From what I am seeing, freeswitch is not honoring the ext-*-ip variables in the invite messages. Using the following command entered on fs_cli: originate {origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212 &javascript(somejavascript.js), the invite message is below. Shouldn't the instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*? This is causing the rtp packets to be sent to the incorrect location, and resulting in 1-way audio.
send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988: ------------------------------------------------------------------------ INVITE sip:13044150838<https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.* SIP/2.0 Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD Max-Forwards: 69 From: "SomeName" <sip:bryan_bryanlemon_com@209.216. *.*;transport=udp>;tag=HpU27XSQHmX1g To: <sip:15555551212@209.216.*.*> Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a CSeq: 16401016 INVITE Contact: <sip:gw+didforsale@204.111.*.*:5080;transport=udp;gw=didforsale> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11 14-15-26 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display Remote-Party-ID: "SomeName" <sip:15555551212@10.0.10.144
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144 s=FreeSWITCH c=IN IP4 10.0.10.144 t=0 0 m=audio 32712 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 10.0.10.144,10.0.10.144 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5060 BIND-URL sip:mod_sofia@10.0.10.144:5060 freeswitch@internal> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5080 BIND-URL sip:mod_sofia@10.0.10.144:5080 Thank you, Bryan Lemon (302) 648-2747 <https://www.google.com/voice/m/caller?number=+13026482747>
Bryan, Can you provide the sofia profile xml? /b On Aug 16, 2011, at 8:46 AM, Bryan Lemon wrote:
From what I am seeing, freeswitch is not honoring the ext-*-ip variables in the invite messages. Using the following command entered on fs_cli: originate {origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212 &javascript(somejavascript.js), the invite message is below. Shouldn't the instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*? This is causing the rtp packets to be sent to the incorrect location, and resulting in 1-way audio.
send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988: ------------------------------------------------------------------------ INVITE sip:13044150838<https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.* SIP/2.0 Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD Max-Forwards: 69 From: "SomeName" <sip:bryan_bryanlemon_com@209.216. *.*;transport=udp>;tag=HpU27XSQHmX1g To: <sip:15555551212@209.216.*.*> Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a CSeq: 16401016 INVITE Contact: <sip:gw+didforsale@204.111.*.*:5080;transport=udp;gw=didforsale> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11 14-15-26 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display Remote-Party-ID: "SomeName" <sip:15555551212@10.0.10.144
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144 s=FreeSWITCH c=IN IP4 10.0.10.144 t=0 0 m=audio 32712 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 10.0.10.144,10.0.10.144 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5060 BIND-URL sip:mod_sofia@10.0.10.144:5060
freeswitch@internal> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5080 BIND-URL sip:mod_sofia@10.0.10.144:5080
Thank you, Bryan Lemon (302) 648-2747 <https://www.google.com/voice/m/caller?number=+13026482747>
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
<profile name="external"> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <!-- <alias name="outbound"/> <alias name="nat"/> --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <!--<param name="aggressive-nat-detection" value="true"/>--> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="$${external_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${external_tls_port}"/> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile> Thank you, Bryan Lemon (302) 648-2747 On Tue, Aug 16, 2011 at 13:03, Brian West <brian@freeswitch.org> wrote:
Bryan, Can you provide the sofia profile xml?
/b
On Aug 16, 2011, at 8:46 AM, Bryan Lemon wrote:
From what I am seeing, freeswitch is not honoring the ext-*-ip variables in the invite messages. Using the following command entered on fs_cli: originate
{origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212
&javascript(somejavascript.js), the invite message is below. Shouldn't the instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*? This is causing the rtp packets to be sent to the incorrect location, and resulting in 1-way audio.
