Re: [Freeswitch-users] Can I get SIP DID working?
Yeah, the call arrives nicely now, but the audio is only 1-way. I have port forwarded all the ports in my router to FreeSwitch, but still experiences just oneway audio, from FreeSwitch to the DID caller. Looks like the RTP is not forwarded correctly to FreeSwitch. Could it be my Linksys router, which is not forwarding RTP ports correctly, or is there still a piece in FreeSwitch I have missed? Ivan Den 20. juni. 2008 kl. 22:52 skrev Brian West: Good to know it snapped into place now! :P /b On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote: This is all making sense to me now. ___ 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
Re: [Freeswitch-users] Can I get SIP DID working?
I put this into internal.xml, and this seems to do the trick: param name=ext-rtp-ip value=$${external_rtp_ip}/ param name=ext-sip-ip value=$${external_sip_ip}/ Ivan Den 21. juni. 2008 kl. 15:47 skrev Ivan C Myrvold: Yeah, the call arrives nicely now, but the audio is only 1-way. I have port forwarded all the ports in my router to FreeSwitch, but still experiences just oneway audio, from FreeSwitch to the DID caller. Looks like the RTP is not forwarded correctly to FreeSwitch. Could it be my Linksys router, which is not forwarding RTP ports correctly, or is there still a piece in FreeSwitch I have missed? Ivan Den 20. juni. 2008 kl. 22:52 skrev Brian West: Good to know it snapped into place now! :P /b On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote: This is all making sense to me now. ___ 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 ___ 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
Re: [Freeswitch-users] Can I get SIP DID working?
Let me guess.. you have phones behind the same NAT talking to this profile also? If that is the case you'll need to create a new profile with ext-sip- ip and ext-rtp-ip set for the outside world to talk to.. then take those settings off the profile your phones behind the nat talk to freeswitch. /b On Jun 21, 2008, at 10:56 AM, Ivan C Myrvold wrote: I put this into internal.xml, and this seems to do the trick: param name=ext-rtp-ip value=$${external_rtp_ip}/ param name=ext-sip-ip value=$${external_sip_ip}/ Ivan Den 21. juni. 2008 kl. 15:47 skrev Ivan C Myrvold: Yeah, the call arrives nicely now, but the audio is only 1-way. I have port forwarded all the ports in my router to FreeSwitch, but still experiences just oneway audio, from FreeSwitch to the DID caller. Looks like the RTP is not forwarded correctly to FreeSwitch. Could it be my Linksys router, which is not forwarding RTP ports correctly, or is there still a piece in FreeSwitch I have missed? Ivan Den 20. juni. 2008 kl. 22:52 skrev Brian West: Good to know it snapped into place now! :P /b On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote: This is all making sense to me now. ___ 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 ___ 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
Re: [Freeswitch-users] Can I get SIP DID working?
http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios Please review Scenario 1 the bold text. ;) /b On Jun 21, 2008, at 10:56 AM, Ivan C Myrvold wrote: I put this into internal.xml, and this seems to do the trick: param name=ext-rtp-ip value=$${external_rtp_ip}/ param name=ext-sip-ip value=$${external_sip_ip}/ Ivan Den 21. juni. 2008 kl. 15:47 skrev Ivan C Myrvold: Yeah, the call arrives nicely now, but the audio is only 1-way. I have port forwarded all the ports in my router to FreeSwitch, but still experiences just oneway audio, from FreeSwitch to the DID caller. Looks like the RTP is not forwarded correctly to FreeSwitch. Could it be my Linksys router, which is not forwarding RTP ports correctly, or is there still a piece in FreeSwitch I have missed? Ivan Den 20. juni. 2008 kl. 22:52 skrev Brian West: Good to know it snapped into place now! :P /b On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote: This is all making sense to me now. ___ 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 ___ 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
Re: [Freeswitch-users] Can I get SIP DID working?
