Hi, There is requirement for me to add multiple sip gateways to Freeswitch, the first sip gateway uses the external.xml in sip profile to route traffic to Sip provider A. The second sip gateway will use a different UA to route traffic to Sip provider B. Here are my questions: 1. Do I need to create another external.xml with its own ip and port which point to the Sip Provider B? if that's case, what do I need to change in the sip profile external.xml file? 2. What is the rtp-ip and sip-ip in the external.xml file? Should that be the local IP of my sip server? 3. What do I need to change in my dialplan so that I can reference the Sip_Provider_B? an example would be good. External.xml <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="1.2.3.4"/> <param name="ext-sip-ip" value="1.2.3.4"/> External/sip_provider_A.xml <gateway name="Sip_Provider_A"> <param name="username" value="xxx"/> <param name="password" value="xxx"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> External/sip_provider_B.xml <gateway name="Sip_Provider_B"> <param name="username" value="aaa"/> <param name="password" value="bbb"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> Cheers, Louie
A single profile can send to any number of gateways. You only need multiple profiles if you need to listen in multiple places, usually to handle either multiple IP addresses or different configurations on different ports. sip-ip and rtp-ip are the local IP of the server where you're receiving packets. It must be a real address, 0.0.0.0 (ie any address) isn't valid. If you're behind NAT they should be the internal IP, there are settings such as ext-sip-ip and ext-rtp-ip for handling NAT traversal. How to select the gateway to use will be very dependant on your use-case. However in general, you pick the destination gateway in the dialstring you bridge with, eg <action application="bridge" data="sofia/gateway/Sip_Provider_A/12345"/> <action application="bridge" data="sofia/gateway/Sip_Provider_B/12345"/> How you pick it is up to you, but as some examples you could set it as a variable in the user directory or based on conditions in the dialstring, you could use mod_distributor, mod_lcr etc, or try each in sequence as above. Steve On 23 July 2014 14:48, Louie Liu <lliu@multinet.net.au> wrote:
Hi,
There is requirement for me to add multiple sip gateways to Freeswitch, the first sip gateway uses the external.xml in sip profile to route traffic to Sip provider A. The second sip gateway will use a different UA to route traffic to Sip provider B. Here are my questions:
1. Do I need to create another external.xml with its own ip and port which point to the Sip Provider B? if that’s case, what do I need to change in the sip profile external.xml file?
2. What is the rtp-ip and sip-ip in the external.xml file? Should that be the local IP of my sip server?
3. What do I need to change in my dialplan so that I can reference the Sip_Provider_B? an example would be good.
External.xml
<param name="sip-port" value="$${external_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="1.2.3.4"/>
<param name="ext-sip-ip" value="1.2.3.4"/>
External/sip_provider_A.xml
<gateway name="Sip_Provider_A">
<param name="username" value="xxx"/>
<param name="password" value="xxx"/>
<param name="from-user" value="03xxxxxxx"/>
<param name="from-domain" value="x.y.z"/>
<param name="auth-username" value="yyyyy"/>
<param name="realm" value="bwas02.voip.izzz.zz.au"/>
<param name="proxy" value="sipconnect.voip.zzzz.zz.au"/>
<param name="register" value="true"/>
<param name="register-transport" value="udp"/>
<param name="context" value="public"/>
</gateway>
External/sip_provider_B.xml
<gateway name="Sip_Provider_B">
<param name="username" value="aaa"/>
<param name="password" value="bbb"/>
<param name="from-user" value="03xxxxxxx"/>
<param name="from-domain" value="x.y.z"/>
<param name="auth-username" value="yyyyy"/>
<param name="realm" value="bwas02.voip.izzz.zz.au"/>
<param name="proxy" value="sipconnect.voip.zzzz.zz.au"/>
<param name="register" value="true"/>
<param name="register-transport" value="udp"/>
<param name="context" value="public"/>
</gateway>
Cheers,
Louie
