doublenat outgoing call issue
Hi, I wonder if some kind soul might be able please to help me by pointing out what I am doing wrong with my doublenat configuration. I have incoming calls from WAN --> LAN working fine, but when I try to call from LAN -->WAN, the Freeswitch CLI says: "[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]" The setup is essentially: phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT (public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z) Phone_1 is able to register with Freeswitch using a "doublenat" sip profile I set up using port 5090, created from the default external profile with the port changed. phone_1 can initiate calls to phone_2 fine with no problems with audio - everything works just fine in that direction. However when I try to call from phone_2 --> phone_1 I get the message "The person at extension 1002 is not available. Record your message at the tone." Meanwhile the CLI shows that Freeswitch says phone_1 is not registered. In trying to solve this I have run a Wireshark trace onthe PC running the phone_1 softphone to check that the SIP registration seems to go as I understand it should. the dialog is as follows: phone_1 --> FS request: REGISTER FS --> phone_1 status: 401 Unauthorised phone_1 --> FS request: REGISTER FS --> phone_1 status: 200 OK phone_1 --> FS request: SUBSCRIBE FS --> phone_1 status: 405 Method Not Allowed There does appear to be a difference here to what happens when phone_2 (on the same LAN as Freeswitch) registers. With phone_2 I see: phone_2 --> FS request: REGISTER FS --> phone_2 status: 401 Unauthorised phone_2 --> FS request: REGISTER FS --> phone_2 status 200 OK phone_2 --> FS request: SUBSCRIBE FS --> phone_2 status: 202 Accepted FS --> phone_2 request: NOTIFY phone_2 --> FS status: 200 OK More configuration details: In conf/sip_profiles/doublenat.xml I have not uncommented <param name="force-register-domain" value="$$(domain)"/> In conf/sip_profiles/doublenat.xml I have tried setting the context to each of public and default. This does not seem to make a difference. In conf/directory/doublenat.xml I have set the domain name to be the domain name registerd for Fs's external IP with DNS, and used by phone_1. In conf/directory/doublenat.xml I have tried both of the following 9I don't know if this is relevant) 1. Firstly with phone_1 registering to an extension in the default configuration, held in conf/directory/default/: <groups> <group name="default"> <users> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </users> </group> 2. Then I tried moving the .xml file for phone_1's extension to conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show: <groups> <group name="doublenat"> <users> <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/> </users> </group> Unfortunately this does not make a difference to the issue though. If anybody had the time to tell me what I am doing wrong I would be hugely grateful ! Many thanks in advance Bruce
Step 1. Revert everything back to SVN trunk. Step 2. Enable UPNP/NATPMP on both routers. Step 3. It should just work. The doublenat profile is NO LONGER NEEDED. If you can't enable UPNP/NATPMP then by all means static map the ports correctly or put the FreeSWITCH box in the DMZ on both ends. It should just work. /b On Mar 8, 2010, at 6:56 AM, Bruce Hopkins wrote:
Hi,
I wonder if some kind soul might be able please to help me by pointing out what I am doing wrong with my doublenat configuration.
I have incoming calls from WAN --> LAN working fine, but when I try to call from LAN -->WAN, the Freeswitch CLI says:
"[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]"
The setup is essentially:
phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT (public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z)
Phone_1 is able to register with Freeswitch using a "doublenat" sip profile I set up using port 5090, created from the default external profile with the port changed. phone_1 can initiate calls to phone_2 fine with no problems with audio - everything works just fine in that direction.
However when I try to call from phone_2 --> phone_1 I get the message "The person at extension 1002 is not available. Record your message at the tone." Meanwhile the CLI shows that Freeswitch says phone_1 is not registered.
