234567@67.55.341.56:5060 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [5031234567@67.55.341.56:5060] 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: [In-2061234567@mydomain.com:5080] variable_sip_req_host: [mydomain.com] variable_sip_to_user: [In-2061234567] variable_sip_to_port: [5080] variable_sip_to_uri: [In-2061234567@mydomain.com:5080] variable_sip_to_host: [mydomain.com] variable_sip_contact_user: [5031234567] variable_sip_contact_port: [5060] variable_sip_contact_uri: [5031234567@67.55.341.56:5060] variable_sip_contact_host: [67.55.341.56] variable_channel_name: [sofia/outbound/5031234567@67.55.341.56:5060] variable_sip_call_id: [637c71185629eca135630a1d3c215d08@67.55.341.56] 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 sofia/outbound/5031234567@67.55.341.56:5060 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 (sofia/outbound/5031234567@67.55.341.56:5060) Ended 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/outbound/5031234567@67.55.341.56:5060 [CS_HANGUP]
Thanks
Josip Djuricic wrote:
Could you please post a FS debug also with sip debug enabled on these event?
Kurt Marasco wrote:
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 $"> <action application="transfer" data="1001 XML default"/> </condition> </extension>
Thanks, Kurt
Josip Djuricic wrote:
Hi there,
if I'm not mistaking (if I am Brian or someone else will tell), you can do it from the public.xml
Example: <extension name="name_of_incoming_extension"> <condition field="destination_number" expression="^(incoming_extension_number_match)$"> <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:
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:
> 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: > > > >> 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. >> >> >> >>> nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 >>> sofia/nat/5035557777@69.64.180.77:5060 >>> [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 >>> sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 >>> (sofia/nat/5035557777@69.64.180.77:5060) Ended >>> 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 >>> switch_core_session_thread() Close Channel >>> sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] >>> >>> >> Any thoughts on what I'm doing wrong would be appreciated. >> >> Kurt >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> <mailto:Freeswitch-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> --- --- ------------------------------------------------------------------
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Kurt, In you excerpt below you're missing brackets in your regex that were present earlier. This will set the value of '$1'. e.g. it needs to be "^(In-2061234567)$" for $1 to be set to In-2061234567. I've tried your dialplan below with brackets inserted and different default.xml actions and it seems to be fine. Also can you try testing by substituting the actual string you're expecting to work? e.g. rather than 1001 hard code In-2061234567. thanks, - Dale On Sat, Mar 15, 2008 at 01:31:15AM -0700, Kurt Marasco wrote:
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: [$1]) and not the contents of the variable. This field shows 1001 when I hard code the final extension.
I found an error in my extension in default xml relating to where the condition is closed, but I fixed it and still do not get a match.
public.xml contains the following: <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="public">
<extension name="unloop"> <condition field="${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:
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 <kmarasco@faithwork.org> wrote:
Here's the FS debug from the console:
nta: received INVITE sip:In-2061234567@mydomain.com:5080 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 sofia/outbound/5031234567@67.55.341.56:5060 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [5031234567@67.55.341.56:5060] 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: [In-2061234567@mydomain.com:5080] variable_sip_req_host: [mydomain.com] variable_sip_to_user: [In-2061234567] variable_sip_to_port: [5080] variable_sip_to_uri: [In-2061234567@mydomain.com:5080] variable_sip_to_host: [mydomain.com] variable_sip_contact_user: [5031234567] variable_sip_contact_port: [5060] variable_sip_contact_uri: [5031234567@67.55.341.56:5060] variable_sip_contact_host: [67.55.341.56] variable_channel_name: [sofia/outbound/5031234567@67.55.341.56:5060] variable_sip_call_id: [637c71185629eca135630a1d3c215d08@67.55.341.56] 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 sofia/outbound/5031234567@67.55.341.56:5060 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 (sofia/outbound/5031234567@67.55.341.56:5060) Ended 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/outbound/5031234567@67.55.341.56:5060 [CS_HANGUP]
Thanks
Josip Djuricic wrote:
Could you please post a FS debug also with sip debug enabled on these event?