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Thanks your prompt response, maybe I haven’t explain my scenario clearly, Iet me explain: My company would like to setup a voip service to be host in the data centre where Sip_Provider_A and B both have a presence, so avoid voip traffic route across the Internet, but each would need to go through different network interface and gateway to reach the sip providers (see diagram below for illustration). PHONE -> FS(192.168.0.1) -> NAT1(Public IP 1.2.3.4) -> Sip_Provider_A I--> NAT2 (Public IP 5.6.7.8) -> Sip_Provider_B To complicate this even more, Sip_Provider_B offers multiple accounts to cover the existing ISDN connections we’ve got. To me it makes sense to setup two external sip profiles with different external port each to handle traffic route to each sip provider. What is not clear is how handle multiple accounts with same sip provider? Can all the accounts share the same sip profile? Can Freeswitch register multiple accounts at once on the same sip profile? How is the dialplan gonna to work in this case? Examples to illustrate would be good. Cheers, Louie From: Steven Ayre [mailto:steveayre@gmail.com] Sent: Thursday, 24 July 2014 1:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] multiple sip gateway configuration A single profile can send to any number of gateways. You only need multiple profiles if you need to listen in multiple places, usually to handle either multiple IP addresses or different configurations on different ports. sip-ip and rtp-ip are the local IP of the server where you're receiving packets. It must be a real address, 0.0.0.0 (ie any address) isn't valid. If you're behind NAT they should be the internal IP, there are settings such as ext-sip-ip and ext-rtp-ip for handling NAT traversal. How to select the gateway to use will be very dependant on your use-case. However in general, you pick the destination gateway in the dialstring you bridge with, eg <action application="bridge" data="sofia/gateway/Sip_Provider_A/12345"/> <action application="bridge" data="sofia/gateway/Sip_Provider_B/12345"/> How you pick it is up to you, but as some examples you could set it as a variable in the user directory or based on conditions in the dialstring, you could use mod_distributor, mod_lcr etc, or try each in sequence as above. Steve On 23 July 2014 14:48, Louie Liu <lliu@multinet.net.au> wrote: Hi, There is requirement for me to add multiple sip gateways to Freeswitch, the first sip gateway uses the external.xml in sip profile to route traffic to Sip provider A. The second sip gateway will use a different UA to route traffic to Sip provider B. Here are my questions: 1. Do I need to create another external.xml with its own ip and port which point to the Sip Provider B? if that’s case, what do I need to change in the sip profile external.xml file? 2. What is the rtp-ip and sip-ip in the external.xml file? Should that be the local IP of my sip server? 3. What do I need to change in my dialplan so that I can reference the Sip_Provider_B? an example would be good. External.xml <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="1.2.3.4"/> <param name="ext-sip-ip" value="1.2.3.4"/> External/sip_provider_A.xml <gateway name="Sip_Provider_A"> <param name="username" value="xxx"/> <param name="password" value="xxx"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> External/sip_provider_B.xml <gateway name="Sip_Provider_B"> <param name="username" value="aaa"/> <param name="password" value="bbb"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> Cheers, Louie _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 Louie, One SIP provider profile (ie. port) may have multiple gateways like we have for our Vitelity SIP trunk. https://wiki.freeswitch.org/wiki/Provider_Configuration:_Vitelity Regards, Robert From: Louie Liu [mailto:lliu@multinet.net.au] Sent: Thursday, July 24, 2014 6:02 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] multiple sip gateway configuration Thanks your prompt response, maybe I haven’t explain my scenario clearly, Iet me explain: My company would like to setup a voip service to be host in the data centre where Sip_Provider_A and B both have a presence, so avoid voip traffic route across the Internet, but each would need to go through different network interface and gateway to reach the sip providers (see diagram below for illustration). PHONE -> FS(192.168.0.1) -> NAT1(Public IP 1.2.3.4) -> Sip_Provider_A I--> NAT2 (Public IP 5.6.7.8) -> Sip_Provider_B To complicate this even more, Sip_Provider_B offers multiple accounts to cover the existing ISDN connections we’ve got. To me it makes sense to setup two external sip profiles with different external port each to handle traffic route to each sip provider. What is not clear is how handle multiple accounts with same sip provider? Can all the accounts share the same sip profile? Can Freeswitch register multiple accounts at once on the same sip profile? How is the dialplan gonna to work in this case? Examples to illustrate would be good. Cheers, Louie From: Steven Ayre [mailto:steveayre@gmail.com] Sent: Thursday, 24 July 2014 1:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] multiple sip gateway configuration A single profile