In trying to solve this I have run a Wireshark trace onthe PC running the phone_1 softphone to check that the SIP registration seems to go as I understand it should. the dialog is as follows:
phone_1 --> FS request: REGISTER FS --> phone_1 status: 401 Unauthorised phone_1 --> FS request: REGISTER FS --> phone_1 status: 200 OK phone_1 --> FS request: SUBSCRIBE FS --> phone_1 status: 405 Method Not Allowed
There does appear to be a difference here to what happens when phone_2 (on the same LAN as Freeswitch) registers. With phone_2 I see:
phone_2 --> FS request: REGISTER FS --> phone_2 status: 401 Unauthorised phone_2 --> FS request: REGISTER FS --> phone_2 status 200 OK phone_2 --> FS request: SUBSCRIBE FS --> phone_2 status: 202 Accepted FS --> phone_2 request: NOTIFY phone_2 --> FS status: 200 OK
More configuration details:
In conf/sip_profiles/doublenat.xml I have not uncommented <param name="force-register-domain" value="$$(domain)"/>
In conf/sip_profiles/doublenat.xml I have tried setting the context to each of public and default. This does not seem to make a difference.
In conf/directory/doublenat.xml I have set the domain name to be the domain name registerd for Fs's external IP with DNS, and used by phone_1.
In conf/directory/doublenat.xml I have tried both of the following 9I don't know if this is relevant)
1. Firstly with phone_1 registering to an extension in the default configuration, held in conf/directory/default/:
<groups> <group name="default"> <users> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </users> </group>
2. Then I tried moving the .xml file for phone_1's extension to conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show:
<groups> <group name="doublenat"> <users> <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/> </users> </group>
Unfortunately this does not make a difference to the issue though.
If anybody had the time to tell me what I am doing wrong I would be hugely grateful !
Many thanks in advance Bruce _______________________________________________ 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
Many thanks for getting back to me with help so quickly Brian. I was using 1.0.5-20100302-0400 so will update to trunk and do as you suggest. thanks again Bruce On 8 March 2010 14:28, Brian West <brian@freeswitch.org> wrote:
Step 1. Revert everything back to SVN trunk. Step 2. Enable UPNP/NATPMP on both routers. Step 3. It should just work. The doublenat profile is NO LONGER NEEDED.
If you can't enable UPNP/NATPMP then by all means static map the ports correctly or put the FreeSWITCH box in the DMZ on both ends. It should just work.
/b
On Mar 8, 2010, at 6:56 AM, Bruce Hopkins wrote:
Hi,
I wonder if some kind soul might be able please to help me by pointing out what I am doing wrong with my doublenat configuration.
I have incoming calls from WAN --> LAN working fine, but when I try to call from LAN -->WAN, the Freeswitch CLI says:
"[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]"
The setup is essentially:
phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT (public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z)
Phone_1 is able to register with Freeswitch using a "doublenat" sip profile I set up using port 5090, created from the default external profile with the port changed. phone_1 can initiate calls to phone_2 fine with no problems with audio - everything works just fine in that direction.
However when I try to call from phone_2 --> phone_1 I get the message "The person at extension 1002 is not available. Record your message at the tone." Meanwhile the CLI shows that Freeswitch says phone_1 is not registered.
In trying to solve this I have run a Wireshark trace onthe PC running the phone_1 softphone to check that the SIP registration seems to go as I understand it should. the dialog is as follows:
phone_1 --> FS request: REGISTER FS --> phone_1 status: 401 Unauthorised phone_1 --> FS request: REGISTER FS --> phone_1 status: 200 OK phone_1 --> FS request: SUBSCRIBE FS --> phone_1 status: 405 Method Not Allowed
There does appear to be a difference here to what happens when phone_2 (on the same LAN as Freeswitch) registers. With phone_2 I see:
phone_2 --> FS request: REGISTER FS --> phone_2 status: 401 Unauthorised phone_2 --> FS request: REGISTER FS --> phone_2 status 200 OK phone_2 --> FS request: SUBSCRIBE FS --> phone_2 status: 202 Accepted FS --> phone_2 request: NOTIFY phone_2 --> FS status: 200 OK
More configuration details:
In conf/sip_profiles/doublenat.xml I have not uncommented <param name="force-register-domain" value="$$(domain)"/>
In conf/sip_profiles/doublenat.xml I have tried setting the context to each of public and default. This does not seem to make a difference.
In conf/directory/doublenat.xml I have set the domain name to be the domain name registerd for Fs's external IP with DNS, and used by phone_1.
