Hello I still got problems with the configuration, since the new layout was added. Today I've updated to latest trunk (config files too) and I've configured the system. I've only done the following changes: 1. I've set domain=mydomain.com in vars.xml 2. I've renamed sip_profiles/default to mydomain.com and nat to nat_mydomain.com 3. I've enabled mod_xml_curl and configured it to fetch dial plans 4. I've commented out mod_dialplan_asterisk and mod_iax 5. I've added my extensions to directory\default (copied brian's config and made appropriate changes) My sip gateway do not use authentication (and I therefore just use " number@mygateway.com" in the dialplan for external calls). The phones register by using mydomain.com as proxy/registrar (not by using an ip). The UA's can register fine and external outbound calls works gr8. Current problems: * I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log * Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log //Jonas
This is why the default config has three profiles. In various cases of nat involved you have to do some special things to allow it to work. Checkout outbound.xml and nat.xml. As for audio you need two profiles.. one for the communicating with phones behind the same nat and one for communicating with the outside world from behind nat. You can't just get away with using one profile for everything in these cases. /b On Jan 8, 2008, at 5:13 AM, freeswitch-users@lists.freeswitch.org wrote:
Hello
I still got problems with the configuration, since the new layout was added.
Today I've updated to latest trunk (config files too) and I've configured the system.
I've only done the following changes: 1. I've set domain=mydomain.com in vars.xml 2. I've renamed sip_profiles/default to mydomain.com and nat to nat_mydomain.com 3. I've enabled mod_xml_curl and configured it to fetch dial plans 4. I've commented out mod_dialplan_asterisk and mod_iax 5. I've added my extensions to directory\default (copied brian's config and made appropriate changes)
My sip gateway do not use authentication (and I therefore just use " number@mygateway.com " in the dialplan for external calls). The phones register by using mydomain.com as proxy/registrar (not by using an ip).
The UA's can register fine and external outbound calls works gr8.
Current problems:
* I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log
* Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log
//Jonas < internal_call .log
<external_call.log>_______________________________________________ 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
Below is a more detailed description of the problem: On Jan 8, 2008, at 5:13 AM, freeswitch-users@lists.freeswitch.org wrote:
Hello
I still got problems with the configuration, since the new layout was added.
Today I've updated to latest trunk (config files too) and I've configured the system.
I've only done the following changes: 1. I've set domain=mydomain.com in vars.xml 2. I've renamed sip_profiles/default to mydomain.com and nat to nat_mydomain.com 3. I've enabled mod_xml_curl and configured it to fetch dial plans 4. I've commented out mod_dialplan_asterisk and mod_iax 5. I've added my extensions to directory\default (copied brian's config and made appropriate changes)
My sip gateway do not use authentication (and I therefore just use " number@mygateway.com " in the dialplan for external calls). The phones register by using mydomain.com as proxy/registrar (not by using an ip).
The UA's can register fine and external outbound calls works gr8.
Current problems:
* I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log
Chances are mydomain.com is your public IP right? and I suspect NAT is involved? If so then you need to setup a profile for Natted phones behind the same nat. One profile for phones outside the nat to register with FreeSWITCH ... and the public one listed below.
* Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log
This is why you need to start another profile to accept calls from the outside world with authentication turned off. That would fix that problem. I honestly like having multiple UA's it solves many problems. /b
* I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log
Chances are mydomain.com is your public IP right? and I suspect NAT is involved? If so then you need to setup a profile for Natted phones behind the same nat. One profile for phones outside the nat to register with FreeSWITCH ... and the public one listed below.
