From GeorgePhelps at gfphelps.com Thu Jan 1 00:37:17 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Wed, 31 Dec 2014 16:37:17 -0500 Subject: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable In-Reply-To: References: <040101d01f98$34ca7d40$9e5f77c0$@gfphelps.com> <052901d0211c$54c93de0$fe5bb9a0$@gfphelps.com> <05b601d022a9$a8859ad0$f990d070$@gfphelps.com> <05e301d022e8$f5be9790$e13bc6b0$@gfphelps.com> <05fe01d022ef$95707e10$c0517a30$@gfphelps.com> <06a201d02361$e5ae4e30$b10aea90$@gfphelps.com> <004f01d02445$2970b520$7c521f60$@gfphelps.com> <008401d0244f$e6ff9430$b4febc90$@gfphelps.com> Message-ID: <013a01d02541$f410d470$dc327d50$@gfphelps.com> Brian West, I wiped out my configuration and started over. I am able to dial from one extension to another ? with 2-way audio. However, I am still not able to dial an external phone number. I also cannot dial the example extensions, x9196, x5000, etc. I am able (via IPv6) to register with my VoIP gateway: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::switch2voip.us gateway sip:9221591504 at 66.33.147.150 REGED external profile sip:mod_sofia at 54.174.255.168:5080 RUNNING (0) external::switch2voip.us gateway sip:9221591504 at 66.33.147.150 FAIL_WAIT 172.31.33.109 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia at 54.174.255.168:5060 RUNNING (0) ================================================================================================= 4 profiles 1 alias A failed dial attempt is shown below. Freeswitch is apparently still attempting to route the call to a non-existent local extension (sip:14049392032 at 172.31.33.109) instead of my VoIP provider. recv 880 bytes from udp/[50.160.141.159]:49334 at 15:57:15.095108: ------------------------------------------------------------------------ INVITE sip:14049392032 at 172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 50.160.141.159:49334;rport;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo Max-Forwards: 70 From: "George F Phelps" ;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: sip:14049392032 at 172.31.33.109 Contact: "George F Phelps" Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria Android 3.2.1 Content-Type: application/sdp Content-Length: 247 v=0 o=- 3629048232 3629048232 IN IP4 50.160.141.159 s=cpc_med c=IN IP4 50.160.141.159 t=0 0 m=audio 4000 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 406 bytes to udp/[50.160.141.159]:49334 at 15:57:15.095386: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo From: "George F Phelps" ;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: sip:14049392032 at 172.31.33.109 Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Content-Length: 0 ------------------------------------------------------------------------ send 802 bytes to udp/[50.160.141.159]:49334 at 15:57:15.097601: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo Max-Forwards: 70 From: "George F Phelps" ;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr To: ;tag=ctKjQv8vKyZtB Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT CSeq: 30128 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "14049392032" ;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ My outbound dialplan ?/usr/local/freeswitch/conf/dialplan/default/00_outbound_calls.xml?: My gateway ?/usr/local/freeswitch/conf/directory/default/example.com.xml?: And in ?/usr/local/freeswitch/conf/vars.xml?: I captured IP packets on my server, and during the dial attempt, there are no IP packets being sent to my VoIP provider (66.33.147.150). I did make the additional configuration updates as documented here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 1:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Wipe out your config and start over from our samples, clearly something was changed that wasn't understood. I could only possibly help if I had access to the system and understood the topology completely. I've only looked over what you've posted and it would seem to me someone has modified the configs and doesn't fully understand how things interact. On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps wrote: Brian West, Okay, I?m sure there is an answer/solution there, but it?s over my head? Questions How do I check to see if I have inadvertently disabled ?auth?? I am 99% sure that I have do changed it. I am 100% sure that I have not touched anything to do with ?allow acl?. But how do I check? So are you saying that dialing an outside in the in the ?public? context is correct? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 10:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth. On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps wrote: Follow on? It appears, to me, that my outbound call is being processed in the ?public? context: 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public Don?t I want it be processed in my ?default? context? My local extensions are in the ?default? context. My dialplan (for my gateway) is in the ?default? context. (Trace segment below.) Thanks, George send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b From: "George F Phelps" >;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U To: sip:4049392032 at 172.31.33.109 Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ CSeq: 10680 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit Content-Length: 0 ------------------------------------------------------------------------ nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680) nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001 at 172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001 at 172.31.33.109 has executed the last dialplan instruction, hanging up. 