In conf/directory/doublenat.xml I have tried both of the following 9I don't know if this is relevant)
1. Firstly with phone_1 registering to an extension in the default configuration, held in conf/directory/default/:
<groups> <group name="default"> <users> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </users> </group>
2. Then I tried moving the .xml file for phone_1's extension to conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show:
<groups> <group name="doublenat"> <users> <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/> </users> </group>
Unfortunately this does not make a difference to the issue though.
If anybody had the time to tell me what I am doing wrong I would be hugely grateful !
Many thanks in advance Bruce _______________________________________________ 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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Hi again, Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem. I updated to build 16938 by means of "make current" I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090. I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build. I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI: [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE. Any further ideas where I am going wrong here please? thanks again in advance Bruce On 8 March 2010 15:15, Brian West <brian@freeswitch.org> wrote:
Well sounds like you need to sync in your configs too.
/b
On Mar 8, 2010, at 9:03 AM, Bruce Hopkins wrote:
Many thanks for getting back to me with help so quickly Brian.
I was using 1.0.5-20100302-0400 so will update to trunk and do as you suggest.
thanks again Bruce
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ok you aren't catching one key thing here.. you no longer need two profiles. /b On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
Ah I see. I will try again using the internal profile and forwarding port 5060. Presumably still creating a directory entry to enable the outside-facing domain to be used. Many thanks for your patient help of a newbie Brian. Cheers Bruce Please excuse the brevity - sent from my mobile. On 8 Mar 2010, at 17:21, Brian West <brian@freeswitch.org> wrote:
ok you aren't catching one key thing here.. you no longer need two profiles.
/b
On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/ directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
_______________________________________________ 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
Ah I see... what? Please share and lead this blind man out the FS wilderness! I don't understand...what happens to the external profile? Do you delete it? And how do you forward port 5060? ...and you thought you were a newbie! Any insight would be much appreciated...loving the journey. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bruce Hopkins Sent: 09 March 2010 08:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] doublenat outgoing call issue Ah I see. I will try again using the internal profile and forwarding port 5060. Presumably still creating a directory entry to enable the outside-facing domain to be used. Many thanks for your patient help of a newbie Brian. Cheers Bruce Please excuse the brevity - sent from my mobile. On 8 Mar 2010, at 17:21, Brian West <brian@freeswitch.org> wrote:
ok you aren't catching one key thing here.. you no longer need two profiles.
/b
On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/ directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
_______________________________________________ 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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Hi Roly, I hope to be able to have a go at this later today. I'll let you know what happens. I have my FS box behind a hardware router/firewall, so I'm assuming I just forward SIP port 5060 from the WAN to the FS box and use internal profile, instead of using external profile with port 5080 forwarded as I have at the moment. We'll see what happens. cheers Bruce On 9 March 2010 19:08, Roly Maz <rm@callrica.co.za> wrote:
Ah I see... what? Please share and lead this blind man out the FS wilderness!
I don't understand...what happens to the external profile? Do you delete it? And how do you forward port 5060?
...and you thought you were a newbie!
Any insight would be much appreciated...loving the journey.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bruce Hopkins Sent: 09 March 2010 08:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] doublenat outgoing call issue
Ah I see. I will try again using the internal profile and forwarding port 5060. Presumably still creating a directory entry to enable the outside-facing domain to be used.
Many thanks for your patient help of a newbie Brian.
Cheers Bruce
Please excuse the brevity - sent from my mobile.
On 8 Mar 2010, at 17:21, Brian West <brian@freeswitch.org> wrote:
ok you aren't catching one key thing here.. you no longer need two profiles.
