Setting Up an Sip Trunk without Authentication
Hi, I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address. I put <extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> in the default.xml under Dialplan When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above. I Appreciate any help you may offer. Dave Goodwin
Dave, If you are simply handling incoming calls then you will need to do two things: Add Velocity's IP addr to the domains section of acl.conf.xml Add an extension to the public context (conf/dialplan/public.xml) Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script. Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge. Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant. -MC On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote:
Hi,
I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.
I put
<extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension>
in the default.xml under Dialplan
When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.
What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.
I Appreciate any help you may offer.
Dave Goodwin
_______________________________________________ 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
Under "domains" in acl.conf.xml I added <node type="allow" value="xxx.xxx.xxx.xxx"/> x's are the IP address it's just a single address not a CIDR. And I created the file velocity_did.xml with the following code.. <include> <extension name="velocity_did"> <condition field="destination_number" expression="^(2064001950)$"> <action application="set" data="domain_name=$${domain}"/> <action application="answer"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> </include> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. Do I need to change the line .. <action application="set" data="domain_name=$${domain}"/> to the IP address? If so what syntax? They have no Domain Name. Thanks again for your help. Dave Goodwin ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 12:15 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication Dave, If you are simply handling incoming calls then you will need to do two things: Add Velocity's IP addr to the domains section of acl.conf.xml Add an extension to the public context (conf/dialplan/public.xml) Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script. Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge. Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant. -MC On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote: Hi, I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address. I put <extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> in the default.xml under Dialplan When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above. I Appreciate any help you may offer. Dave Goodwin _______________________________________________ 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
On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave@clancysystems.com> wrote:
Under "domains" in acl.conf.xml I added
<node type="allow" value="xxx.xxx.xxx.xxx"/>
The correct syntax is: <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
x's are the IP address it's just a single address not a CIDR.
And I created the file velocity_did.xml with the following code..
<include> <extension name="velocity_did">
<condition field="destination_number" expression="^(2064001950)$"> <action application="set" data="domain_name=$${domain}"/> <action application="answer"/>
<action application="lua" data="cme_ivr.lua channel_name"/>
</condition> </extension> </include> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.
Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that. For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them. -MC Do I need to change the line ..
<action application="set" data="domain_name=$${domain}"/> to the IP address? If so what syntax?
They have no Domain Name.
Thanks again for your help.
Dave Goodwin
----- Original Message ----- *From:* Michael Collins <msc@freeswitch.org> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Thursday, June 16, 2011 12:15 PM *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
Dave,
If you are simply handling incoming calls then you will need to do two things:
Add Velocity's IP addr to the domains section of acl.conf.xml Add an extension to the public context (conf/dialplan/public.xml)
Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.
Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.
Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.
-MC
On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote:
Hi,
I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.
I put
<extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension>
in the default.xml under Dialplan
When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.
What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.
I Appreciate any help you may offer.
Dave Goodwin
_______________________________________________ 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
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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
Yes, I am only testing inbound. When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox) I Get a busy signal and at that moment FreeSWITCH shows... "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP. If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP. ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 2:11 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave@clancysystems.com> wrote: Under "domains" in acl.conf.xml I added <node type="allow" value="xxx.xxx.xxx.xxx"/> The correct syntax is: <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/> x's are the IP address it's just a single address not a CIDR. And I created the file velocity_did.xml with the following code.. <include> <extension name="velocity_did"> <condition field="destination_number" expression="^(2064001950)$"> <action application="set" data="domain_name=$${domain}"/> <action application="answer"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> </include> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that. For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them. -MC Do I need to change the line .. <action application="set" data="domain_name=$${domain}"/> to the IP address? If so what syntax? They have no Domain Name. Thanks again for your help. Dave Goodwin ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 12:15 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication Dave, If you are simply handling incoming calls then you will need to do two things: Add Velocity's IP addr to the domains section of acl.conf.xml Add an extension to the public context (conf/dialplan/public.xml) Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script. Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge. Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant. -MC On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote: Hi, I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address. I put <extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> in the default.xml under Dialplan When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above. I Appreciate any help you may offer. Dave Goodwin _______________________________________________ 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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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
Did you reloadxml and restart the sofia profile after the changes? Also, you need to "reloadacl" after doing a change to acl.conf.xml. -MC On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave@clancysystems.com> wrote:
Yes, I am only testing inbound. When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox) I Get a busy signal and at that moment FreeSWITCH shows... "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP.
If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP.
----- Original Message ----- *From:* Michael Collins <msc@freeswitch.org> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Thursday, June 16, 2011 2:11 PM *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave@clancysystems.com> wrote:
Under "domains" in acl.conf.xml I added
<node type="allow" value="xxx.xxx.xxx.xxx"/>
The correct syntax is: <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
x's are the IP address it's just a single address not a CIDR.
And I created the file velocity_did.xml with the following code..
<include> <extension name="velocity_did">
<condition field="destination_number" expression="^(2064001950)$"> <action application="set" data="domain_name=$${domain}"/> <action application="answer"/>
<action application="lua" data="cme_ivr.lua channel_name"/>
</condition> </extension> </include> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.
Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.
For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them.
-MC
Do I need to change the line ..
<action application="set" data="domain_name=$${domain}"/> to the IP address? If so what syntax?
They have no Domain Name.
Thanks again for your help.
Dave Goodwin
----- Original Message ----- *From:* Michael Collins <msc@freeswitch.org> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Thursday, June 16, 2011 12:15 PM *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
Dave,
If you are simply handling incoming calls then you will need to do two things:
Add Velocity's IP addr to the domains section of acl.conf.xml Add an extension to the public context (conf/dialplan/public.xml)
Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.
Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.
Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.
-MC
On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote:
Hi,
I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.
I put
<extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension>
in the default.xml under Dialplan
When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.
What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.
I Appreciate any help you may offer.
Dave Goodwin
_______________________________________________ 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
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
I did both, and even restarted the service. same results. Velocity did a call capture on thier end and sent it to me. They say that FreeSWITCH is requiring proxy authentication and that that is the issue. SIP Status: 407 Proxy Authentication Required ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 2:46 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication Did you reloadxml and restart the sofia profile after the changes? Also, you need to "reloadacl" after doing a change to acl.conf.xml. -MC On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave@clancysystems.com> wrote: Yes, I am only testing inbound. When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox) I Get a busy signal and at that moment FreeSWITCH shows... "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP. If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP. ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 2:11 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave@clancysystems.com> wrote: Under "domains" in acl.conf.xml I added <node type="allow" value="xxx.xxx.xxx.xxx"/> The correct syntax is: <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/> x's are the IP address it's just a single address not a CIDR. And I created the file velocity_did.xml with the following code.. <include> <extension name="velocity_did"> <condition field="destination_number" expression="^(2064001950)$"> <action application="set" data="domain_name=$${domain}"/> <action application="answer"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> </include> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that. For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them. -MC Do I need to change the line .. <action application="set" data="domain_name=$${domain}"/> to the IP address? If so what syntax? They have no Domain Name. Thanks again for your help. Dave Goodwin ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 12:15 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication Dave, If you are simply handling incoming calls then you will need to do two things: Add Velocity's IP addr to the domains section of acl.conf.xml Add an extension to the public context (conf/dialplan/public.xml) Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script. Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge. Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant. -MC On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote: Hi, I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address. I put <extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> in the default.xml under Dialplan When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above. I Appreciate any help you may offer. Dave Goodwin _______________________________________________ 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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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
Okay, I want you to go read up on gather data and dropping it on pastebin. The skills you hone there will server you well. :) http://wiki.freeswitch.org/wiki/Reporting_Bugs Need a debug log and a sip trace. Drop it on pastebin. Hint: use "sofia global siptrace on" to get all sip traffic logged to the console. Put the pb link in this email thread. -MC On Thu, Jun 16, 2011 at 2:18 PM, Dave <dave@clancysystems.com> wrote:
I did both, and even restarted the service. same results. Velocity did a call capture on thier end and sent it to me. They say that FreeSWITCH is requiring proxy authentication and that that is the issue. SIP Status: 407 Proxy Authentication Required
----- Original Message ----- *From:* Michael Collins <msc@freeswitch.org> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Thursday, June 16, 2011 2:46 PM *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
Did you reloadxml and restart the sofia profile after the changes? Also, you need to "reloadacl" after doing a change to acl.conf.xml.
-MC
On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave@clancysystems.com> wrote:
Yes, I am only testing inbound. When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox) I Get a busy signal and at that moment FreeSWITCH shows... "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP.
If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP.
----- Original Message ----- *From:* Michael Collins <msc@freeswitch.org> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Thursday, June 16, 2011 2:11 PM *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave@clancysystems.com> wrote:
Under "domains" in acl.conf.xml I added
<node type="allow" value="xxx.xxx.xxx.xxx"/>
The correct syntax is: <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
x's are the IP address it's just a single address not a CIDR.
And I created the file velocity_did.xml with the following code..
<include> <extension name="velocity_did">
<condition field="destination_number" expression="^(2064001950)$"> <action application="set" data="domain_name=$${domain}"/> <action application="answer"/>
<action application="lua" data="cme_ivr.lua channel_name"/>
</condition> </extension> </include> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.
Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.
For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them.
-MC
Do I need to change the line ..
<action application="set" data="domain_name=$${domain}"/> to the IP address? If so what syntax?
They have no Domain Name.
Thanks again for your help.
Dave Goodwin
----- Original Message ----- *From:* Michael Collins <msc@freeswitch.org> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Thursday, June 16, 2011 12:15 PM *Subject:* Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
Dave,
If you are simply handling incoming calls then you will need to do two things:
Add Velocity's IP addr to the domains section of acl.conf.xml Add an extension to the public context (conf/dialplan/public.xml)
Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.
Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.
Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.
-MC
On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote:
Hi,
I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.
I put
<extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension>
in the default.xml under Dialplan
When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.
What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.
I Appreciate any help you may offer.
