Hello I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -> Gateway And I'm struggling to get NAT working properly. I'm running freeswitch with the "-nonat" option and have tried different ext-rtp-ip/ext-sip-ip combinations in external/internal profiles. The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly. Currently I have this setup: API CALL [sofia(status profile external)] output: ======================================================== Name external Domain Name N/A Context public Challenge Realm auto_to RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 Ext-SIP-IP 85.89.XX.XX OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false API CALL [sofia(status profile default)] output: ======================================================== Name default Domain Name N/A Alias Of internal Context public Challenge Realm auto_from RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED false STUN-AUTO-DISABLE false Sample phone registration: Call-ID: Xmbw9PyQ5Q6L2MnQ@192.168.1.121 User: u1000009@default Contact: "u1000009" <sip:u1000009@192.168.1.121:6094> Agent: IP PHONE 3 V1.58.004 CFG0 Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40) Host: jonas-PC IP: 192.168.1.121 Port: 6094 Auth-User: u1000009 Auth-Realm: default MWI-Account: u1000009@default Outbound INVITE: send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000: ------------------------------------------------------------------------ INVITE sip:0706930XXX@sipgw2.XXXXX.se <sip%3A0706930XXX@sipgw2.XXXXX.se>SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp Max-Forwards: 69 From: "Kundtjänst Arne" <sip:0500650XXX@85.89.XX.XX>;tag=B7pve7F6eeH7c To: <sip:0706930821@sipgw2.XXXXX.se <sip%3A0706930821@sipgw2.XXXXX.se>> Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 CSeq: 123379614 INVITE Contact: <sip:mod_sofia@192.168.1.110:5060> Call-Info: <answer-after=400> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 293 X-FS-Support: update_display Remote-Party-ID: "Kundtjänst Arne" <sip:0500650XXX@85.89.XX.XX
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110 s=FreeSWITCH c=IN IP4 192.168.1.110 t=0 0 m=audio 24986 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Many thanks, Jonas
You set the ext-rtp-ip on the profile the phones talk too... but you shouldn't be doing that. /b On Nov 23, 2009, at 11:08 AM, Jonas Gauffin wrote:
Hello
I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -
Gateway
And I'm struggling to get NAT working properly. I'm running freeswitch with the "-nonat" option and have tried different ext-rtp- ip/ext-sip-ip combinations in external/internal profiles. The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly.
Currently I have this setup:
Ok. Found the problem. I had started using "sofia/outbound/ XXXXXX@sipgw2.XXXX.se" as bridge destination to try to get outbound_caller_id_name/outbound_caller_id_number working. It works if I use the correct profile name, "sofia/internal/ XXXXXX@sipgw2.XXXX.se" When do FS use outbound_caller_id instead of effective_caller_id? On Mon, Nov 23, 2009 at 6:08 PM, Jonas Gauffin <jonas.gauffin@gmail.com>wrote:
Hello
I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -> Gateway
And I'm struggling to get NAT working properly. I'm running freeswitch with the "-nonat" option and have tried different ext-rtp-ip/ext-sip-ip combinations in external/internal profiles. The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly.
Currently I have this setup:
API CALL [sofia(status profile external)] output: ======================================================== Name external Domain Name N/A Context public Challenge Realm auto_to RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 Ext-SIP-IP 85.89.XX.XX OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false
API CALL [sofia(status profile default)] output: ======================================================== Name default Domain Name N/A Alias Of internal Context public Challenge Realm auto_from RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED false STUN-AUTO-DISABLE false
Sample phone registration: Call-ID: Xmbw9PyQ5Q6L2MnQ@192.168.1.121 User: u1000009@default Contact: "u1000009" <sip:u1000009@192.168.1.121:6094> Agent: IP PHONE 3 V1.58.004 CFG0 Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40) Host: jonas-PC IP: 192.168.1.121 Port: 6094 Auth-User: u1000009 Auth-Realm: default MWI-Account: u1000009@default
Outbound INVITE: send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000: ------------------------------------------------------------------------ INVITE sip:0706930XXX@sipgw2.XXXXX.se<sip%3A0706930XXX@sipgw2.XXXXX.se>SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp Max-Forwards: 69 From: "Kundtjänst Arne" <sip:0500650XXX@85.89.XX.XX>;tag=B7pve7F6eeH7c To: <sip:0706930821@sipgw2.XXXXX.se <sip%3A0706930821@sipgw2.XXXXX.se>> Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 CSeq: 123379614 INVITE Contact: <sip:mod_sofia@192.168.1.110:5060> Call-Info: <answer-after=400> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 293 X-FS-Support: update_display Remote-Party-ID: "Kundtjänst Arne" <sip:0500650XXX@85.89.XX.XX
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110 s=FreeSWITCH c=IN IP4 192.168.1.110 t=0 0 m=audio 24986 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
Many thanks, Jonas
outbound_caller_id is a made up variable that is used in the defaults that are used in the examples only. /b On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote:
Ok. Found the problem. I had started using "sofia/outbound/XXXXXX@sipgw2.XXXX.se " as bridge destination to try to get outbound_caller_id_name/ outbound_caller_id_number working. It works if I use the correct profile name, "sofia/internal/XXXXXX@sipgw2.XXXX.se "
When do FS use outbound_caller_id instead of effective_caller_id?
Ok. It would be a nice feature if outbound_caller_id was used by freeswitch. I do quite often bridge to both internal and external destinations in the same bridge command (as in "sofia/internal/5530,sofia/internal/ 070123456@sipgw2.XXXX.se). This forces me to always use complete phone numbers in the caller id since my gateway would reject the call otherwise. It would be really neat if FS could use effective_caller_id (5531) for the internal bridge and outbound_caller_id (+4681235531) for the external bridge. On Mon, Nov 23, 2009 at 6:31 PM, Brian West <brian@freeswitch.org> wrote:
outbound_caller_id is a made up variable that is used in the defaults that are used in the examples only.
/b
On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote:
Ok. Found the problem. I had started using "sofia/outbound/ XXXXXX@sipgw2.XXXX.se" as bridge destination to try to get outbound_caller_id_name/outbound_caller_id_number working. It works if I use the correct profile name, "sofia/internal/ XXXXXX@sipgw2.XXXX.se"
When do FS use outbound_caller_id instead of effective_caller_id?
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See default config it lsets you do that. Use the variables to store two versions of the callerid then set it depending on if its outside or inside... its rather easy to do. /b On Nov 23, 2009, at 11:50 AM, Jonas Gauffin wrote:
Ok. It would be a nice feature if outbound_caller_id was used by freeswitch. I do quite often bridge to both internal and external destinations in the same bridge command (as in "sofia/internal/5530,sofia/internal/070123456@sipgw2.XXXX.se ). This forces me to always use complete phone numbers in the caller id since my gateway would reject the call otherwise.
It would be really neat if FS could use effective_caller_id (5531) for the internal bridge and outbound_caller_id (+4681235531) for the external bridge.
participants (2)
-
Brian West
-
Jonas Gauffin