/b
On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/ directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
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Hey Roly, Are you getting anywhere? I'm still struggling, I'm afraid. I'm able to call from WAN --> LAN using the internal profile on port 5060, but I still can't call out from LAN --> WAN so there is obviously something really basic I'm still not getting. In sip_profiles/internal.xml I have: <param name="aggressive-nat-detection" value="true"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <!-- the external_rtp_ip etc being defined in vars.xml --> In this example I have left <param name="force-register-domain" value="$${domain}"/> etc uncommented to force the registration to go to default.xml in the dialplan. I thought maybe this was my problem and tried putting a separate file in the directory to enable the WAN phone to register using the external DNS name for the server as its domain, and uncommenting force-register-domain, etc. But then I got an error saying that the local (to Freeswitch) phone in the LAN was not registered when I tried to make the call from the WAN. Any thoughts? cheers Bruce On 10 March 2010 12:00, Bruce Hopkins <jbrucehopkins@gmail.com> wrote:
Hi Roly,
I hope to be able to have a go at this later today. I'll let you know what happens.
I have my FS box behind a hardware router/firewall, so I'm assuming I just forward SIP port 5060 from the WAN to the FS box and use internal profile, instead of using external profile with port 5080 forwarded as I have at the moment. We'll see what happens.
cheers Bruce
On 9 March 2010 19:08, Roly Maz <rm@callrica.co.za> wrote:
Ah I see... what? Please share and lead this blind man out the FS wilderness!
I don't understand...what happens to the external profile? Do you delete it? And how do you forward port 5060?
...and you thought you were a newbie!
Any insight would be much appreciated...loving the journey.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bruce Hopkins Sent: 09 March 2010 08:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] doublenat outgoing call issue
Ah I see. I will try again using the internal profile and forwarding port 5060. Presumably still creating a directory entry to enable the outside-facing domain to be used.
Many thanks for your patient help of a newbie Brian.
Cheers Bruce
Please excuse the brevity - sent from my mobile.
On 8 Mar 2010, at 17:21, Brian West <brian@freeswitch.org> wrote:
ok you aren't catching one key thing here.. you no longer need two profiles.
/b
On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/ directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
_______________________________________________ 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
Interestingly I *can* call outbound to the WAN a SIP client registered with FS via Fring. I wonder if Fring have some additional nat traversal/SBC malarkey going on as they proxy the calls. I can't call outbound to an Eyebeam softphone registered directly wth FS though - although said Eyebeam is registered and can call inwards. I must have my nat handling configured wrongly. B On 10 March 2010 21:28, Bruce Hopkins <jbrucehopkins@gmail.com> wrote:
Hey Roly,
Are you getting anywhere? I'm still struggling, I'm afraid.
I'm able to call from WAN --> LAN using the internal profile on port 5060, but I still can't call out from LAN --> WAN so there is obviously something really basic I'm still not getting.
In sip_profiles/internal.xml I have:
<param name="aggressive-nat-detection" value="true"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
<!-- the external_rtp_ip etc being defined in vars.xml -->
In this example I have left <param name="force-register-domain" value="$${domain}"/> etc uncommented to force the registration to go to default.xml in the dialplan.
I thought maybe this was my problem and tried putting a separate file in the directory to enable the WAN phone to register using the external DNS name for the server as its domain, and uncommenting force-register-domain, etc. But then I got an error saying that the local (to Freeswitch) phone in the LAN was not registered when I tried to make the call from the WAN.
Any thoughts?
cheers Bruce
On 10 March 2010 12:00, Bruce Hopkins <jbrucehopkins@gmail.com> wrote:
Hi Roly,
I hope to be able to have a go at this later today. I'll let you know what happens.
I have my FS box behind a hardware router/firewall, so I'm assuming I just forward SIP port 5060 from the WAN to the FS box and use internal profile, instead of using external profile with port 5080 forwarded as I have at the moment. We'll see what happens.
cheers Bruce
On 9 March 2010 19:08, Roly Maz <rm@callrica.co.za> wrote:
Ah I see... what? Please share and lead this blind man out the FS wilderness!
I don't understand...what happens to the external profile? Do you delete it? And how do you forward port 5060?
...and you thought you were a newbie!
Any insight would be much appreciated...loving the journey.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bruce Hopkins Sent: 09 March 2010 08:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] doublenat outgoing call issue
Ah I see. I will try again using the internal profile and forwarding port 5060. Presumably still creating a directory entry to enable the outside-facing domain to be used.
