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 >>> >>> Website: https://www.oldhamtechnology.com >>> >>> >>> On January 1, 2015 7:41:45 PM MST, Moishe Grunstein >>> > wrote: >>> >>> Are you registering to the FQDN of the domain? >>> >>> Is there any NAT involved? PBX behind NAT? phones >>> behind NAT? >>> >>> >>> >>> Thanks, >>> >>> >>> >>> Moishe Grunstein >>> >>> Tornado Computer Systems, Inc. >>> >>> 212.400.7650 888.IPPBX.US >>> >>> *Service Request Email: support at nysolutions.com >>> * >>> >>> Polycom Certified VAR >>> Microsoft Small Business Specialist, Cisco SMB >>> Select Certified >>> >>> cid:image001.jpg at 01C72F94.9EE45D60 >>> >>> >>> Computer Networking * Managed Services * IP Video >>> Surveillance * Network Assessments * Web Solutions * >>> Voice over IP * Disaster Recovery * Network Security >>> * Site Surveys * CMS >>> >>> >>> >>> *From:*freeswitch-users-bounces at lists.freeswitch.org >>> >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org ] >>> *On Behalf Of *Robert Oldham >>> *Sent:* Thursday, January 1, 2015 7:49 PM >>> *To:* freeswitch-users at lists.freeswitch.org >>> >>> *Subject:* [Freeswitch-users] multi-tenant >>> registration for Cisco spa 112 >>> >>> >>> >>> I am a freeswitch noob, so please forgive me if my >>> questions are not coherent with proper concepts. I >>> have made an earnest attempt to find clues to my >>> problem on the wiki and elsewhere, but suspect I am >>> asking the wrong questions. >>> >>> Problem: >>> I cannot get a Cisco spa 112 to register properly. >>> >>> Context: >>> I am trying to put together a proof of concept >>> multi-tenant freeswitch server. After reading the >>> book FreeSWITCH 1.2 by Anthony Minessale, I used the >>> FusionPBX install script to get freeswitch 1.4.14 >>> built on a fresh Ubuntu 12.04 install. I configured >>> a domain name to point to the new server. On the >>> server I added the domain in the directory and 3 >>> users/extensions along with several sip gateways. >>> The gateways work. One of the extensions has a >>> Grandstream GXP2160 registered successfully, another >>> has a Zoiper soft phone registered successfully. >>> >>> Details: >>> I have tried to configure and get a Cisco SPA 112 to >>> register on the 3rd extension. However, it is >>> failing to register. I have watched the sip trace >>> and packet capture. Both show the same thing: the >>> Cisco SPA sends a sip register request to which >>> sofia responds with a 401 "unauthorized". Then, >>> instead of sending a sip register with an >>> authorization digest as the other phones do, the >>> Cisco spa 112, sends another generic sip register >>> request, which elicits another 401 "unauthorized" >>> from sofia. This repeats a few times, and then the >>> spa 112 gives up, waits 40 seconds and then does it >>> all over again. >>> >>> I have had this device registered with an asterisk >>> server without trouble. I have reset the device to >>> factory defaults and updated the firmware, all >>> without any change. >>> >>> The domain name is fairly long. Don't know if that >>> might be a factor. >>> >>> Any clues where I might look next or tests I might >>> try for more information? >>> >>> Thank you, >>> >>> Robert Oldham >>> Oldham Technology >>> Phone: (801) 877-2190 >>> Email: robert at oldhamtechnology.com >>> >>> Website: https://www.oldhamtechnology.com >>> >>> ------------------------------------------------------------------------ >>> >>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> 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 >> >> >> >> >> _________________________________________________________________________ >> 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 > > > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/752a2649/attachment-0001.html From mike at jerris.com Mon Jan 5 21:02:01 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2015 13:02:01 -0500 Subject: [Freeswitch-users] Detecting multiple tones with freeswitch In-Reply-To: <54A7E4A8.6090702@gmail.com> References: <54A7E4A8.6090702@gmail.com> Message-ID: <012A2476-B5CC-4F5A-BDF9-E13AE726319E@jerris.com> You can never reliably depend on the tones, particularly with international traffic. This is never going to work reliably unless you get useful sip signaling. I would be demanding the carrier change settings back. > On Jan 3, 2015, at 7:46 AM, Bunea Lucian wrote: > > Hello, > > My SIP provider (Vodafone) decided to make me a Christmas present: they > change the configuration of their SIP server. > > Using the old