As I understand the ACL, it only controls which external machines with the IP address range given in the acl.conf.xml are allowed into FreeSwitch. So this doesn't help me much with the DID, as FreeSwitch sends a Proxy Authentication Required SIP message back to the Voxbone server. How can I configure FreeSwitch to allow the call into FreeSwitch without proxy authentication? Ivan Den 18. juni. 2008 kl. 09:25 skrev Ivan C Myrvold: I added to my acl.conf.xml this http://pastebin.freeswitch.org/4640 and this to external.xml param name=apply-inbound-acl value=voxbone/ When I start freeswitch, I see that the settings are used: 2008-06-18 09:11:09 [CONSOLE] switch_core.c:769 switch_load_network_lists() Created ip list voxbone default (deny) 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791 switch_load_network_lists() Adding 81.201.82.0/24 (allow) to list voxbone 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791 switch_load_network_lists() Adding 81.201.86.0/24 (allow) to list voxbone 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791 switch_load_network_lists() Adding 81.201.83.0/24 (allow) to list voxbone 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791 switch_load_network_lists() Adding 81.201.84.0/24 (allow) to list voxbone But still I get the same result as before: http://pastebin.freeswitch.org/4639 The list of IP addresses are at http://www.voxbone.com/members/faq-technical.jsf , and I am not absolutely sure I put this into the acl.conf.xml list correctly? Ivan Den 17. juni. 2008 kl. 16:08 skrev Brian West: You'll need to setup an ACL to let them in without authentication. Look at the Asterisk to FreeSWITCH section on the wiki. http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk /b On Jun 17, 2008, at 12:31 AM, Ivan C Myrvold wrote: I have used Freeswitch with a DID from Voxbone working on IAX, and that have been working so well. But now Voxbone will discontinue the IAX service, so I have to get DID working on SIP, and I am wondering if that will work at all in my configuration. Freeswitch is behind nat, so when I get a call from Voxbone, see http://pastebin.freeswitch.org/4629 , it will come in on port 5060. Have you covered this situation in the documentation? Ivan ___ 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 Brian West sip:[EMAIL PROTECTED] ___ 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 ___ 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
Re: [Freeswitch-users] Can I get SIP DID working?
Ivan, Do you have your outbound registration to voxbone on the default/ internal profile? If so then your acl's might be wrong. The one sure fire way to do this is to just setup another profile without auth on a different port and run with that. /b On Jun 20, 2008, at 3:20 AM, Ivan C Myrvold wrote: As I understand the ACL, it only controls which external machines with the IP address range given in the acl.conf.xml are allowed into FreeSwitch. So this doesn't help me much with the DID, as FreeSwitch sends a Proxy Authentication Required SIP message back to the Voxbone server. How can I configure FreeSwitch to allow the call into FreeSwitch without proxy authentication? Ivan Brian West sip:[EMAIL PROTECTED] ___ 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
Re: [Freeswitch-users] Can I get SIP DID working?
Let me guess they only send to port 5060? If you have your ACL's setup correctly those two IP's will be let in without auth. If you have the profile that runs on 5060 on your FreeSWITCH box with auth-calls=true Add this param name=apply-inbound-acl value=voxbone/ to that profile list name=voxbone default=deny node type=allow cidr=81.201.82.20/32/ node type=allow cidr=81.201.82.21/32/ /list This should allow them thru without auth. If you have problems with that add 81.201.82.0/24 to the list and see if that makes a difference. /b On Jun 20, 2008, at 11:28 AM, Ivan C Myrvold wrote: I do not have outbound registation to Voxbone, because Voxbone is only incoming. I am not registrating Voxbone at all. In their FAQ, they have how to configure for Asterisk: [81.201.82.20] host = 81.201.82.20 type = friend insecure = very context = your-context canreinvite=no [81.201.82.21] host = 81.201.82.21 type = friend insecure = very context = your-context canreinvite=no ... ( Do it for all voxbone IPs address listed above ) Den 20. juni. 2008 kl. 17:33 skrev Brian West: Ivan, Do you have your outbound registration to voxbone on the default/ internal profile? If so then your acl's might be wrong. The one sure fire way to do this is to just setup another profile without auth on a different port and run with that. /b On Jun 20, 2008, at 3:20 AM, Ivan C Myrvold wrote: As I understand the ACL, it only controls which external machines with the IP address range given in the acl.conf.xml are allowed into FreeSwitch. So this doesn't help me much with the DID, as FreeSwitch sends a Proxy Authentication Required SIP message back to the Voxbone server. How can I configure FreeSwitch to allow the call into FreeSwitch without proxy authentication? Ivan Brian West sip:[EMAIL PROTECTED] ___ 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 ___ 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
Re: [Freeswitch-users] Can I get SIP DID working?