can send to any number of gateways. You only need multiple profiles if you need to listen in multiple places, usually to handle either multiple IP addresses or different configurations on different ports. sip-ip and rtp-ip are the local IP of the server where you're receiving packets. It must be a real address, 0.0.0.0 (ie any address) isn't valid. If you're behind NAT they should be the internal IP, there are settings such as ext-sip-ip and ext-rtp-ip for handling NAT traversal. How to select the gateway to use will be very dependant on your use-case. However in general, you pick the destination gateway in the dialstring you bridge with, eg <action application="bridge" data="sofia/gateway/Sip_Provider_A/12345"/> <action application="bridge" data="sofia/gateway/Sip_Provider_B/12345"/> How you pick it is up to you, but as some examples you could set it as a variable in the user directory or based on conditions in the dialstring, you could use mod_distributor, mod_lcr etc, or try each in sequence as above. Steve On 23 July 2014 14:48, Louie Liu <lliu@multinet.net.au<mailto:lliu@multinet.net.au>> wrote: Hi, There is requirement for me to add multiple sip gateways to Freeswitch, the first sip gateway uses the external.xml in sip profile to route traffic to Sip provider A. The second sip gateway will use a different UA to route traffic to Sip provider B. Here are my questions: 1. Do I need to create another external.xml with its own ip and port which point to the Sip Provider B? if that’s case, what do I need to change in the sip profile external.xml file? 2. What is the rtp-ip and sip-ip in the external.xml file? Should that be the local IP of my sip server? 3. What do I need to change in my dialplan so that I can reference the Sip_Provider_B? an example would be good. External.xml <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="1.2.3.4"/> <param name="ext-sip-ip" value="1.2.3.4"/> External/sip_provider_A.xml <gateway name="Sip_Provider_A"> <param name="username" value="xxx"/> <param name="password" value="xxx"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au<http://bwas02.voip.izzz.zz.au>"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au<http://sipconnect.voip.zzzz.zz.au>"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> External/sip_provider_B.xml <gateway name="Sip_Provider_B"> <param name="username" value="aaa"/> <param name="password" value="bbb"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au<http://bwas02.voip.izzz.zz.au>"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au<http://sipconnect.voip.zzzz.zz.au>"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> Cheers, Louie _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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
Suppose I have to create a separate gateway xml for each account I’ve had with same sip provider. Upon registration, I should see all these accounts register with same Sip Provider as separate gateways in sofia. As I know that each sip profile is a user agent. A user agent can service exactly one IP:Port. In my case, all my accounts/gateways are register through the same IP:Port to the same sip provider, how does FS keep track of what voip call belongs which account? I am a bit confuse about the concept of multiple gateways share the same IP:port. Would someone enlighten me on this topic? Cheers, Louie From: Robert Hadley [mailto:robert.hadley@teotech.com] Sent: Friday, 25 July 2014 2:33 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] multiple sip gateway configuration Hi Louie, One SIP provider profile (ie. port) may have multiple gateways like we have for our Vitelity SIP trunk. https://wiki.freeswitch.org/wiki/Provider_Configuration:_Vitelity Regards, Robert From: Louie Liu [mailto:lliu@multinet.net.au] Sent: Thursday, July 24, 2014 6:02 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] multiple sip gateway configuration Thanks your prompt response, maybe I haven’t explain my scenario clearly, Iet me explain: My company would like to setup a voip service to be host in the data centre where Sip_Provider_A and B both have a presence, so avoid voip traffic route across the Internet, but each would need to go through different network interface and gateway to reach the sip providers (see diagram below for illustration). PHONE -> FS(192.168.0.1) -> NAT1(Public IP 1.2.3.4) -> Sip_Provider_A I--> NAT2 (Public IP 5.6.7.8) -> Sip_Provider_B To complicate this even more, Sip_Provider_B offers multiple accounts to cover the existing ISDN connections we’ve got. To me it makes sense to setup two external sip profiles with different external port each to handle traffic route to each sip provider. What is not