configuration they were sending SIP codes for decline, > busy and unavailable. > Now they are sending early media with tones (for decline and busy) and a > one minute message for unavailable. > Each status is followed by a SIP/480. > > Since I need to be able to distinguish between different statuses, I > have determined the following: > - if I don't receive a ring tone within 7 second, the called number is > unavailable; > - if I receive a busy tone within 4 seconds the called number is busy; > - if I receive a busy tone after 4 second the call number has declined > the call; > > This is how I tried to implement it: > - after receiving early media (SIP/180 or SIP/183) a timer is started > for call hangup (sched_hangup) > - if I receive a ring tone within 7 seconds the timer is canceled > (sched_cancel); > - if I receive a busy tone a lua script is called; > > > > > > > > > data="sofia/external/${destination_number}@XXX.XXX.XXX.XXX"/> > > But, for some reason, the tones are detected, but the applications are > never called. > > > If I add one of the following lines to the dialplan the corresponding > application is called: > > > > or > > > > but I can't tell which tone was detected... > > What am I missing? > > PS: Tested with FreeSWITCH Version 1.4.14-1~64bit. > > Regards, > Lucian From mike at jerris.com Mon Jan 5 21:06:56 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2015 13:06:56 -0500 Subject: [Freeswitch-users] MP2 In-Reply-To: <54A719B6.2050609@virtues.net> References: <54A71282.9060307@virtues.net> <54A71601.1070403@virtues.net> <54A719B6.2050609@virtues.net> Message-ID: <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> When g722 is used over isdn, it is a voice channel marked as g722, NOT a data channel with voice codec data. > On Jan 2, 2015, at 5:20 PM, Thomas Auge wrote: > > There are ISDN devices which establish a digital (data) ISDN connection and then use all kinds of codecs on top of that. > Think FreeTDM with G722 - same thing. As far as I know FreeTDM doesn't care which codec you choose for a digital > connection. G722 is just the most common use case. > > MP2 is still a widely used standard in broadcast, i.e. for remote contributions. > > On 02.01.2015 19:08, Brian West wrote: >> I'm still puzzled as to why mp2, since ISDN is native PCM (ulaw/alaw), Do you have some legacy files or systems you're >> working with that require this? Trying to fully understand your thought process and needs. >> >> On Fri, Jan 2, 2015 at 4:04 PM, Thomas Auge >> wrote: >> >> Digital ISDN calls, so I think as far as freeswitch is concerned, voice calls, yes. >> >> >> On 02.01.2015 18:52, Brian West wrote: >>> In what context? playing sound files, voice calls what? >>> >>> On Fri, Jan 2, 2015 at 3:49 PM, Thomas Auge > >>> wrote: >>> >>> Hello list, >>> >>> would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for it? >>> >>> Cheers! >>> >>> Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/e8d4d493/attachment.html From mike at jerris.com Mon Jan 5 21:23:20 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2015 13:23:20 -0500 Subject: [Freeswitch-users] MP2 In-Reply-To: <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> References: <54A71282.9060307@virtues.net> <54A71601.1070403@virtues.net> <54A719B6.2050609@virtues.net> <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> Message-ID: <18E4692B-3BD9-4E61-AEEA-21F2A4CCA79A@jerris.com> To understand this a little better. Is MP2 over data channel a standard that needs to be interoperated with? Is there details how this is signalled to the other side or are both sides just configured to support it? > On Jan 5, 2015, at 1:06 PM, Michael Jerris wrote: > > When g722 is used over isdn, it is a voice channel marked as g722, NOT a data channel with voice codec data. > >> On Jan 2, 2015, at 5:20 PM, Thomas Auge > wrote: >> >> There are ISDN devices which establish a digital (data) ISDN connection and then use all kinds of codecs on top of that. >> Think FreeTDM with G722 - same thing. As far as I know FreeTDM doesn't care which codec you choose for a digital >> connection. G722 is just the most common use case. >> >> MP2 is still a widely used standard in broadcast, i.e. for remote contributions. >> >> On 02.01.2015 19:08, Brian West wrote: >>> I'm still puzzled as to why mp2, since ISDN is native PCM (ulaw/alaw), Do you have some legacy files or systems you're >>> working with that require this? Trying to fully understand your thought process and needs. >>> >>> On Fri, Jan 2, 2015 at 4:04 