mydomain.com points on my public ip, correct. The phones and FS are behind a router (on the same lan). FS have loaded the three profiles (nat, default and outbound). I have made no changes to them. Currently I have just tested with phones that is on the same lan as FS. here is a registration trans: REGISTER sip:mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.123:5060;rport;branch=z9hG4bK6f95bcc8e01f69dda9f5ebf92adc2c93 From: <sip:tobbe@mydomain.com>;tag=66055002 To: <sip:tobbe@mydomain.com> Call-ID: 58ee65fb0b696f5a370d839058387f2c@192.168.1.123 Contact: <sip:tobbe@192.168.1.123:5060> CSeq: 53 REGISTER Expires: 300 Authorization: Digest username="tobbe",realm="mydomain.com",[... some info stripped out ...],uri="sip:mydomain.com",nc=00000001,algorithm=MD5 User-Agent: CMI CM5K Content-Length: 0 tobbe has user_context set to "default" in conf/directory/default/tobbe.xml
* Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log
This is why you need to start another profile to accept calls from the outside world with authentication turned off. That would fix that problem.
well. the nat profile exists and it has auth_calls set to false. but maybe it isnt used in my config?
I honestly like having multiple UA's it solves many problems.
what do u mean by that? I'm using multiple UA's, all of them are currently on the same LAN as FS right now (testing enviroment), but later on, they may also be elsewhere. But currently I'm trying to get two UA's to be able to talk with each other (as I said, the audio is just going one way). And I'm not saying that it's one UA to rule them all.. =) I'm just trying to understand this all new configuration how a profile is selected (for inbound/outbound external calls, internal calls, internal registration (on the same lan as FS) and external registration (not on the same lan as FS)). thanks, jonas
But trying to use the domain behind nat will never work because it resolves to your public ip. You can't do that in this case unless you have split DNS. Its all about how it all loops out and back thru nat to the box behind nat. /b Sent from my iPhone On Jan 8, 2008, at 7:39 AM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
* I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log
Chances are mydomain.com is your public IP right? and I suspect NAT is involved? If so then you need to setup a profile for Natted phones behind the same nat. One profile for phones outside the nat to register with FreeSWITCH ... and the public one listed below.
mydomain.com points on my public ip, correct. The phones and FS are behind a router (on the same lan). FS have loaded the three profiles (nat, default and outbound). I have made no changes to them. Currently I have just tested with phones that is on the same lan as FS.
here is a registration trans:
REGISTER sip:mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.123: 5060;rport;branch=z9hG4bK6f95bcc8e01f69dda9f5ebf92adc2c93 From: <sip:tobbe@mydomain.com>;tag=66055002 To: <sip:tobbe@mydomain.com> Call-ID: 58ee65fb0b696f5a370d839058387f2c@192.168.1.123 Contact: <sip:tobbe@192.168.1.123:5060> CSeq: 53 REGISTER Expires: 300 Authorization: Digest username="tobbe",realm="mydomain.com",[... some info stripped out ...],uri="sip:mydomain.com",nc=00000001,algorithm=MD5 User-Agent: CMI CM5K Content-Length: 0
tobbe has user_context set to "default" in conf/directory/default/ tobbe.xml
* Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log
This is why you need to start another profile to accept calls from the outside world with authentication turned off. That would fix that problem.
well. the nat profile exists and it has auth_calls set to false. but maybe it isnt used in my config?
I honestly like having multiple UA's it solves many problems.
what do u mean by that? I'm using multiple UA's, all of them are currently on the same LAN as FS right now (testing enviroment), but later on, they may also be elsewhere. But currently I'm trying to get two UA's to be able to talk with each other (as I said, the audio is just going one way).
And I'm not saying that it's one UA to rule them all.. =) I'm just trying to understand this all new configuration how a profile is selected (for inbound/outbound external calls, internal calls, internal registration (on the same lan as FS) and external registration (not on the same lan as FS)).
thanks, jonas
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Btw if you put back the config to the default. It will work because the ip is the domain and the packet don't loop thur your nat router needlessly. /b Sent from my iPhone On Jan 8, 2008, at 7:39 AM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
* I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log
Chances are mydomain.com is your public IP right? and I suspect NAT is involved? If so then you need to setup a profile for Natted phones behind the same nat. One profile for phones outside the nat to register with FreeSWITCH ... and the public one listed below.