clear is how handle multiple accounts with same sip provider? Can all the accounts share the same sip profile? Can Freeswitch register multiple accounts at once on the same sip profile? How is the dialplan gonna to work in this case? Examples to illustrate would be good. Cheers, Louie From: Steven Ayre [mailto:steveayre@gmail.com] Sent: Thursday, 24 July 2014 1:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] multiple sip gateway configuration A single profile can send to any number of gateways. You only need multiple profiles if you need to listen in multiple places, usually to handle either multiple IP addresses or different configurations on different ports. sip-ip and rtp-ip are the local IP of the server where you're receiving packets. It must be a real address, 0.0.0.0 (ie any address) isn't valid. If you're behind NAT they should be the internal IP, there are settings such as ext-sip-ip and ext-rtp-ip for handling NAT traversal. How to select the gateway to use will be very dependant on your use-case. However in general, you pick the destination gateway in the dialstring you bridge with, eg <action application="bridge" data="sofia/gateway/Sip_Provider_A/12345"/> <action application="bridge" data="sofia/gateway/Sip_Provider_B/12345"/> How you pick it is up to you, but as some examples you could set it as a variable in the user directory or based on conditions in the dialstring, you could use mod_distributor, mod_lcr etc, or try each in sequence as above. Steve On 23 July 2014 14:48, Louie Liu <lliu@multinet.net.au> wrote: Hi, There is requirement for me to add multiple sip gateways to Freeswitch, the first sip gateway uses the external.xml in sip profile to route traffic to Sip provider A. The second sip gateway will use a different UA to route traffic to Sip provider B. Here are my questions: 1. Do I need to create another external.xml with its own ip and port which point to the Sip Provider B? if that’s case, what do I need to change in the sip profile external.xml file? 2. What is the rtp-ip and sip-ip in the external.xml file? Should that be the local IP of my sip server? 3. What do I need to change in my dialplan so that I can reference the Sip_Provider_B? an example would be good. External.xml <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="1.2.3.4"/> <param name="ext-sip-ip" value="1.2.3.4"/> External/sip_provider_A.xml <gateway name="Sip_Provider_A"> <param name="username" value="xxx"/> <param name="password" value="xxx"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> External/sip_provider_B.xml <gateway name="Sip_Provider_B"> <param name="username" value="aaa"/> <param name="password" value="bbb"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> Cheers, Louie _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 guys, Have a minor issue. I have multiple freeswitch boxes that handle different DIDs, my vendor only allows me to have 1 binding per IP, so I do some NATing to get around that. I ran in to a simple issue, that I just don't know how to implement, quite sure it's possible though. My provider expects 404 sip response if the DID isn't served by the PBX it's trying (it tries all my binding in order, until one succeeds). Issue I'm having is that the DID I'm trying to configure is on the second binding in the order of my provider, but my first binding (first PBX) returns I believe a 503, or party did not respond, how would I get freeswitch to return 404 if the DID being sent to it is not hosted there? Thanks for your help.
Hi, using the following dialplan-extension, FreeSWITCH returns a "SIP/2.0 404 Not Found" error when calling the specified number. <extension name="generate404"> <condition field="destination_number" expression="^call404$"> *<action application="hangup" data="UNALLOCATED_NUMBER" />* </condition> </extension> The key to this is the hangup cause. A complete list can be found here: https://wiki.freeswitch.org/wiki/Hangup_Causes Regards, Bernhard Am 09.08.2014 23:18, schrieb Pasha:
Hi guys,
Have a minor issue. I have multiple freeswitch boxes that handle different DIDs, my vendor only allows me to have 1 binding per IP, so I do some NATing to get around that.
I ran in to a simple issue, that I just don't know how to implement, quite sure it's possible though.
My provider expects 404 sip response if the DID isn't served by the PBX it's trying (it tries all my binding in order, until one succeeds).
Issue I'm having is that the DID I'm trying to configure is on the second binding in the order of my provider, but my first binding (first PBX) returns I believe a 503, or party did not respond, how would I get freeswitch to return 404 if the DID being sent to it is not hosted there?