PM, Thomas Auge >> wrote: >>> >>> Digital ISDN calls, so I think as far as freeswitch is concerned, voice calls, yes. >>> >>> >>> On 02.01.2015 18:52, Brian West wrote: >>>> In what context? playing sound files, voice calls what? >>>> >>>> On Fri, Jan 2, 2015 at 3:49 PM, Thomas Auge > >>> wrote: >>>> >>>> Hello list, >>>> >>>> would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for it? >>>> >>>> Cheers! >>>> >>>> Thomas > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/a234475f/attachment-0001.html From mike at jerris.com Mon Jan 5 21:25:12 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2015 13:25:12 -0500 Subject: [Freeswitch-users] lua custom variables In-Reply-To: References: Message-ID: sounds like the second time its executing on a different session > On Jan 4, 2015, at 9:01 PM, David Villasmil Govea wrote: > > Hello guys, > > Any help with this? > > Thanks > > David > > On Sun, Jan 4, 2015 at 7:12 AM, David Villasmil Govea > wrote: > Update: > > I execute 2 scripts. > > I set an execute_on_answer lua script (lua2), and execute another (lua1). > like so: > > > > > > > > > > > > > > Now, the lua1 executes, i set a variable "custom_dur" which is the duration for the call. > When lua3 executes on answer, the variable custom_duration is still there, no problems. Then I transfer the call to "9999", which is: > > > > > > > > > Now, on lua3 I try to get the variable and it's not there anymore! > > I also tried the following: On lua1 I create a new variable, which is seen on ALL scripts, including lua3, but I reset it to a new value on lua2, and on lua3 I don't see the change!! The variable is still the value I set on lua1!!! wth?? > > Am I missing something on variables?? > On Sun, Jan 4, 2015 at 3:30 AM, David Villasmil Govea > wrote: > Hello Guys, > > I have this lus script, which sets a custom variable, this is ok. > The the call is transferred to another extension, and the variable is not there anymore, is this by design? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/294ca86a/attachment.html From auge at virtues.net Mon Jan 5 21:27:03 2015 From: auge at virtues.net (Thomas Auge) Date: Mon, 05 Jan 2015 15:27:03 -0300 Subject: [Freeswitch-users] MP2 In-Reply-To: <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> References: <54A71282.9060307@virtues.net> <54A71601.1070403@virtues.net> <54A719B6.2050609@virtues.net> <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> Message-ID: <54AAD777.5010102@virtues.net> Then "unrestricted digital" is still a voice channel? I'm completely clueless as to how this works on the ISDN side of things, I'm just assuming that any codec would work, because else the "unrestricted-digital-codec" option wouldn't make sense. :) I assumed it was a data connection, because some of the hardware boxes offer 2-channel bonding on BRIs (an option, we're also exploring with FreeTDM) and the only way to do this I know of is through a data connection. On 05.01.2015 15:06, Michael Jerris wrote: > When g722 is used over isdn, it is a voice channel marked as g722, NOT a data channel with voice codec data. > >> On Jan 2, 2015, at 5:20 PM, Thomas Auge > wrote: >> >> There are ISDN devices which establish a digital (data) ISDN connection and then use all kinds of codecs on top of that. >> Think FreeTDM with G722 - same thing. As far as I know FreeTDM doesn't care which codec you choose for a digital >> connection. G722 is just the most common use case. >> >> MP2 is still a widely used standard in broadcast, i.e. for remote contributions. >> >> On 02.01.2015 19:08, Brian West wrote: >>> I'm still puzzled as to why mp2, since ISDN is native PCM (ulaw/alaw), Do you have some legacy files or systems you're >>> working with that require this? Trying to fully understand your thought process and needs. >>> >>> On Fri, Jan 2, 2015 at 4:04 PM, Thomas Auge > wrote: >>> >>> Digital ISDN calls, so I think as far as freeswitch is concerned, voice calls, yes. >>> >>> >>> On 02.01.2015 18:52, Brian West wrote: >>>> In what context? playing sound files, voice calls what? >>>> >>>> On Fri, Jan 2, 2015 at 3:49 PM, Thomas Auge >>>> >> wrote: >>>> >>>> Hello list, >>>> >>>> would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for it? >>>> >>>> Cheers! >>>> >>>> Thomas > > > > _________________________________________________________________________ > 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 > From mike at jerris.com Mon Jan 5 21:33:50 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2015 13:33:50 -0500 Subject: [Freeswitch-users] Multiple P-Asserted-Identity Headers In-Reply-To: <076a01d028de$96357180$c2a05480$@gfphelps.com> References: <076a01d028de$96357180$c2a05480$@gfphelps.com> Message-ID: <7CA4DE25-0C25-4AFA-B4C2-6D887A3A328F@jerris.com> put them together as a single header separated by a comma. http://www.ietf.org/mail-archive/web/sip/current/msg17485.html has more info. > On Jan 5, 2015, at 6:56 AM, George F. Phelps wrote: > > How do I create two (2) unique, P-Asserted-Identity headers? As required by RFC 3325 : > > A P-Asserted-Identity header field value MUST consist of exactly one name-addr or addr-spec. There may be one or two P-Asserted-Identity values. If there is one value, it MUST be a sip, sips, or tel URI. If there are two values, one value MUST be a sip or sips URI and the other MUST be a tel URI. > > A valid, multiple P-Asserted-Identity headers example, taken from RFC 3325: > > INVITE sip:+14085551212 at proxy.pstn.net SIP/2.0 > Via: SIP/2.0/TCP useragent.cisco.com ;branch=z9hG4bK-124 > Via: SIP/2.0/TCP proxy.cisco.com ;branch=z9hG4bK-abc > To: > > From: "Anonymous" >;tag=9802748 > Call-ID: 245780247857024504 > CSeq: 2 INVITE > Max-Forwards: 69 > P-Asserted-Identity: "Cullen Jennings" > > P-Asserted-Identity: tel:+14085264000 > Privacy: id > > When I use the following two (2) statements in a Freeswitch dialplan, the second ?set,? of course, overwrites the data stored by the first ?set.? > > > > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/1d40caca/attachment-0001.html From auge at virtues.net Mon Jan 5 21:34:17 2015 From: auge at virtues.net (Thomas Auge) Date: Mon, 05 Jan 2015 15:34:17 -0300 Subject: [Freeswitch-users] MP2 In-Reply-To: <18E4692B-3BD9-4E61-AEEA-21F2A4CCA79A@jerris.com> References: <54A71282.9060307@virtues.net> <54A71601.1070403@virtues.net> <54A719B6.2050609@virtues.net> <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> <18E4692B-3BD9-4E61-AEEA-21F2A4CCA79A@jerris.com> Message-ID: <54AAD929.5070809@virtues.net> > To understand this a little better. Is MP2 over data channel a standard that needs to be interoperated with? Is there > details how this is signalled to the other side or are both sides just configured to support it? Unfortunately I don't know. I didn't know there were multiple approaches to this. The currently implemented G722 method works with these boxes, so the assumption was it's all the same with other codecs. If a data channel is not a common method to achieve this, then it's most likely not what they are doing. >> On Jan 5, 2015, at 1:06 PM, Michael Jerris > wrote: >> >> When g722 is used over isdn, it is a voice channel marked as g722, NOT a data channel with voice codec data. >> >>> On Jan 2, 2015, at 5:20 PM, Thomas Auge > wrote: >>> >>> There are ISDN devices which establish a digital (data) ISDN connection and then use all kinds of codecs on top of that. >>> Think FreeTDM with G722 - same thing. As far as I know FreeTDM doesn't care which codec you choose for a digital >>> connection. G722 is just the most common use case. >>> >>> MP2 is still a widely used standard in broadcast, i.e. for remote contributions. >>> >>> On 02.01.2015 19:08, Brian West wrote: >>>> I'm still puzzled as to why mp2, since ISDN is native PCM (ulaw/alaw), Do you have some legacy files or systems you're >>>> working with that require this? Trying to fully understand your thought process and needs. >>>> >>>> On Fri, Jan 2, 2015 at 4:04 PM, Thomas Auge > wrote: >>>> >>>> Digital ISDN calls, so I think as far as freeswitch is concerned, voice calls, yes. >>>> >>>> >>>> On 02.01.2015 18:52, Brian West wrote: >>>>> In what context? playing sound files, voice calls what? >>>>> >>>>> On Fri, Jan 2, 2015 at 3:49 PM, Thomas Auge >>>>> >> wrote: >>>>> >>>>> Hello list, >>>>> >>>>> would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for it? >>>>> >>>>> Cheers! >>>>> >>>>> Thomas >> > > > > _________________________________________________________________________ > 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 > From mike at jerris.com Mon Jan 5 21:38:53 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2015 13:38:53 -0500 Subject: [Freeswitch-users] MP2 In-Reply-To: <54AAD929.5070809@virtues.net> References: <54A71282.9060307@virtues.net> <54A71601.1070403@virtues.net> <54A719B6.2050609@virtues.net> <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> <18E4692B-3BD9-4E61-AEEA-21F2A4CCA79A@jerris.com> <54AAD929.5070809@virtues.net> Message-ID: For any hope to achieve this we are going to needs some more info... preferably