Ah, I put the apply-inbound-acl in the wrong XML file. I put it in the external.xml. When I instead put it in the internal.xml, I got it working. That is of course the correct place, because it is bound to port 5060, and the inbound from Voxbone comes in on port 5060, as you correctly guessed. This is all making sense to me now. Ivan Den 20. juni. 2008 kl. 19:17 skrev Brian West: Let me guess they only send to port 5060? If you have your ACL's setup correctly those two IP's will be let in without auth. If you have the profile that runs on 5060 on your FreeSWITCH box with auth-calls=true Add this param name=apply-inbound-acl value=voxbone/ to that profile list name=voxbone default=deny node type=allow cidr=81.201.82.20/32/ node type=allow cidr=81.201.82.21/32/ /list This should allow them thru without auth. If you have problems with that add 81.201.82.0/24 to the list and see if that makes a difference. /b On Jun 20, 2008, at 11:28 AM, Ivan C Myrvold wrote: I do not have outbound registation to Voxbone, because Voxbone is only incoming. I am not registrating Voxbone at all. In their FAQ, they have how to configure for Asterisk: [81.201.82.20] host = 81.201.82.20 type = friend insecure = very context = your-context canreinvite=no [81.201.82.21] host = 81.201.82.21 type = friend insecure = very context = your-context canreinvite=no ... ( Do it for all voxbone IPs address listed above ) Den 20. juni. 2008 kl. 17:33 skrev Brian West: Ivan, Do you have your outbound registration to voxbone on the default/ internal profile? If so then your acl's might be wrong. The one sure fire way to do this is to just setup another profile without auth on a different port and run with that. /b On Jun 20, 2008, at 3:20 AM, Ivan C Myrvold wrote: As I understand the ACL, it only controls which external machines with the IP address range given in the acl.conf.xml are allowed into FreeSwitch. So this doesn't help me much with the DID, as FreeSwitch sends a Proxy Authentication Required SIP message back to the Voxbone server. How can I configure FreeSwitch to allow the call into FreeSwitch without proxy authentication? Ivan Brian West sip:[EMAIL PROTECTED] ___ 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 ___ 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
Re: [Freeswitch-users] Can I get SIP DID working?
Good to know it snapped into place now! :P /b On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote: This is all making sense to me now. ___ 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
Re: [Freeswitch-users] Can I get SIP DID working?
I added to my acl.conf.xml this http://pastebin.freeswitch.org/4640 and this to external.xml param name=apply-inbound-acl value=voxbone/ When I start freeswitch, I see that the settings are used: 2008-06-18 09:11:09 [CONSOLE] switch_core.c:769 switch_load_network_lists() Created ip list voxbone default (deny) 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791 switch_load_network_lists() Adding 81.201.82.0/24 (allow) to list voxbone 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791 switch_load_network_lists() Adding 81.201.86.0/24 (allow) to list voxbone 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791 switch_load_network_lists() Adding 81.201.83.0/24 (allow) to list voxbone 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791 switch_load_network_lists() Adding 81.201.84.0/24 (allow) to list voxbone But still I get the same result as before: http://pastebin.freeswitch.org/4639 The list of IP addresses are at http://www.voxbone.com/members/faq-technical.jsf , and I am not absolutely sure I put this into the acl.conf.xml list correctly? Ivan Den 17. juni. 2008 kl. 16:08 skrev Brian West: You'll need to setup an ACL to let them in without authentication. Look at the Asterisk to FreeSWITCH section on the wiki. http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk /b On Jun 17, 2008, at 12:31 AM, Ivan C Myrvold wrote: I have used Freeswitch with a DID from Voxbone working on IAX, and that have been working so well. But now Voxbone will discontinue the IAX service, so I have to get DID working on SIP, and I am wondering if that will work at all in my configuration. Freeswitch is behind nat, so when I get a call from Voxbone, see http://pastebin.freeswitch.org/4629 , it will come in on port 5060. Have you covered this situation in the documentation? Ivan ___ 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 Brian West sip:[EMAIL PROTECTED] ___ 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