Many thanks for your patient help of a newbie Brian.
Cheers Bruce
Please excuse the brevity - sent from my mobile.
On 8 Mar 2010, at 17:21, Brian West <brian@freeswitch.org> wrote:
ok you aren't catching one key thing here.. you no longer need two profiles.
/b
On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/ directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
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OK - I have put a Wireshark trace across the router between FS and the internet, and it will not surprise you to hear that when I tried ot call from LAN-->WAN, FS was sending packets to the private IP of the remote phone (behind its own NAT). So how to sort this? I tried: 1. Commenting these out in the internal profile <!--all inbound reg will look in this domain for the users --> <!--<param name="force-register-domain" value="$${domain}"/>--> <!--force the domain in subscriptions to this value --> <!--<param name="force-subscription-domain" value="$${domain}"/>--> <!--all inbound reg will stored in the db using this domain --> <!--<param name="force-register-db-domain" value="$${domain}"/>--> 2. then creating a n antry in the directory refering to the same default group of extensions as the default.xml, but with: <domain name="dns.aname.for.FS"> and in the variables: <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> which I hoped from what I read in the Wiki might do the trick vis a vis forceing FS to send the SIP packets to the public IP of the remote phone. Alas it does not seem to work, and even phoning in from the WAS -->LAN now I see USER_NOT_REGISTERED errors . I get the same error in both directions. Please please please could somebody put me out of my (your ;-) ) misery here ....? What do I configure to tell FS to send outbound INVITE's to the public IP the remote phone registered from behind, rather than its private NAT'ed ip? many thanks in advance. Bruce On 10 March 2010 22:04, Bruce Hopkins <jbrucehopkins@gmail.com> wrote:
Interestingly I *can* call outbound to the WAN a SIP client registered with FS via Fring. I wonder if Fring have some additional nat traversal/SBC malarkey going on as they proxy the calls.
I can't call outbound to an Eyebeam softphone registered directly wth FS though - although said Eyebeam is registered and can call inwards.
I must have my nat handling configured wrongly.
B
On 10 March 2010 21:28, Bruce Hopkins <jbrucehopkins@gmail.com> wrote:
Hey Roly,
Are you getting anywhere? I'm still struggling, I'm afraid.
I'm able to call from WAN --> LAN using the internal profile on port 5060, but I still can't call out from LAN --> WAN so there is obviously something really basic I'm still not getting.
In sip_profiles/internal.xml I have:
<param name="aggressive-nat-detection" value="true"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
<!-- the external_rtp_ip etc being defined in vars.xml -->
In this example I have left <param name="force-register-domain" value="$${domain}"/> etc uncommented to force the registration to go to default.xml in the dialplan.
I thought maybe this was my problem and tried putting a separate file in the directory to enable the WAN phone to register using the external DNS name for the server as its domain, and uncommenting force-register-domain, etc. But then I got an error saying that the local (to Freeswitch) phone in the LAN was not registered when I tried to make the call from the WAN.
Any thoughts?
cheers Bruce
On 10 March 2010 12:00, Bruce Hopkins <jbrucehopkins@gmail.com> wrote:
Hi Roly,
I hope to be able to have a go at this later today. I'll let you know what happens.
I have my FS box behind a hardware router/firewall, so I'm assuming I just forward SIP port 5060 from the WAN to the FS box and use internal profile, instead of using external profile with port 5080 forwarded as I have at the moment. We'll see what happens.
cheers Bruce
On 9 March 2010 19:08, Roly Maz <rm@callrica.co.za> wrote:
Ah I see... what? Please share and lead this blind man out the FS wilderness!
I don't understand...what happens to the external profile? Do you delete it? And how do you forward port 5060?
...and you thought you were a newbie!
Any insight would be much appreciated...loving the journey.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bruce Hopkins Sent: 09 March 2010 08:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] doublenat outgoing call issue
Ah I see. I will try again using the internal profile and forwarding port 5060. Presumably still creating a directory entry to enable the outside-facing domain to be used.
Many thanks for your patient help of a newbie Brian.
Cheers Bruce
Please excuse the brevity - sent from my mobile.