Kurt Marasco wrote:
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 $"> <action application="transfer" data="1001 XML default"/> </condition> </extension>
Thanks, Kurt
Josip Djuricic wrote:
Hi there,
if I'm not mistaking (if I am Brian or someone else will tell), you can do it from the public.xml
Example: <extension name="name_of_incoming_extension"> <condition field="destination_number" expression="^(incoming_extension_number_match)$"> <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:
> 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: > > >> 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: >> >> >> >>> 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. >>> >>> >>> >>>> nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 >>>> sofia/nat/5035557777@69.64.180.77:5060 >>>> [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 >>>> sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 >>>> (sofia/nat/5035557777@69.64.180.77:5060) Ended >>>> 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 >>>> switch_core_session_thread() Close Channel >>>> sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] >>>> >>>> >>> Any thoughts on what I'm doing wrong would be appreciated. >>> >>> Kurt >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users@lists.freeswitch.org >>> <mailto:Freeswitch-users@lists.freeswitch.org> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> > --- > --- > ------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > --- --- ------------------------------------------------------------------
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Thanks Dale, I now have a working inbound did and outbound calling via a voip provider. I did try hard coding the string "In-2061234567" in public.xml, but I was still missing the parenthesis in default.xml so the same pcre error was stopping it from working. By fixing the pattern in both places , per your suggestion, I am successful:) Thanks to all who helped. I'm sure that I'll see you here a bit more. Your responsiveness is apprecated!! Kurt freeswitch@dalethatcher.com wrote:
Kurt,
In you excerpt below you're missing brackets in your regex that were present earlier. This will set the value of '$1'. e.g. it needs to be "^(In-2061234567)$" for $1 to be set to In-2061234567.
I've tried your dialplan below with brackets inserted and different default.xml actions and it seems to be fine.
Also can you try testing by substituting the actual string you're expecting to work? e.g. rather than 1001 hard code In-2061234567.
thanks,
- Dale
On Sat, Mar 15, 2008 at 01:31:15AM -0700, Kurt Marasco wrote:
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: [$1]) and not the contents of the variable. This field shows 1001 when I hard code the final extension.
I found an error in my extension in default xml relating to where the condition is closed, but I fixed it and still do not get a match.
public.xml contains the following: <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="public">
<extension name="unloop"> <condition field="${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:
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 <kmarasco@faithwork.org> wrote:
Here's the FS debug from the console:
nta: received INVITE sip:In-2061234567@mydomain.com:5080 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 sofia/outbound/5031234567@67.55.341.56:5060 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [5031234567@67.55.341.56:5060] 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: [In-2061234567@mydomain.com:5080] variable_sip_req_host: [mydomain.com] variable_sip_to_user: [In-2061234567] variable_sip_to_port: [5080] variable_sip_to_uri: [In-2061234567@mydomain.com:5080] variable_sip_to_host: [mydomain.com] variable_sip_contact_user: [5031234567] variable_sip_contact_port: [5060] variable_sip_contact_uri: [5031234567@67.55.341.56:5060] variable_sip_contact_host: [67.55.341.56] variable_channel_name: [sofia/outbound/5031234567@67.55.341.56:5060] variable_sip_call_id: [637c71185629eca135630a1d3c215d08@67.55.341.56] 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 sofia/outbound/5031234567@67.55.341.56:5060 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 (sofia/outbound/5031234567@67.55.341.56:5060) Ended 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/outbound/5031234567@67.55.341.56:5060 [CS_HANGUP]
Thanks
Josip Djuricic wrote:
Could you please post a FS debug also with sip debug enabled on these event?
Kurt Marasco wrote:
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 $"> <action application="transfer" data="1001 XML default"/> </condition> </extension>
Thanks, Kurt
Josip Djuricic wrote:
> Hi there, > > if I'm not mistaking (if I am Brian or someone else will tell), you > can do it from the public.xml > > Example: > <extension name="name_of_incoming_extension"> > <condition field="destination_number" > expression="^(incoming_extension_number_match)$"> > <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: > > > >> 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: >> >> >> >>> 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: >>> >>> >>> >>> >>>> 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. >>>> >>>> >>>> >>>> >>>>> nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 >>>>> sofia/nat/5035557777@69.64.180.77:5060 >>>>> [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 >>>>> sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 >>>>> (sofia/nat/5035557777@69.64.180.77:5060) Ended >>>>> 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 >>>>> switch_core_session_thread() Close Channel >>>>> sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] >>>>> >>>>> >>>>> >>>> Any thoughts on what I'm doing wrong would be appreciated. >>>> >>>> Kurt >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users@lists.freeswitch.org >>>> <mailto:Freeswitch-users@lists.freeswitch.org> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >> --- >> --- >> ------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > --- > --- > ------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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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:
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: [$1]) and not the contents of the variable. This field shows 1001 when I hard code the final extension.