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001 at 172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Monday, December 29, 2014 7:21 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Chris Tunbridge, I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: ------------------------------------------------------------------------ INVITE sip:17708410143 at 172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport Max-Forwards: 70 Contact: To: "George Phelps" From: "George F Phelps";tag=13860149 Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.5 stamp 71238 Content-Length: 264 v=0 o=- 13064325826971649 1 IN IP4 192.168.1.100 s=Bria 3 release 3.5.5 stamp 71238 c=IN IP4 192.168.1.100 t=0 0 m=audio 50404 RTP/AVP 9 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Sunday, December 28, 2014 11:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea wrote: Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps wrote: The full output from the ?xml_locate dialplan? command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: My suspicion is that some other dialplan, other than my ?switch2voip.us? dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud server. ?4049392032? is a real phone number ? not an extension. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032 at 172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps wrote: New ?pastebin? created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the ?https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. I?m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch ?show codecs? command indicates support for ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the ?xml_locate dialplan? output ? but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a And in ?/usr/local/freeswitch/conf/vars.xml?: I captured IP packets on my server, and during the dial attempt, there are no IP packets being sent to my VoIP provider (66.33.147.150). I did make the additional configuration updates as documented here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 1:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Wipe out your config and start over from our samples, clearly something was changed that wasn't understood. I could only possibly help if I had access to the system and understood the topology completely. I've only looked over what you've posted and it would seem to me someone has modified the configs and doesn't fully understand how things interact. On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps wrote: Brian West, Okay, I?m sure there is an answer/solution there, but it?s over my head? Questions How do I check to see if I have inadvertently disabled ?auth?? I am 99% sure that I have do changed it. I am 100% sure that I have not touched anything to do with ?allow acl?. But how do I check? So are you saying that dialing an outside in the in the ?public? context is correct? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 10:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth. On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps wrote: Follow on? It appears, to me, that my outbound call is being processed in the ?public? context: 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public Don?t I want it be processed in my ?default? context? My local extensions are in the ?default? context. My dialplan (for my gateway) is in the ?default? context. (Trace segment below.) Thanks, George send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b From: "George F Phelps" >;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U To: sip:4049392032 at 172.31.33.109 Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ CSeq: 10680 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit Content-Length: 0 ------------------------------------------------------------------------ nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680) nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001 at 172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001 at 172.31.33.109 has executed the last dialplan instruction, hanging up. 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001 at 172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Monday, December 29, 2014 7:21 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Chris Tunbridge, I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: ------------------------------------------------------------------------ INVITE sip:17708410143 at 172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport Max-Forwards: 70 Contact: To: "George Phelps" From: "George F Phelps";tag=13860149 Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.5 stamp 71238 Content-Length: 264 v=0 o=- 13064325826971649 1 IN IP4 192.168.1.100 s=Bria 3 release 3.5.5 stamp 71238 c=IN IP4 192.168.1.100 t=0 0 m=audio 50404 RTP/AVP 9 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Sunday, December 28, 2014 11:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea wrote: Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps wrote: The full output from the ?xml_locate dialplan? command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: My suspicion is that some other dialplan, other than my ?switch2voip.us? dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud server. ?4049392032? is a real phone number ? not an extension. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032 at 172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps wrote: New ?pastebin? created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the ?https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. I?m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch ?show codecs? command indicates support for ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the ?xml_locate dialplan? output ? but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a And in ?/usr/local/freeswitch/conf/vars.xml?: I captured IP packets on my server, and during the dial attempt, there are no IP packets being sent to my VoIP provider (66.33.147.150). I did make the additional configuration updates as documented here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 1:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Wipe out your config and start over from our samples, clearly something was changed that wasn't understood. I could only possibly help if I had access to the system and understood the topology completely. I've only looked over what you've posted and it would seem to me someone has modified the configs and doesn't fully understand how things interact. On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps wrote: Brian West, Okay, I?m sure there is an answer/solution there, but it?s over my head? Questions How do I check to see if I have inadvertently disabled ?auth?? I am 99% sure that I have do changed it. I am 100% sure that I have not touched anything to do with ?allow acl?. But how do I check? So are you saying that dialing an outside in the in the ?public? context is correct? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 30, 2014 10:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have the context set to public unless you auth to prevent someone from opening up their default context by accident if they happen to turn off auth. On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps wrote: Follow on? It appears, to me, that my outbound call is being processed in the ?public? context: 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public Don?t I want it be processed in my ?default? context? My local extensions are in the ?default? context. My dialplan (for my gateway) is in the ?default? context. (Trace segment below.) Thanks, George send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b From: "George F Phelps" >;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U To: sip:4049392032 at 172.31.33.109 Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ CSeq: 10680 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit Content-Length: 0 ------------------------------------------------------------------------ nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680) nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001 at 172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001 at 172.31.33.109 has executed the last dialplan instruction, hanging up. 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001 at 172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Monday, December 29, 2014 7:21 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable Chris Tunbridge, I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: ------------------------------------------------------------------------ INVITE sip:17708410143 at 172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport Max-Forwards: 70 Contact: To: "George Phelps" From: "George F Phelps";tag=13860149 Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: Bria 3 release 3.5.5 stamp 71238 Content-Length: 264 v=0 o=- 13064325826971649 1 IN IP4 192.168.1.100 s=Bria 3 release 3.5.5 stamp 71238 c=IN IP4 192.168.1.100 t=0 0 m=audio 50404 RTP/AVP 9 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Sunday, December 28, 2014 11:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the /usr/local/freeswitch/conf/dialplan/default/ folder? The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea wrote: Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps wrote: The full output from the ?xml_locate dialplan? command is already in the previously pasted logfile. Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us: My suspicion is that some other dialplan, other than my ?switch2voip.us? dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address ?172.31.33.109? is the local/internal IP address for my AWS virtual cloud server. ?4049392032? is a real phone number ? not an extension. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 5:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable can you share your dialplan? It looks like you're dialing "To: sip:4049392032 at 172.31.33.109" but have no extension for that... On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps wrote: New ?pastebin? created: http://pastebin.com/UwmgJGGg George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Sunday, December 28, 2014 4:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please login to view it." On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps wrote: Chris Tunbridge, 1) I made the updates to my configuration, as suggested in the ?https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. I?m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio. 2) Extension x9161 is one of the default dialplan applications. 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Saturday, December 27, 2014 2:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range. 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here. 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps wrote: Chris Tunbridge, et al., 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch ?show codecs? command indicates support for ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the ?xml_locate dialplan? output ? but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (switch2voip.us) gateway. 