On 8 Mar 2010, at 17:21, Brian West <brian@freeswitch.org> wrote:
ok you aren't catching one key thing here.. you no longer need two profiles.
/b
On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/ directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
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Hi Roly, I promised I'd let you know when I got it working - so here it is: As Brian said, doublenat is no longer needed. It looks like I must have just been confusing things horribly by using various things I'd read about, including: <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <param name="aggressive-nat-detection" value="true"/> These, and who knows what other mucking about I had been doing, must have confused everything as none of this is now necessary. All I did - after a reinstall - to get everything working through double NATs is: 1. In vars.xml <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<my.fs.ext.ip>"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=<my.fs.ext.ip>"/> 2. Then in prefix/sip_profiles/internal.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip)"/> Then as the man says, it just works. I decided not to enable options pings to keep the far-end NAT open to enable calling the remote phone - instead I configured it to send keep-alives. And that's it. I can't believe the knots I was tying myself in. It just works ! cheers Bruce On 9 March 2010 19:08, Roly Maz <rm@callrica.co.za> wrote:
Ah I see... what? Please share and lead this blind man out the FS wilderness!
I don't understand...what happens to the external profile? Do you delete it? And how do you forward port 5060?
...and you thought you were a newbie!
Any insight would be much appreciated...loving the journey.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bruce Hopkins Sent: 09 March 2010 08:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] doublenat outgoing call issue
Ah I see. I will try again using the internal profile and forwarding port 5060. Presumably still creating a directory entry to enable the outside-facing domain to be used.
Many thanks for your patient help of a newbie Brian.
Cheers Bruce
Please excuse the brevity - sent from my mobile.
On 8 Mar 2010, at 17:21, Brian West <brian@freeswitch.org> wrote:
ok you aren't catching one key thing here.. you no longer need two profiles.
/b
On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/ directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
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Its a common thing to try too hard. I personally put in countless hours to make this stuff JUST work. ;) Also if you're traversing the nat and you expect to talk to clients behind the nat and outside the nat on the same profile you will be required to prefix both ext-rtp-ip and ext-sip-ip with "autonat:<my.fs.ext.ip>" and make sure local-network-acl is set to localnet.auto... I think thats the one in the defaults too. This gives FreeSWITCH the ability to dynamically lie to the client about which IP is which so that the same profile can talk to clients on either side of the nat. /b On Mar 13, 2010, at 4:56 PM, Bruce Hopkins wrote:
And that's it. I can't believe the knots I was tying myself in. It just works !
Hi Bruce/Brian, Success on my side too! Thanks for sharing your experience. I too reset my FS and made minimal changes to my config. Voila! Anyone new to FS, and running it on Windows server- I strongly recommend you pull a copy of Wireshark off the net. It has great tools for analysing SIP/IP packets. I'll be adding my config and service provider setup to the wiki this week. Rgds From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bruce Hopkins Sent: 14 March 2010 12:57 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] doublenat outgoing call issue Hi Roly, I promised I'd let you know when I got it working - so here it is: As Brian said, doublenat is no longer needed. It looks like I must have just been confusing things horribly by using various things I'd read about, including: <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <param name="aggressive-nat-detection" value="true"/> These, and who knows what other mucking about I had been doing, must have confused everything as none of this is now necessary. All I did - after a reinstall - to get everything working through double NATs is: 1. In vars.xml <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<my.fs.ext.ip>"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=<my.fs.ext.ip>"/> 2. Then in prefix/sip_profiles/internal.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip)"/> Then as the man says, it just works. I decided not to enable options pings to keep the far-end NAT open to enable calling the remote phone - instead I configured it to send keep-alives. And that's it. I can't believe the knots I was tying myself in. It just works ! cheers Bruce On 9 March 2010 19:08, Roly Maz <rm@callrica.co.za> wrote: Ah I see... what? Please share and lead this blind man out the FS wilderness! I don't understand...what happens to the external profile? Do you delete it? And how do you forward port 5060? ...and you thought you were a newbie! Any insight would be much appreciated...loving the journey. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bruce Hopkins Sent: 09 March 2010 08:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] doublenat outgoing call issue Ah I see. I will try again using the internal profile and forwarding port 5060. Presumably still creating a directory entry to enable the outside-facing domain to be used. Many thanks for your patient help of a newbie Brian. Cheers Bruce Please excuse the brevity - sent from my mobile. On 8 Mar 2010, at 17:21, Brian West <brian@freeswitch.org> wrote:
ok you aren't catching one key thing here.. you no longer need two profiles.