I found an error in my extension in default xml relating to where the condition is closed, but I fixed it and still do not get a match.
public.xml contains the following: <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="public">
<extension name="unloop"> <condition field="${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:
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 <kmarasco@faithwork.org> wrote:
Here's the FS debug from the console:
nta: received INVITE sip:In-2061234567@mydomain.com:5080 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 sofia/outbound/5031234567@67.55.341.56:5060 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [5031234567@67.55.341.56:5060] 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: [In-2061234567@mydomain.com:5080] variable_sip_req_host: [mydomain.com] variable_sip_to_user: [In-2061234567] variable_sip_to_port: [5080] variable_sip_to_uri: [In-2061234567@mydomain.com:5080] variable_sip_to_host: [mydomain.com] variable_sip_contact_user: [5031234567] variable_sip_contact_port: [5060] variable_sip_contact_uri: [5031234567@67.55.341.56:5060] variable_sip_contact_host: [67.55.341.56] variable_channel_name: [sofia/outbound/5031234567@67.55.341.56:5060] variable_sip_call_id: [637c71185629eca135630a1d3c215d08@67.55.341.56] 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 sofia/outbound/5031234567@67.55.341.56:5060 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 (sofia/outbound/5031234567@67.55.341.56:5060) Ended 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/outbound/5031234567@67.55.341.56:5060 [CS_HANGUP]
Thanks
Josip Djuricic wrote:
Could you please post a FS debug also with sip debug enabled on these event?
Kurt Marasco wrote:
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 $"> <action application="transfer" data="1001 XML default"/> </condition> </extension>
Thanks, Kurt
Josip Djuricic wrote:
Hi there,
if I'm not mistaking (if I am Brian or someone else will tell), you can do it from the public.xml
Example: <extension name="name_of_incoming_extension"> <condition field="destination_number" expression="^(incoming_extension_number_match)$"> <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:
> 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: > > > >> 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: >> >> >> >> >>> 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. >>> >>> >>> >>> >>>> nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 >>>> sofia/nat/5035557777@69.64.180.77:5060 >>>> [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 >>>> sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 >>>> (sofia/nat/5035557777@69.64.180.77:5060) Ended >>>> 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 >>>> switch_core_session_thread() Close Channel >>>> sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] >>>> >>>> >>>> >>> Any thoughts on what I'm doing wrong would be appreciated. >>> >>> Kurt >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users@lists.freeswitch.org >>> <mailto:Freeswitch-users@lists.freeswitch.org> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> > --- > --- > ------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > --- --- ------------------------------------------------------------------
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We must have been typing at the same time. You are exactly right. Just resolved it a bit ago. Thank you. Josip Djuricic wrote:
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:
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: [$1]) and not the contents of the variable. This field shows 1001 when I hard code the final extension.
I found an error in my extension in default xml relating to where the condition is closed, but I fixed it and still do not get a match.
public.xml contains the following: <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="public">
<extension name="unloop"> <condition field="${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:
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 <kmarasco@faithwork.org> wrote:
Here's the FS debug from the console:
nta: received INVITE sip:In-2061234567@mydomain.com:5080 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 sofia/outbound/5031234567@67.55.341.56:5060 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [5031234567@67.55.341.56:5060] 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: [In-2061234567@mydomain.com:5080] variable_sip_req_host: [mydomain.com] variable_sip_to_user: [In-2061234567] variable_sip_to_port: [5080] variable_sip_to_uri: [In-2061234567@mydomain.com:5080] variable_sip_to_host: [mydomain.com] variable_sip_contact_user: [5031234567] variable_sip_contact_port: [5060] variable_sip_contact_uri: [5031234567@67.55.341.56:5060] variable_sip_contact_host: [67.55.341.56] variable_channel_name: [sofia/outbound/5031234567@67.55.341.56:5060] variable_sip_call_id: [637c71185629eca135630a1d3c215d08@67.55.341.56] 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 sofia/outbound/5031234567@67.55.341.56:5060 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 (sofia/outbound/5031234567@67.55.341.56:5060) Ended 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/outbound/5031234567@67.55.341.56:5060 [CS_HANGUP]
Thanks
Josip Djuricic wrote:
Could you please post a FS debug also with sip debug enabled on these event?