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge Sent: Thursday, December 25, 2014 9:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) 3) Do you have an outbound route configured that matches your dial string? 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a > > > > > > > > > > > > > > > > > > And in ?/usr/local/freeswitch/conf/vars.xml?: > > > > > > > > > > > > > > > > I captured IP packets on my server, and during the dial attempt, there are no IP packets being sent to my VoIP provider > (66.33.147.150). > > I did make the additional configuration updates as documented here: > > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? > > Thanks, > > George > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Tuesday, December 30, 2014 1:24 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > Wipe out your config and start over from our samples, clearly something was changed that wasn't understood. I could > only possibly help if I had access to the system and understood the topology completely. I've only looked over what > you've posted and it would seem to me someone has modified the configs and doesn't fully understand how things interact. > > On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps > wrote: > > Brian West, > > Okay, I?m sure there is an answer/solution there, but it?s over my head? > > *Questions* > > How do I check to see if I have inadvertently disabled ?auth?? I am 99% sure that I have do changed it. > > I am 100% sure that I have not touched anything to do with ?allow acl?. But how do I check? > > So are you saying that dialing an outside in the in the ?public? context is correct? > > Thanks, > > George > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org ] *On Behalf > Of *Brian West > *Sent:* Tuesday, December 30, 2014 10:32 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > Smells like someone has either disabled auth, or setup an allow acl, because the internal profile in the defaults have > the context set to public unless you auth to prevent someone from opening up their default context by accident if they > happen to turn off auth. > > On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps > wrote: > > Follow on? > > It appears, to me, that my outbound call is being processed in the ?public? context: > > 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 > in context public > > Don?t I want it be processed in my ?default? context? My local extensions are in the ?default? context. My dialplan > (for my gateway) is in the ?default? context. > > (Trace segment below.) > > Thanks, > > George > > send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b > > From: "George F Phelps" >;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U > > To: sip:4049392032 @172.31.33.109 > > Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ > > CSeq: 10680 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit > > Content-Length: 0 > > ------------------------------------------------------------------------ > > nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680) > > nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying > > nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer > > soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called > > nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying > > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > > 2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001 at 172.31.33.109 > [3a7e8146-7167-4276-a90c-70955ed5c250] > > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering > > nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil) > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > 2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 > in context public > > 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001 at 172.31.33.109 > has executed the last dialplan instruction, hanging up. > > 2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001 at 172.31.33.109 > [CS_EXECUTE] [NORMAL_CLEARING] > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org ] *On Behalf > Of *George F. Phelps > *Sent:* Monday, December 29, 2014 7:21 AM > > > *To:* 'FreeSWITCH Users Help' > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > Chris Tunbridge, > > I move my dialplan to the other folder. I am still not able to place a call. Is it still trying to dial a local extension? > > recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167: > > ------------------------------------------------------------------------ > > INVITE sip:17708410143 at 172.31.33.109 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport > > Max-Forwards: 70 > > Contact: > > To: "George Phelps" > > From: "George F Phelps";tag=13860149 > > Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk > > CSeq: 1 INVITE > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO > > Content-Type: application/sdp > > Supported: replaces > > User-Agent: Bria 3 release 3.5.5 stamp 71238 > > Content-Length: 264 > > v=0 > > o=- 13064325826971649 1 IN IP4 192.168.1.100 > > s=Bria 3 release 3.5.5 stamp 71238 > > c=IN IP4 192.168.1.100 > > t=0 0 > > m=audio 50404 RTP/AVP 9 8 0 18 101 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=yes > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=sendrecv > > ------------------------------------------------------------------------ > > Thanks, > > George > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Tunbridge > *Sent:* Sunday, December 