Thanks for your help.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 Bernard, That worked very well thank you! I just added that as the last thing on the public dialplan to process and if no DID matched on that box, it returned the 404 as per your code. Much appreciated! On 14-08-09 02:34 PM, Bernhard Wendel wrote:
Hi,
using the following dialplan-extension, FreeSWITCH returns a "SIP/2.0 404 Not Found" error when calling the specified number.
<extension name="generate404"> <condition field="destination_number" expression="^call404$"> *<action application="hangup" data="UNALLOCATED_NUMBER" />* </condition> </extension>
The key to this is the hangup cause. A complete list can be found here: https://wiki.freeswitch.org/wiki/Hangup_Causes
Regards, Bernhard
Am 09.08.2014 23:18, schrieb Pasha:
Hi guys,
Have a minor issue. I have multiple freeswitch boxes that handle different DIDs, my vendor only allows me to have 1 binding per IP, so I do some NATing to get around that.
I ran in to a simple issue, that I just don't know how to implement, quite sure it's possible though.
My provider expects 404 sip response if the DID isn't served by the PBX it's trying (it tries all my binding in order, until one succeeds).
Issue I'm having is that the DID I'm trying to configure is on the second binding in the order of my provider, but my first binding (first PBX) returns I believe a 503, or party did not respond, how would I get freeswitch to return 404 if the DID being sent to it is not hosted there?
Thanks for your help.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Why not have a routing box that figures out which host the did os on and sends the call there this can be accomplished by using a shared db with reg info from every server Sent from my iPhone
On Aug 9, 2014, at 4:18 PM, Pasha <pasha@prosperity4ever.com> wrote:
Hi guys,
Have a minor issue. I have multiple freeswitch boxes that handle different DIDs, my vendor only allows me to have 1 binding per IP, so I do some NATing to get around that.
I ran in to a simple issue, that I just don't know how to implement, quite sure it's possible though.
My provider expects 404 sip response if the DID isn't served by the PBX it's trying (it tries all my binding in order, until one succeeds).
Issue I'm having is that the DID I'm trying to configure is on the second binding in the order of my provider, but my first binding (first PBX) returns I believe a 503, or party did not respond, how would I get freeswitch to return 404 if the DID being sent to it is not hosted there?
Thanks for your help.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
This sounds like a good idea (would that be a proxy I would need to setup)? Any chance you can refer to me to some documentation / how-to for this setup? Thanks Paul On 14-08-09 02:37 PM, Ken Rice wrote:
Why not have a routing box that figures out which host the did os on and sends the call there this can be accomplished by using a shared db with reg info from every server Sent from my iPhone
On Aug 9, 2014, at 4:18 PM, Pasha <pasha@prosperity4ever.com> wrote:
Hi guys,
Have a minor issue. I have multiple freeswitch boxes that handle different DIDs, my vendor only allows me to have 1 binding per IP, so I do some NATing to get around that.
I ran in to a simple issue, that I just don't know how to implement, quite sure it's possible though.
My provider expects 404 sip response if the DID isn't served by the PBX it's trying (it tries all my binding in order, until one succeeds).
Issue I'm having is that the DID I'm trying to configure is on the second binding in the order of my provider, but my first binding (first PBX) returns I believe a 503, or party did not respond, how would I get freeswitch to return 404 if the DID being sent to it is not hosted there?
Thanks for your help.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Is this what you need? https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_respond Sent from my BlackBerry® PlayBook™ www.blackberry.com ------------------------------ *From:* "Pasha" <pasha@prosperity4ever.com> *To:* "freeswitch-users@lists.freeswitch.org" < freeswitch-users@lists.freeswitch.org> *Sent:* 9 August, 2014 5:20 PM *Subject:* [Freeswitch-users] SIP Response Hi guys, Have a minor issue. I have multiple freeswitch boxes that handle different DIDs, my vendor only allows me to have 1 binding per IP, so I do some NATing to get around that. I ran in to a simple issue, that I just don't know how to implement, quite sure it's possible though. My provider expects 404 sip response if the DID isn't served by the PBX it's trying (it tries all my binding in order, until one succeeds). Issue I'm having is that the DID I'm trying to configure is on the second binding in the order of my provider, but my first binding (first PBX) returns I believe a 503, or party did not respond, how would I get freeswitch to return 404 if the DID being sent to it is not hosted there? Thanks for your help. _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
Thanks for the response. It was close, but for a noob still wasn't enough to give me a clear idea of what I needed to do. One of the other reponses worked for me. Thanks Russel. On 14-08-09 02:39 PM, Russell Treleaven wrote:
Is this what you need?