mydomain.com points on my public ip, correct. The phones and FS are behind a router (on the same lan). FS have loaded the three profiles (nat, default and outbound). I have made no changes to them. Currently I have just tested with phones that is on the same lan as FS.
here is a registration trans:
REGISTER sip:mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.123: 5060;rport;branch=z9hG4bK6f95bcc8e01f69dda9f5ebf92adc2c93 From: <sip:tobbe@mydomain.com>;tag=66055002 To: <sip:tobbe@mydomain.com> Call-ID: 58ee65fb0b696f5a370d839058387f2c@192.168.1.123 Contact: <sip:tobbe@192.168.1.123:5060> CSeq: 53 REGISTER Expires: 300 Authorization: Digest username="tobbe",realm="mydomain.com",[... some info stripped out ...],uri="sip:mydomain.com",nc=00000001,algorithm=MD5 User-Agent: CMI CM5K Content-Length: 0
tobbe has user_context set to "default" in conf/directory/default/ tobbe.xml
* Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log
This is why you need to start another profile to accept calls from the outside world with authentication turned off. That would fix that problem.
well. the nat profile exists and it has auth_calls set to false. but maybe it isnt used in my config?
I honestly like having multiple UA's it solves many problems.
what do u mean by that? I'm using multiple UA's, all of them are currently on the same LAN as FS right now (testing enviroment), but later on, they may also be elsewhere. But currently I'm trying to get two UA's to be able to talk with each other (as I said, the audio is just going one way).
And I'm not saying that it's one UA to rule them all.. =) I'm just trying to understand this all new configuration how a profile is selected (for inbound/outbound external calls, internal calls, internal registration (on the same lan as FS) and external registration (not on the same lan as FS)).
thanks, jonas
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Back to basics. I've reverted to the default sip profile configs. I've now specified "192.168.1.102" as the domain (the FS ip). Both phones do now register using that ip. The registration goes OK. But when I call from one phone to another (both on the same lan as FS), I only get audio one way. I call 1203 which brigdes to sofia/192.168.1.102/conny in the dialplan. This must be the simplest test case, right? On Jan 8, 2008 2:00 PM, Brian West <brian.west@mac.com> wrote:
Btw if you put back the config to the default. It will work because the ip is the domain and the packet don't loop thur your nat router needlessly.
/b
Sent from my iPhone
On Jan 8, 2008, at 7:39 AM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
* I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log
Chances are mydomain.com is your public IP right? and I suspect NAT is involved? If so then you need to setup a profile for Natted phones behind the same nat. One profile for phones outside the nat to register with FreeSWITCH ... and the public one listed below.
mydomain.com points on my public ip, correct. The phones and FS are behind a router (on the same lan). FS have loaded the three profiles (nat, default and outbound). I have made no changes to them. Currently I have just tested with phones that is on the same lan as FS.
here is a registration trans:
REGISTER sip:mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.123: 5060;rport;branch=z9hG4bK6f95bcc8e01f69dda9f5ebf92adc2c93 From: <sip:tobbe@mydomain.com>;tag=66055002 To: <sip:tobbe@mydomain.com> Call-ID: 58ee65fb0b696f5a370d839058387f2c@192.168.1.123 Contact: <sip:tobbe@192.168.1.123:5060> CSeq: 53 REGISTER Expires: 300 Authorization: Digest username="tobbe",realm="mydomain.com",[... some info stripped out ...],uri="sip:mydomain.com",nc=00000001,algorithm=MD5 User-Agent: CMI CM5K Content-Length: 0
tobbe has user_context set to "default" in conf/directory/default/ tobbe.xml
* Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log
This is why you need to start another profile to accept calls from the outside world with authentication turned off. That would fix that problem.
well. the nat profile exists and it has auth_calls set to false. but maybe it isnt used in my config?