28, 2014 11:07 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the > /usr/local/freeswitch/conf/dialplan/default/ folder? > > The main folder (not /default/) is used for context's, so it wouldn't get included in your default context. > > On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea > wrote: > > Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like: > > > > > > > > > > > > On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps > wrote: > > The full output from the ?xml_locate dialplan? command is already in the previously pasted logfile. > > Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us : > > > > > > > > > > > > > > > > > > > > My suspicion is that some other dialplan, other than my ?switch2voip.us ? dialplan, is being > invoked. My SIP Proxy is at 66.33.147.150. IP address ?172.31.33.109? is the local/internal IP address for my AWS > virtual cloud server. ?4049392032? is a real phone number ? not an extension. > > Thanks, > > George > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org ] *On Behalf > Of *David Villasmil Govea > *Sent:* Sunday, December 28, 2014 5:05 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > can you share your dialplan? It looks like you're dialing > > "To: sip:4049392032 @172.31.33.109 " > > but have no extension for that... > > On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps > wrote: > > New ?pastebin? created: > > http://pastebin.com/UwmgJGGg > > George > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org ] *On Behalf > Of *David Villasmil Govea > *Sent:* Sunday, December 28, 2014 4:04 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, > please login to view it." > > On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps > wrote: > > Chris Tunbridge, > > 1) I made the updates to my configuration, as suggested in the > ?https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2? link. I?m still not able to make a call to an outside > number. A call to an extension connects, but there is still no audio. > > 2) Extension x9161 is one of the default dialplan applications. > > 3) Call failure log posted at: http://pastebin.com/E4sqTLa4 > > Thanks, > > George > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org ] *On Behalf > Of *Chris Tunbridge > *Sent:* Saturday, December 27, 2014 2:30 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > 1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The important part is the external sip ip and external > rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as > long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml > > 2) Your issue you said was with extension x9196, is this another sip endpoint or a dialplan application? If this is a > sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the > 10XX range. > > 3) Can you post a log here http://pastebin.freeswitch.org of a call attempt? My guess is that something's not matching > the request, a complete log of a call attempt would help most here. > > 4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts. > > On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps > wrote: > > Chris Tunbridge, et al., > > 1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria > softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The > Freeswitch ?show codecs? command indicates support for ?codec, G.711 ulaw, CORE_PCM_MODULE? ? which is the codec that I > am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms. > > 2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords. > > 3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It > shows up in the ?xml_locate dialplan? output ? but as the very last entry. My guess is that Freeswitch is attempting to > us some other (default, example?) gateway instead of my desired (switch2voip.us ) gateway. > > > > > > > > > > > > > > > > > > > > 4) The ?mod_v8? issue is now resolved. The module was not being built. I?m not sure why the downloaded default > build/install files were not building it, but were attempting to load it. Sounds like a bug to me? > > Thanks, > > George > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org ] *On Behalf > Of *Chris Tunbridge > *Sent:* Thursday, December 25, 2014 9:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable > > 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration > > 2) If you're using default configs, its configured to look for extensions 10XX, you can see this in > conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside) > > 3) Do you have an outbound route configured that matches your dial string? > > 4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml > find the line that says mod_v8 and put a And for new Freeswitch users, it would probably be good to add a comment that SIP gateways defined in ?/usr/local/freeswitch/conf/sip_profiles/external/? are implicitly IPv4 gateways. Sure, it all makes sense now? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, January 02, 2015 5:21 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Control of IPv6 vs. IPv4 I think he's in the unique position to have a provider that does ipv6, I've never seen one yet, its like the unicorn of voip. :P On Fri, Jan 2, 2015 at 4:03 PM, Steven Ayre wrote: Do you have the same gateway configured on both the external and external-ipv6 profiles? It looks like way, which would mean you actually have two user agents registering with the same details at the same time - one over ipv4 and one over ipv6. Check your external-ipv6 is not including the gateways in the external subdirectory. On 2 January 2015 at 18:12, George F. Phelps wrote: Brian West, With my mostly default, current configuration, I am seeing Freeswitch send out simultaneous IPv4 and IPv6 registration attempts ? not just one or the other. I am only configuring the (IPv4) IP address of the SIP proxy. I assume Freeswitch is defaulting to use port 5060. And empirically, it?s completely random as to which type of registration (IPv4 vs. IPv6) succeeds. And if, for example, IPv6 registration succeeds, then the registration attempts for IPv4 continue retrying in the background. See my ?sofia status? output below ? IPv6=REGED and IPv4=TRYING (retry: 20s). My specific question is what Freeswitch configuration should I change to only have one type of registration? How do I: ??you should pick where you want your gateway to register at, and only allow it there and there only??? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, January 02, 2015 12:13 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Control of IPv6 vs. IPv4 FreeSWITCH does one or the other, you should pick where you want your gateway to register at, and only allow it there and there only. On Fri, Jan 2, 2015 at 10:02 AM, George F. Phelps wrote: My VoIP provider supports both IPv4 and IPV6 registrations. I can only have one registration active at a time. Freeswitch is attempting both IPv4 and IPv6 connections. Randomly, sometimes a IPv4 connection is the first (only) registration established; other times it is the IPv6 connection. How to I configure Freeswitch to deterministically only attempt one type (my choice of either IPv4 or IPv6) of connection? freeswitch at ip-172-31-33-109.ec2.internal> sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::switch2voip.us gateway sip:XXXXXXXXXX at 66.33.147.150 REGED external profile sip:mod_sofia at 54.174.255.168:5080 RUNNING (0) external::switch2voip.us gateway sip:XXXXXXXXXX at 66.33.147.150 TRYING (retry: 20s) 172.31.33.109 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia at 54.174.255.168:5060 RUNNING (0) ================================================================================================= 4 profiles 1 alias freeswitch at ip-172-31-33-109.ec2.internal> Thanks, George _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/9a6acf02/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 6528 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/9a6acf02/attachment-0001.bin From robert at oldhamtechnology.com Mon Jan 5 19:13:25 2015 From: robert at oldhamtechnology.com (Robert Oldham) Date: Mon, 05 Jan 2015 09:13:25 -0700 Subject: [Freeswitch-users] multi-tenant registration for Cisco spa 112 In-Reply-To: References: <5A1A81ED-172F-4CDA-A4C6-BE9FBD9B7642@oldhamtechnology.com> <54A6CA12.8050304@oldhamtechnology.com> <54A6E406.1040503@oldhamtechnology.com> Message-ID: <54AAB825.4050203@oldhamtechnology.com> Luis, NDLB-force-rport=true solved my problem. I tried other options, but they did not help. Thank you for your help with this. Robert Oldham ------------------------------------------------------------------------ Oldham Technology W: 801-877-2190 x801 E: robert at oldhamtechnology.com http://www.oldhamtechnology.com On 01/02/2015 12:55 PM, Luis Daniel Lucio Quiroz wrote: > If it is a NAT issue, you may want to try NDLB-force-rport=true/safe > > Luis Daniel Lucio Quiroz > CISSP, CISM, CISA > Linux, VoIP and much more fun > www.okay.com.mx > > Need LCR? Check out LCR for FusionPBX with FreeSWITCH > Need Billing? Check out Billing for FusionPBX with FreeSWITCH > > 2015-01-02 13:31 GMT-05:00 Robert Oldham >: > > Thank you Vik. I'll try that too. > > Robert Oldham > ------------------------------------------------------------------------ > Oldham Technology > W: 801-877-2190 x801 > E: robert at oldhamtechnology.com > http://www.oldhamtechnology.com > > > On 01/02/2015 10:01 AM, Vik Killa wrote: >> I have Cisco SPA5XX setup in multi-tenant mode. I used the proxy >> params to accomplish it, that way I did not need a FQDN to >> resolve. Here is an example from my prov files (192.168.0.100 is >> IP of FS and '3.local' is made up domain for a tenant) >> >> 1 >> E. 1000 >> > group="Phone/Line_Key_1">private >> > group="Ext_1/Share_Line_Appearance">private >> 3.local >> > group="Ext_1/Proxy_and_Registration">192.168.0.100 >> > group="Ext_1/Proxy_and_Registration">Yes >> > group="Ext_1/Proxy_and_Registration">Yes >> 1000 >> > group="Ext_1/Subscriber_Information">********** >> > group="Ext_1/SIP_Settings">No >> [x*]. >> 1000 at 3.local >> >> > group="Ext_1/Proxy_and_Registration">300 >> > group="Ext_1/Share_Line_Appearance">300 >> > group="Ext_1/Call_Feature_Settings">300 >> >> >> >> >> >> On Fri, Jan 2, 2015 at 11:40 AM, Robert Oldham >> > > wrote: >> >> Florent, >> >> I am not sure I understand which logs you mean. Logs from the >> Cisco spa 112, FreeSWITCH, the firewall, or all of the above? >> >> Thanks, >> Robert Oldham >> ------------------------------------------------------------------------ >> Oldham Technology >> W: 801-877-2190 x801 >> E: robert at oldhamtechnology.com >> >> http://www.oldhamtechnology.com >> >> >> On 01/02/2015 12:45 AM, Florent Krieg wrote: >>> >>> Hello, >>> >>> What do you see in the logs? >>> That might be relevant to understand what it tries to do. >>> >>> Florent >>> >>> Le 2 janv. 2015 08:15, "Robert Oldham" >>> >> > a ?crit : >>> >>> I am registering to the FQDN of the domain. >>> >>> There is a NAT involved. The PBX is not behind NAT. The >>> phones are behind NAT on separate network. >>> >>> Thank you, >>> >>> Robert Oldham >>> Oldham Technology >>> Phone: (801) 877-2190 >>> Email: robert at oldhamtechnology.com >>>