https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_respond
Sent from my BlackBerry® PlayBook^(TM) www.blackberry.com <http://www.blackberry.com>
------------------------------------------------------------------------ *From:* "Pasha" <pasha@prosperity4ever.com <mailto:pasha@prosperity4ever.com>> *To:* "freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>" <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> *Sent:* 9 August, 2014 5:20 PM *Subject:* [Freeswitch-users] SIP Response
Hi guys,
Have a minor issue. I have multiple freeswitch boxes that handle different DIDs, my vendor only allows me to have 1 binding per IP, so I do some NATing to get around that.
I ran in to a simple issue, that I just don't know how to implement, quite sure it's possible though.
My provider expects 404 sip response if the DID isn't served by the PBX it's trying (it tries all my binding in order, until one succeeds).
Issue I'm having is that the DID I'm trying to configure is on the second binding in the order of my provider, but my first binding (first PBX) returns I believe a 503, or party did not respond, how would I get freeswitch to return 404 if the DID being sent to it is not hosted there?
Thanks for your help.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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, I currently experiencing a 10 minutes drop problem from the gateway registration with our SIP provider, after some diagnostic, I have found that our SIP provider has recently implement a 10 min update check which confirm to the SIP RFC3311 protocol. My question is 1. Does FS sofia stack support SIP RFC3311 protocol? What parameters to use to turn it on at the profile level? 2. I am running a old version of FS: v1.0 in the production, does this protocol implement in version 1.0 of FS? If not, what options do I have to resolve this issue? Upgrade to latest would be the last resort, avoid it if I can. Cheers, Louie
Upgrading to latest FS is not something you should resist. You'll very quickly find yourself 10 years behind with *no* upgrade path, rather than 5 years behind with a difficult one. On 11 August 2014 08:56, Louie Liu <lliu@multinet.net.au> wrote:
Hi,
I currently experiencing a 10 minutes drop problem from the gateway registration with our SIP provider, after some diagnostic, I have found that our SIP provider has recently implement a 10 min update check which confirm to the SIP RFC3311 protocol.
My question is
1. Does FS sofia stack support SIP RFC3311 protocol? What parameters to use to turn it on at the profile level?
2. I am running a old version of FS: v1.0 in the production, does this protocol implement in version 1.0 of FS? If not, what options do I have to resolve this issue? Upgrade to latest would be the last resort, avoid it if I can.
Cheers,
Louie
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 Agree with Ben, Dont find yourself left behind. Id suggest posting a SIP trace of what exactly is going on. Another more radical option is to move to another provider... I can speak for www.netsip.com.au :) But seriously, a SIP trace is your best bet AND schedule time to perform the upgrade. Jay On 11 August 2014 21:56, Louie Liu <lliu@multinet.net.au> wrote:
Hi,
I currently experiencing a 10 minutes drop problem from the gateway registration with our SIP provider, after some diagnostic, I have found that our SIP provider has recently implement a 10 min update check which confirm to the SIP RFC3311 protocol.
My question is
1. Does FS sofia stack support SIP RFC3311 protocol? What parameters to use to turn it on at the profile level?
2. I am running a old version of FS: v1.0 in the production, does this protocol implement in version 1.0 of FS? If not, what options do I have to resolve this issue? Upgrade to latest would be the last resort, avoid it if I can.
Cheers,
Louie
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Sincerely Jay
participants (9)
-
Ben Langfeld
-
Bernhard Wendel
-
jay binks
-
Ken Rice
-
Louie Liu
-
Pasha
-
Robert Hadley
-
Russell Treleaven
-
Steven Ayre