/b
On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/ directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
_______________________________________________ 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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Doubles all round ! cheers Bruce On 14 March 2010 18:41, Roly Maz <rm@callrica.co.za> wrote:
Hi Bruce/Brian,
Success on my side too!
Thanks for sharing your experience. I too reset my FS and made minimal changes to my config. Voila!
Anyone new to FS, and running it on Windows server– I strongly recommend you pull a copy of Wireshark off the net. It has great tools for analysing SIP/IP packets.
I’ll be adding my config and service provider setup to the wiki this week.
Rgds
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Bruce Hopkins *Sent:* 14 March 2010 12:57 AM
*To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] doublenat outgoing call issue
Hi Roly,
I promised I'd let you know when I got it working - so here it is:
As Brian said, doublenat is no longer needed.
It looks like I must have just been confusing things horribly by using various things I'd read about, including:
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
<param name="aggressive-nat-detection" value="true"/>
These, and who knows what other mucking about I had been doing, must have confused everything as none of this is now necessary.
All I did - after a reinstall - to get everything working through double NATs is:
1. In vars.xml
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=<my.fs.ext.ip>"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=<my.fs.ext.ip>"/>
2. Then in prefix/sip_profiles/internal.xml
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip)"/>
Then as the man says, it just works.
I decided not to enable options pings to keep the far-end NAT open to enable calling the remote phone - instead I configured it to send keep-alives.
And that's it. I can't believe the knots I was tying myself in. It just works !
cheers
Bruce
On 9 March 2010 19:08, Roly Maz <rm@callrica.co.za> wrote:
Ah I see... what? Please share and lead this blind man out the FS wilderness!
I don't understand...what happens to the external profile? Do you delete it? And how do you forward port 5060?
...and you thought you were a newbie!
Any insight would be much appreciated...loving the journey.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bruce Hopkins Sent: 09 March 2010 08:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] doublenat outgoing call issue
Ah I see. I will try again using the internal profile and forwarding port 5060. Presumably still creating a directory entry to enable the outside-facing domain to be used.
Many thanks for your patient help of a newbie Brian.
Cheers Bruce
Please excuse the brevity - sent from my mobile.
On 8 Mar 2010, at 17:21, Brian West <brian@freeswitch.org> wrote:
ok you aren't catching one key thing here.. you no longer need two profiles.
/b
On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
Hi again,
Thanks for the help so far. Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
I updated to build 16938 by means of "make current"
I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/ directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on. I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
Any further ideas where I am going wrong here please?
thanks again in advance Bruce
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Brian West <brian@freeswitch.org> wrote:
Its nice to hear more people have this working. The more that depend on this .. the simpler my job is... because If we break it they start to open jira's about it.
If both parties behind NAT could just go and register with http://tunelbroker.net/ or http://www.freenet6.net/ or http://www.sixxs.net/ or a similar service, or lobby their ISPs to introduce native IPv6, this problem would start to go away.
Very interesting. Thanks for sharing this. I must read up on it. cheersBruce On 15 March 2010 02:17, Jason White <jason@jasonjgw.net> wrote:
Brian West <brian@freeswitch.org> wrote:
Its nice to hear more people have this working. The more that depend on this .. the simpler my job is... because If we break it they start to open jira's about it.
If both parties behind NAT could just go and register with http://tunelbroker.net/ or http://www.freenet6.net/ or http://www.sixxs.net/ or a similar service, or lobby their ISPs to introduce native IPv6, this problem would start to go away.
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participants (4)
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Brian West
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Bruce Hopkins
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Jason White
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Roly Maz