send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988:
------------------------------------------------------------------------
INVITE sip:13044150838< https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.* SIP/2.0 Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD Max-Forwards: 69 From: "SomeName" <sip:bryan_bryanlemon_com@209.216. *.*;transport=udp>;tag=HpU27XSQHmX1g To: <sip:15555551212@209.216.*.*> Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a CSeq: 16401016 INVITE Contact: <sip:gw+didforsale@204.111. *.*:5080;transport=udp;gw=didforsale> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11 14-15-26 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display Remote-Party-ID: "SomeName" <sip:15555551212@10.0.10.144
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144 s=FreeSWITCH c=IN IP4 10.0.10.144 t=0 0 m=audio 32712 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
sofia status profile internal
=================================================================================================
Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 10.0.10.144,10.0.10.144 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5060 BIND-URL sip:mod_sofia@10.0.10.144:5060
freeswitch@internal> sofia status profile external
=================================================================================================
Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5080 BIND-URL sip:mod_sofia@10.0.10.144:5080
Thank you, Bryan Lemon (302) 648-2747 < https://www.google.com/voice/m/caller?number=+13026482747>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
<profile name="external"> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <!-- <alias name="outbound"/> <alias name="nat"/> --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <!--<param name="aggressive-nat-detection" value="true"/>--> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="$${external_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${external_tls_port}"/> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile> Thank you, Bryan Lemon (302) 648-2747 On Tue, Aug 16, 2011 at 13:03, Brian West <brian@freeswitch.org> wrote:
Bryan, Can you provide the sofia profile xml?
/b
On Aug 16, 2011, at 8:46 AM, Bryan Lemon wrote:
From what I am seeing, freeswitch is not honoring the ext-*-ip variables in the invite messages. Using the following command entered on fs_cli: originate
{origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212
&javascript(somejavascript.js), the invite message is below. Shouldn't the instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*? This is causing the rtp packets to be sent to the incorrect location, and resulting in 1-way audio.
send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988:
------------------------------------------------------------------------
INVITE sip:13044150838< https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.* SIP/2.0 Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD Max-Forwards: 69 From: "SomeName" <sip:bryan_bryanlemon_com@209.216. *.*;transport=udp>;tag=HpU27XSQHmX1g To: <sip:15555551212@209.216.*.*> Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a CSeq: 16401016 INVITE Contact: <sip:gw+didforsale@204.111. *.*:5080;transport=udp;gw=didforsale> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11 14-15-26 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display Remote-Party-ID: "SomeName" <sip:15555551212@10.0.10.144
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144 s=FreeSWITCH c=IN IP4 10.0.10.144 t=0 0 m=audio 32712 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
sofia status profile internal
=================================================================================================
Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 10.0.10.144,10.0.10.144 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5060 BIND-URL sip:mod_sofia@10.0.10.144:5060
freeswitch@internal> sofia status profile external
=================================================================================================
Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5080 BIND-URL sip:mod_sofia@10.0.10.144:5080
Thank you, Bryan Lemon (302) 648-2747 < https://www.google.com/voice/m/caller?number=+13026482747>
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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I tried it, and it still does not work. Relevant details: freeswitch@MEDIAPC> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5080 BIND-URL sip:mod_sofia@10.0.10.144:5080 HOLD-MUSIC local_stream://moh external.xml <profile name="external"> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <!-- <alias name="outbound"/> <alias name="nat"/> --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <!--<param name="aggressive-nat-detection" value="true"/>--> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="autonat:204.111.187.118"/> <param name="ext-sip-ip" value="autonat:204.111.187.118"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="$${external_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${external_tls_port}"/> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile> Thank you, Bryan Lemon (302) 648-2747 On Wed, Aug 17, 2011 at 12:55, Brian West <brian@freeswitch.org> wrote:
You need to change these to be "autonat:x.x.x.x" and specify the IP to use.
/b
On Aug 17, 2011, at 10:10 AM, Bryan Lemon wrote:
<param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/>
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
Any other input/ideas on this one? Thank you, Bryan Lemon (302) 648-2747 On Wed, Aug 17, 2011 at 23:08, Bryan Lemon <bryan@bryanlemon.com> wrote:
I tried it, and it still does not work.