I honestly like having multiple UA's it solves many problems.
what do u mean by that? I'm using multiple UA's, all of them are currently on the same LAN as FS right now (testing enviroment), but later on, they may also be elsewhere. But currently I'm trying to get two UA's to be able to talk with each other (as I said, the audio is just going one way).
And I'm not saying that it's one UA to rule them all.. =) I'm just trying to understand this all new configuration how a profile is selected (for inbound/outbound external calls, internal calls, internal registration (on the same lan as FS) and external registration (not on the same lan as FS)).
thanks, jonas
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On Jan 8, 2008 3:20 PM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
Back to basics. I've reverted to the default sip profile configs. I've now specified "192.168.1.102" as the domain (the FS ip). Both phones do now register using that ip. The registration goes OK.
But when I call from one phone to another (both on the same lan as FS), I only get audio one way. I call 1203 which brigdes to sofia/192.168.1.102/conny in the dialplan. This must be the simplest test case, right?
Are the phones using stun for these calls? If so they may try to use the external IP instead of their internal one.
no, they do not. On Jan 8, 2008 3:24 PM, Mailing List <mailinglists.1234@gmail.com> wrote:
On Jan 8, 2008 3:20 PM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
Back to basics. I've reverted to the default sip profile configs. I've now specified "192.168.1.102" as the domain (the FS ip). Both phones do now register using that ip. The registration goes OK.
But when I call from one phone to another (both on the same lan as FS), I only get audio one way. I call 1203 which brigdes to sofia/192.168.1.102/conny in the dialplan. This must be the simplest test case, right?
Are the phones using stun for these calls? If so they may try to use the external IP instead of their internal one.
_______________________________________________ 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
no, they do not. On 1/8/08, Mailing List <mailinglists.1234@gmail.com> wrote:
On Jan 8, 2008 3:20 PM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
Back to basics. I've reverted to the default sip profile configs. I've now specified "192.168.1.102" as the domain (the FS ip). Both phones do now register using that ip. The registration goes OK.
But when I call from one phone to another (both on the same lan as FS), I only get audio one way. I call 1203 which brigdes to sofia/192.168.1.102/conny in the dialplan. This must be the simplest test case, right?
Are the phones using stun for these calls? If so they may try to use the external IP instead of their internal one.
Yo, can you monitor the traffic between and attach it to us (for xample with wireshark)? maybe it would be clearer then if it's nat issue, or something else? J. Jonas Gauffin wrote:
Back to basics. I've reverted to the default sip profile configs. I've now specified "192.168.1.102" as the domain (the FS ip). Both phones do now register using that ip. The registration goes OK.
But when I call from one phone to another (both on the same lan as FS), I only get audio one way. I call 1203 which brigdes to sofia/192.168.1.102/conny in the dialplan. This must be the simplest test case, right?
On Jan 8, 2008 2:00 PM, Brian West <brian.west@mac.com> wrote:
Btw if you put back the config to the default. It will work because the ip is the domain and the packet don't loop thur your nat router needlessly.
/b
Sent from my iPhone
On Jan 8, 2008, at 7:39 AM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
* I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log
Chances are mydomain.com is your public IP right? and I suspect NAT is involved? If so then you need to setup a profile for Natted phones behind the same nat. One profile for phones outside the nat to register with FreeSWITCH ... and the public one listed below.
mydomain.com points on my public ip, correct. The phones and FS are behind a router (on the same lan). FS have loaded the three profiles (nat, default and outbound). I have made no changes to them. Currently I have just tested with phones that is on the same lan as FS.
here is a registration trans:
REGISTER sip:mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.123: 5060;rport;branch=z9hG4bK6f95bcc8e01f69dda9f5ebf92adc2c93 From: <sip:tobbe@mydomain.com>;tag=66055002 To: <sip:tobbe@mydomain.com> Call-ID: 58ee65fb0b696f5a370d839058387f2c@192.168.1.123 Contact: <sip:tobbe@192.168.1.123:5060> CSeq: 53 REGISTER Expires: 300 Authorization: Digest username="tobbe",realm="mydomain.com",[... some info stripped out ...],uri="sip:mydomain.com",nc=00000001,algorithm=MD5 User-Agent: CMI CM5K Content-Length: 0
tobbe has user_context set to "default" in conf/directory/default/ tobbe.xml
* Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log
This is why you need to start another profile to accept calls from the outside world with authentication turned off. That would fix that problem.
well. the nat profile exists and it has auth_calls set to false. but maybe it isnt used in my config?
