This feel's like a very stupid question, but i've scoured for hours through the documents, and samples I can find without finding an answer. I'm assuming I'm missing something very obvious. I'm just trying to have freeswitch accept a call from a static IP address, then forward that call to a provider, essentially a very static SIP proxy. I've tried added my IP in question to the acl.xml file, but no success. All calls are rejected with a 407 Auth required. Thanks for any pointers. I think I will write a paragraph or two on this when I get it working, as it's something I think a lot of people are looking to try. Asterisk will not scale enough, and I don't want to do this via OpenSER/RTPProxy, etc. -Jed
Jed, here are the list of things you can do: 1. sip_profiles/default.xml -> change context to default and set auth- calls=false 2. Then you can use ${network_addr} in your conditions or the ${acl()} function an example is in the default.xml dialplan. /b On Apr 29, 2008, at 4:44 PM, Jed Stafford wrote:
This feel's like a very stupid question, but i've scoured for hours through the documents, and samples I can find without finding an answer. I'm assuming I'm missing something very obvious.
I'm just trying to have freeswitch accept a call from a static IP address, then forward that call to a provider, essentially a very static SIP proxy.
I've tried added my IP in question to the acl.xml file, but no success. All calls are rejected with a 407 Auth required.
Thanks for any pointers. I think I will write a paragraph or two on this when I get it working, as it's something I think a lot of people are looking to try. Asterisk will not scale enough, and I don't want to do this via OpenSER/RTPProxy, etc.
-Jed
_______________________________________________ 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
it seems a bit confusing but if you want to do ip auth the way you describe, you actually have to disable the digest auth so that the other end is not challenged for auth credentials on top of the ip auth. search your profile for this: <param name="auth-calls" value="true"/> and comment it by encapsulating it in <!-- and --> then add this line: <param name="apply-inbound-acl" value="sip_ip_auth"/> Then make sip_ip_auth by editing acl.conf.xml and add a new list to the <network-lists> tag. assuming the ip you trust is 200.2.2.2: <list name="sip_ip_auth" default="deny"> <node type="allow" cidr="200.2.2.2/32"/> </list> Now all sip calls will be rejected unless they are originated by 200.2.2.2 On Tue, Apr 29, 2008 at 8:17 PM, Brian West <brian@freeswitch.org> wrote:
Jed, here are the list of things you can do:
1. sip_profiles/default.xml -> change context to default and set auth- calls=false 2. Then you can use ${network_addr} in your conditions or the ${acl()} function an example is in the default.xml dialplan.
/b
On Apr 29, 2008, at 4:44 PM, Jed Stafford wrote:
This feel's like a very stupid question, but i've scoured for hours through the documents, and samples I can find without finding an answer. I'm assuming I'm missing something very obvious.
I'm just trying to have freeswitch accept a call from a static IP address, then forward that call to a provider, essentially a very static SIP proxy.
I've tried added my IP in question to the acl.xml file, but no success. All calls are rejected with a 407 Auth required.
Thanks for any pointers. I think I will write a paragraph or two on this when I get it working, as it's something I think a lot of people are looking to try. Asterisk will not scale enough, and I don't want to do this via OpenSER/RTPProxy, etc.
-Jed
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I have added this ACL topic to the meeting today so we can make sure its documented better on the wiki. 1pm CST sip:888@conference.freeswitch.org /b On Apr 30, 2008, at 8:14 AM, Anthony Minessale wrote:
it seems a bit confusing but if you want to do ip auth the way you describe, you actually have to disable the digest auth so that the other end is not challenged for auth credentials on top of the ip auth.
search your profile for this:
<param name="auth-calls" value="true"/>
and comment it by encapsulating it in <!-- and -->
then add this line:
<param name="apply-inbound-acl" value="sip_ip_auth"/>
Then make sip_ip_auth by editing acl.conf.xml and add a new list to the <network-lists> tag. assuming the ip you trust is 200.2.2.2:
<list name="sip_ip_auth" default="deny"> <node type="allow" cidr="200.2.2.2/32"/> </list>
Now all sip calls will be rejected unless they are originated by 200.2.2.2
On Tue, Apr 29, 2008 at 8:17 PM, Brian West <brian@freeswitch.org> wrote: Jed, here are the list of things you can do:
1. sip_profiles/default.xml -> change context to default and set auth- calls=false 2. Then you can use ${network_addr} in your conditions or the ${acl()} function an example is in the default.xml dialplan.
/b
On Apr 29, 2008, at 4:44 PM, Jed Stafford wrote:
This feel's like a very stupid question, but i've scoured for hours through the documents, and samples I can find without finding an answer. I'm assuming I'm missing something very obvious.
I'm just trying to have freeswitch accept a call from a static IP address, then forward that call to a provider, essentially a very static SIP proxy.
I've tried added my IP in question to the acl.xml file, but no success. All calls are rejected with a 407 Auth required.
Thanks for any pointers. I think I will write a paragraph or two on this when I get it working, as it's something I think a lot of people are looking to try. Asterisk will not scale enough, and I don't want to do this via OpenSER/RTPProxy, etc.
-Jed
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ 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
Brian West sip:brian@freeswitch.org
Thank you, all is working much better now. On Wed, Apr 30, 2008 at 6:14 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
it seems a bit confusing but if you want to do ip auth the way you describe, you actually have to disable the digest auth so that the other end is not challenged for auth credentials on top of the ip auth.
search your profile for this:
<param name="auth-calls" value="true"/>
and comment it by encapsulating it in <!-- and -->
then add this line:
<param name="apply-inbound-acl" value="sip_ip_auth"/>
Then make sip_ip_auth by editing acl.conf.xml and add a new list to the <network-lists> tag. assuming the ip you trust is 200.2.2.2:
<list name="sip_ip_auth" default="deny"> <node type="allow" cidr="200.2.2.2/32"/> </list>
Now all sip calls will be rejected unless they are originated by 200.2.2.2
On Tue, Apr 29, 2008 at 8:17 PM, Brian West <brian@freeswitch.org> wrote:
Jed, here are the list of things you can do:
1. sip_profiles/default.xml -> change context to default and set auth- calls=false 2. Then you can use ${network_addr} in your conditions or the ${acl()} function an example is in the default.xml dialplan.
/b
On Apr 29, 2008, at 4:44 PM, Jed Stafford wrote:
This feel's like a very stupid question, but i've scoured for hours through the documents, and samples I can find without finding an answer. I'm assuming I'm missing something very obvious.
I'm just trying to have freeswitch accept a call from a static IP address, then forward that call to a provider, essentially a very static SIP proxy.
I've tried added my IP in question to the acl.xml file, but no success. All calls are rejected with a 407 Auth required.
Thanks for any pointers. I think I will write a paragraph or two on this when I get it working, as it's something I think a lot of people are looking to try. Asterisk will not scale enough, and I don't want to do this via OpenSER/RTPProxy, etc.
-Jed
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ 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)
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Anthony Minessale
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Brian West
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Jed Stafford