Relevant details: freeswitch@MEDIAPC> sofia status profile external
================================================================================================= Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5080 BIND-URL sip:mod_sofia@10.0.10.144:5080 HOLD-MUSIC local_stream://moh
external.xml <profile name="external"> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways>
<aliases> <!--
<alias name="outbound"/>
<alias name="nat"/>
--> </aliases>
<domains> <domain name="all" alias="false" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <!--<param name="aggressive-nat-detection" value="true"/>--> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="autonat:204.111.187.118"/> <param name="ext-sip-ip" value="autonat:204.111.187.118"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="$${external_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${external_tls_port}"/> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/>
</settings> </profile>
Thank you, Bryan Lemon (302) 648-2747
On Wed, Aug 17, 2011 at 12:55, Brian West <brian@freeswitch.org> wrote:
You need to change these to be "autonat:x.x.x.x" and specify the IP to use.
/b
On Aug 17, 2011, at 10:10 AM, Bryan Lemon wrote:
<param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/>
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
Just a standard dd-wrt router. Thank you, Bryan Lemon (302) 648-2747 On Mon, Aug 22, 2011 at 13:27, Brian West <brian@freeswitch.org> wrote:
What kind of nat / firewall are you traversing?
/b
On Aug 22, 2011, at 10:20 AM, Bryan Lemon wrote:
Any other input/ideas on this one?
Thank you, Bryan Lemon (302) 648-2747
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'm using dd-wrt and it's working with auto-nat. did u enable UPnP on the router? -nandy On Tue, Aug 23, 2011 at 1:48 AM, Bryan Lemon <bryan@bryanlemon.com> wrote:
Just a standard dd-wrt router.
Thank you, Bryan Lemon (302) 648-2747
On Mon, Aug 22, 2011 at 13:27, Brian West <brian@freeswitch.org> wrote:
What kind of nat / firewall are you traversing?
/b
On Aug 22, 2011, at 10:20 AM, Bryan Lemon wrote:
Any other input/ideas on this one?
Thank you, Bryan Lemon (302) 648-2747
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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if you know the exact value of the ip use that in place of the word "auto-nat" which is specifically designed for nat-pnp enabled routers. On Tue, Aug 16, 2011 at 12:07 PM, Bryan Lemon <bryan@bryanlemon.com> wrote:
<profile name="external"> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <!--
<alias name="outbound"/>
<alias name="nat"/>
--> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <!--<param name="aggressive-nat-detection" value="true"/>--> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="$${external_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${external_tls_port}"/> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile>
Thank you, Bryan Lemon (302) 648-2747
On Tue, Aug 16, 2011 at 13:03, Brian West <brian@freeswitch.org> wrote:
Bryan, Can you provide the sofia profile xml?
/b
On Aug 16, 2011, at 8:46 AM, Bryan Lemon wrote:
From what I am seeing, freeswitch is not honoring the ext-*-ip variables in the invite messages. Using the following command entered on fs_cli: originate
{origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212 &javascript(somejavascript.js), the invite message is below. Shouldn't the instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*? This is causing the rtp packets to be sent to the incorrect location, and resulting in 1-way audio.
send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988:
------------------------------------------------------------------------ INVITE sip:13044150838<https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.* SIP/2.0 Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD Max-Forwards: 69 From: "SomeName" <sip:bryan_bryanlemon_com@209.216. *.*;transport=udp>;tag=HpU27XSQHmX1g To: <sip:15555551212@209.216.*.*> Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a CSeq: 16401016 INVITE Contact: <sip:gw+didforsale@204.111.*.*:5080;transport=udp;gw=didforsale> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11 14-15-26 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display Remote-Party-ID: "SomeName" <sip:15555551212@10.0.10.144
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144 s=FreeSWITCH c=IN IP4 10.0.10.144 t=0 0 m=audio 32712 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
sofia status profile internal
================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 10.0.10.144,10.0.10.144 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5060 BIND-URL sip:mod_sofia@10.0.10.144:5060
freeswitch@internal> sofia status profile external
================================================================================================= Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5080 BIND-URL sip:mod_sofia@10.0.10.144:5080
Thank you, Bryan Lemon (302) 648-2747 <https://www.google.com/voice/m/caller?number=+13026482747>
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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (4)
-
Anthony Minessale
-
Brian West
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Bryan Lemon
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Nandy Dagondon