Kurt Marasco wrote:
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 $"> <action application="transfer" data="1001 XML default"/> </condition> </extension>
Thanks, Kurt
Josip Djuricic wrote:
> Hi there, > > if I'm not mistaking (if I am Brian or someone else will tell), you > can do it from the public.xml > > Example: > <extension name="name_of_incoming_extension"> > <condition field="destination_number" > expression="^(incoming_extension_number_match)$"> > <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: > > > >> 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: >> >> >> >>> 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: >>> >>> >>> >>> >>>> 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. >>>> >>>> >>>> >>>> >>>>> nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 >>>>> sofia/nat/5035557777@69.64.180.77:5060 >>>>> [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 >>>>> sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 >>>>> (sofia/nat/5035557777@69.64.180.77:5060) Ended >>>>> 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 >>>>> switch_core_session_thread() Close Channel >>>>> sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] >>>>> >>>>> >>>>> >>>> Any thoughts on what I'm doing wrong would be appreciated. >>>> >>>> Kurt >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users@lists.freeswitch.org >>>> <mailto:Freeswitch-users@lists.freeswitch.org> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >> --- >> --- >> ------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > --- > --- > ------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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yeah m8 :) nice to hear you resolved it Kurt Marasco wrote:
We must have been typing at the same time. You are exactly right. Just resolved it a bit ago. Thank you.
Josip Djuricic wrote:
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:
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: [$1]) and not the contents of the variable. This field shows 1001 when I hard code the final extension.
I found an error in my extension in default xml relating to where the condition is closed, but I fixed it and still do not get a match.
public.xml contains the following: <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> <include> <context name="public">
<extension name="unloop"> <condition field="${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:
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 <kmarasco@faithwork.org> wrote:
Here's the FS debug from the console:
nta: received INVITE sip:In-2061234567@mydomain.com:5080 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 sofia/outbound/5031234567@67.55.341.56:5060 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [sofia/outbound/5031234567@67.55.341.56:5060] 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: [5031234567@67.55.341.56:5060] 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: [In-2061234567@mydomain.com:5080] variable_sip_req_host: [mydomain.com] variable_sip_to_user: [In-2061234567] variable_sip_to_port: [5080] variable_sip_to_uri: [In-2061234567@mydomain.com:5080] variable_sip_to_host: [mydomain.com] variable_sip_contact_user: [5031234567] variable_sip_contact_port: [5060] variable_sip_contact_uri: [5031234567@67.55.341.56:5060] variable_sip_contact_host: [67.55.341.56] variable_channel_name: [sofia/outbound/5031234567@67.55.341.56:5060] variable_sip_call_id: [637c71185629eca135630a1d3c215d08@67.55.341.56] 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 sofia/outbound/5031234567@67.55.341.56:5060 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 sofia/outbound/5031234567@67.55.341.56:5060 [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 (sofia/outbound/5031234567@67.55.341.56:5060) Ended 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/outbound/5031234567@67.55.341.56:5060 [CS_HANGUP]
Thanks
Josip Djuricic wrote:
Could you please post a FS debug also with sip debug enabled on these event?
Kurt Marasco wrote:
> 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 > $"> > <action application="transfer" data="1001 XML default"/> > </condition> > </extension> > > Thanks, > Kurt > > Josip Djuricic wrote: > > > > >> Hi there, >> >> if I'm not mistaking (if I am Brian or someone else will tell), you >> can do it from the public.xml >> >> Example: >> <extension name="name_of_incoming_extension"> >> <condition field="destination_number" >> expression="^(incoming_extension_number_match)$"> >> <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: >> >> >> >> >>> 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: >>> >>> >>> >>> >>>> 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: >>>> >>>> >>>> >>>> >>>> >>>>> 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. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 >>>>>> sofia/nat/5035557777@69.64.180.77:5060 >>>>>> [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 >>>>>> sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 >>>>>> (sofia/nat/5035557777@69.64.180.77:5060) Ended >>>>>> 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 >>>>>> switch_core_session_thread() Close Channel >>>>>> sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] >>>>>> >>>>>> >>>>>> >>>>>> >>>>> Any thoughts on what I'm doing wrong would be appreciated. >>>>> >>>>> Kurt >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users@lists.freeswitch.org >>>>> <mailto:Freeswitch-users@lists.freeswitch.org> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >> --- >> --- >> ------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > --- ---------------------------------------------------------------------
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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:
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:
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.
nta: received INVITE sip:In-2061234567@mydomain.com:5070 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 sofia/nat/5035557777@69.64.180.77:5060 [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 sofia/nat/5035557777@69.64.180.77:5060 [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 sip:In-2061234567@mydomain.com:5070 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 (sofia/nat/5035557777@69.64.180.77:5060) Ended 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 switch_core_session_thread() Close Channel sofia/nat/5035557777@69.64.180.77:5060 [CS_HANGUP] Any thoughts on what I'm doing wrong would be appreciated.
Kurt _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (5)
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Brian West
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freeswitch＠dalethatcher.com
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Josip Djuricic
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Kurt Marasco
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Michael Jerris