Dave Goodwin
_______________________________________________ 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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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
Will Do. Thanks very much for your help. I bought the book:) Dave ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 3:22 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication Okay, I want you to go read up on gather data and dropping it on pastebin. The skills you hone there will server you well. :) http://wiki.freeswitch.org/wiki/Reporting_Bugs Need a debug log and a sip trace. Drop it on pastebin. Hint: use "sofia global siptrace on" to get all sip traffic logged to the console. Put the pb link in this email thread. -MC On Thu, Jun 16, 2011 at 2:18 PM, Dave <dave@clancysystems.com> wrote: I did both, and even restarted the service. same results. Velocity did a call capture on thier end and sent it to me. They say that FreeSWITCH is requiring proxy authentication and that that is the issue. SIP Status: 407 Proxy Authentication Required ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 2:46 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication Did you reloadxml and restart the sofia profile after the changes? Also, you need to "reloadacl" after doing a change to acl.conf.xml. -MC On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave@clancysystems.com> wrote: Yes, I am only testing inbound. When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox) I Get a busy signal and at that moment FreeSWITCH shows... "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP. If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP. ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 2:11 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave@clancysystems.com> wrote: Under "domains" in acl.conf.xml I added <node type="allow" value="xxx.xxx.xxx.xxx"/> The correct syntax is: <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/> x's are the IP address it's just a single address not a CIDR. And I created the file velocity_did.xml with the following code.. <include> <extension name="velocity_did"> <condition field="destination_number" expression="^(2064001950)$"> <action application="set" data="domain_name=$${domain}"/> <action application="answer"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> </include> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that. For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them. -MC Do I need to change the line .. <action application="set" data="domain_name=$${domain}"/> to the IP address? If so what syntax? They have no Domain Name. Thanks again for your help. Dave Goodwin ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 12:15 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication Dave, If you are simply handling incoming calls then you will need to do two things: Add Velocity's IP addr to the domains section of acl.conf.xml Add an extension to the public context (conf/dialplan/public.xml) Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script. Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge. Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant. -MC On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote: Hi, I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address. I put <extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension> in the default.xml under Dialplan When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more. What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above. I Appreciate any help you may offer. Dave Goodwin _______________________________________________ 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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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
Dave, if you use the default config, I would recommend you configure Vitelity to send you the packets to your port 5080 (profile external). Auth is disabled on external, so I think this port is more adapted to be used for unauth trunks. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 16/06/2011 à 23:46, Dave a écrit :
Will Do. Thanks very much for your help. I bought the book:)
Dave
----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 3:22 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
Okay, I want you to go read up on gather data and dropping it on pastebin. The skills you hone there will server you well. :)
http://wiki.freeswitch.org/wiki/Reporting_Bugs
Need a debug log and a sip trace. Drop it on pastebin. Hint: use "sofia global siptrace on" to get all sip traffic logged to the console.
Put the pb link in this email thread. -MC
On Thu, Jun 16, 2011 at 2:18 PM, Dave <dave@clancysystems.com> wrote: I did both, and even restarted the service. same results. Velocity did a call capture on thier end and sent it to me. They say that FreeSWITCH is requiring proxy authentication and that that is the issue. SIP Status: 407 Proxy Authentication Required
----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 2:46 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
Did you reloadxml and restart the sofia profile after the changes? Also, you need to "reloadacl" after doing a change to acl.conf.xml.
-MC
On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave@clancysystems.com> wrote: Yes, I am only testing inbound. When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox) I Get a busy signal and at that moment FreeSWITCH shows... "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP.
If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP.
----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 2:11 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave@clancysystems.com> wrote:
Under "domains" in acl.conf.xml I added
<node type="allow" value="xxx.xxx.xxx.xxx"/>
The correct syntax is: <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
x's are the IP address it's just a single address not a CIDR.
And I created the file velocity_did.xml with the following code..
<include> <extension name="velocity_did">
<condition field="destination_number" expression="^(2064001950)$"> <action application="set" data="domain_name=$${domain}"/> <action application="answer"/>
<action application="lua" data="cme_ivr.lua channel_name"/>
</condition> </extension> </include> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.
Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.
For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them.
-MC
Do I need to change the line ..
<action application="set" data="domain_name=$${domain}"/> to the IP address? If so what syntax?
They have no Domain Name.
Thanks again for your help.
Dave Goodwin
----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, June 16, 2011 12:15 PM Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
Dave,
If you are simply handling incoming calls then you will need to do two things:
Add Velocity's IP addr to the domains section of acl.conf.xml Add an extension to the public context (conf/dialplan/public.xml)
Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.
Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.
Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.
-MC
On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote: Hi,
I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.
I put
<extension name="velocity"> <condition field="destination_number" expression="^2064001950$"> <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/> <action application="lua" data="cme_ivr.lua channel_name"/> </condition> </extension>
in the default.xml under Dialplan
When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.
What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.
I Appreciate any help you may offer.
Dave Goodwin
_______________________________________________ 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
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
participants (3)
-
Dave
-
David Ponzone
-
Michael Collins