I honestly like having multiple UA's it solves many problems.
what do u mean by that? I'm using multiple UA's, all of them are currently on the same LAN as FS right now (testing enviroment), but later on, they may also be elsewhere. But currently I'm trying to get two UA's to be able to talk with each other (as I said, the audio is just going one way).
And I'm not saying that it's one UA to rule them all.. =) I'm just trying to understand this all new configuration how a profile is selected (for inbound/outbound external calls, internal calls, internal registration (on the same lan as FS) and external registration (not on the same lan as FS)).
thanks, jonas
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You revert the profiles too? /b On Jan 8, 2008, at 9:20 AM, Jonas Gauffin wrote:
Back to basics. I've reverted to the default sip profile configs. I've now specified "192.168.1.102" as the domain (the FS ip). Both phones do now register using that ip. The registration goes OK.
But when I call from one phone to another (both on the same lan as FS), I only get audio one way. I call 1203 which brigdes to sofia/192.168.1.102/conny in the dialplan. This must be the simplest test case, right?
On Jan 8, 2008 2:00 PM, Brian West <brian.west@mac.com> wrote:
Btw if you put back the config to the default. It will work because the ip is the domain and the packet don't loop thur your nat router needlessly.
/b
Sent from my iPhone
On Jan 8, 2008, at 7:39 AM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
* I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log
Chances are mydomain.com is your public IP right? and I suspect NAT is involved? If so then you need to setup a profile for Natted phones behind the same nat. One profile for phones outside the nat to register with FreeSWITCH ... and the public one listed below.
mydomain.com points on my public ip, correct. The phones and FS are behind a router (on the same lan). FS have loaded the three profiles (nat, default and outbound). I have made no changes to them. Currently I have just tested with phones that is on the same lan as FS.
here is a registration trans:
REGISTER sip:mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.123: 5060;rport;branch=z9hG4bK6f95bcc8e01f69dda9f5ebf92adc2c93 From: <sip:tobbe@mydomain.com>;tag=66055002 To: <sip:tobbe@mydomain.com> Call-ID: 58ee65fb0b696f5a370d839058387f2c@192.168.1.123 Contact: <sip:tobbe@192.168.1.123:5060> CSeq: 53 REGISTER Expires: 300 Authorization: Digest username="tobbe",realm="mydomain.com",[... some info stripped out ...],uri="sip:mydomain.com",nc=00000001,algorithm=MD5 User-Agent: CMI CM5K Content-Length: 0
tobbe has user_context set to "default" in conf/directory/default/ tobbe.xml
* Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log
This is why you need to start another profile to accept calls from the outside world with authentication turned off. That would fix that problem.
well. the nat profile exists and it has auth_calls set to false. but maybe it isnt used in my config?
I honestly like having multiple UA's it solves many problems.
what do u mean by that? I'm using multiple UA's, all of them are currently on the same LAN as FS right now (testing enviroment), but later on, they may also be elsewhere. But currently I'm trying to get two UA's to be able to talk with each other (as I said, the audio is just going one way).
And I'm not saying that it's one UA to rule them all.. =) I'm just trying to understand this all new configuration how a profile is selected (for inbound/outbound external calls, internal calls, internal registration (on the same lan as FS) and external registration (not on the same lan as FS)).
thanks, jonas
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participants (6)
-
Brian West
-
freeswitch-users＠lists.freeswitch.org
-
Jonas Gauffin
-
jonas.gauffin＠gmail.com
-
Josip Djuricic
-
Mailing List