in the way of specifications. It's maybe possible if you can capture the isdn traffic from working equipment using this that you could decode it (Wireshark does a decent job of this) and see what exactly its signaling. > On Jan 5, 2015, at 1:34 PM, Thomas Auge wrote: > >> To understand this a little better. Is MP2 over data channel a standard that needs to be interoperated with? Is there >> details how this is signalled to the other side or are both sides just configured to support it? > > Unfortunately I don't know. I didn't know there were multiple approaches to this. The currently implemented G722 method > works with these boxes, so the assumption was it's all the same with other codecs. If a data channel is not a common > method to achieve this, then it's most likely not what they are doing. > > >>> On Jan 5, 2015, at 1:06 PM, Michael Jerris > wrote: >>> >>> When g722 is used over isdn, it is a voice channel marked as g722, NOT a data channel with voice codec data. >>> >>>> On Jan 2, 2015, at 5:20 PM, Thomas Auge > wrote: >>>> >>>> There are ISDN devices which establish a digital (data) ISDN connection and then use all kinds of codecs on top of that. >>>> Think FreeTDM with G722 - same thing. As far as I know FreeTDM doesn't care which codec you choose for a digital >>>> connection. G722 is just the most common use case. >>>> >>>> MP2 is still a widely used standard in broadcast, i.e. for remote contributions. >>>> >>>> On 02.01.2015 19:08, Brian West wrote: >>>>> I'm still puzzled as to why mp2, since ISDN is native PCM (ulaw/alaw), Do you have some legacy files or systems you're >>>>> working with that require this? Trying to fully understand your thought process and needs. >>>>> >>>>> On Fri, Jan 2, 2015 at 4:04 PM, Thomas Auge > wrote: >>>>> >>>>> Digital ISDN calls, so I think as far as freeswitch is concerned, voice calls, yes. >>>>> >>>>> >>>>> On 02.01.2015 18:52, Brian West wrote: >>>>>> In what context? playing sound files, voice calls what? >>>>>> >>>>>> On Fri, Jan 2, 2015 at 3:49 PM, Thomas Auge >>>>>> >> wrote: >>>>>> >>>>>> Hello list, >>>>>> >>>>>> would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for it? >>>>>> >>>>>> Cheers! >>>>>> >>>>>> Thomas >>> >> From auge at virtues.net Mon Jan 5 21:49:19 2015 From: auge at virtues.net (Thomas Auge) Date: Mon, 05 Jan 2015 15:49:19 -0300 Subject: [Freeswitch-users] MP2 In-Reply-To: References: <54A71282.9060307@virtues.net> <54A71601.1070403@virtues.net> <54A719B6.2050609@virtues.net> <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> <18E4692B-3BD9-4E61-AEEA-21F2A4CCA79A@jerris.com> <54AAD929.5070809@virtues.net> Message-ID: <54AADCAF.4050909@virtues.net> These are hardware boxes. Can I capture the data with another ISDN card on the same BRI? On 05.01.2015 15:38, Michael Jerris wrote: > For any hope to achieve this we are going to needs some more info... preferably in the way of specifications. It's > maybe possible if you can capture the isdn traffic from working equipment using this that you could decode it > (Wireshark does a decent job of this) and see what exactly its signaling. > > > > >> On Jan 5, 2015, at 1:34 PM, Thomas Auge wrote: >> >>> To understand this a little better. Is MP2 over data channel a standard that needs to be interoperated with? Is >>> there details how this is signalled to the other side or are both sides just configured to support it? >> >> Unfortunately I don't know. I didn't know there were multiple approaches to this. The currently implemented G722 >> method works with these boxes, so the assumption was it's all the same with other codecs. If a data channel is not >> a common method to achieve this, then it's most likely not what they are doing. >> >> >>>> On Jan 5, 2015, at 1:06 PM, Michael Jerris > wrote: >>>> >>>> When g722 is used over isdn, it is a voice channel marked as g722, NOT a data channel with voice codec data. >>>> >>>>> On Jan 2, 2015, at 5:20 PM, Thomas Auge > wrote: >>>>> >>>>> There are ISDN devices which establish a digital (data) ISDN connection and then use all kinds of codecs on >>>>> top of that. Think FreeTDM with G722 - same thing. As far as I know FreeTDM doesn't care which codec you >>>>> choose for a digital connection. G722 is just the most common use case. >>>>> >>>>> MP2 is still a widely used standard in broadcast, i.e. for remote contributions. >>>>> >>>>> On 02.01.2015 19:08, Brian West wrote: >>>>>> I'm still puzzled as to why mp2, since ISDN is native PCM (ulaw/alaw), Do you have some legacy files or >>>>>> systems you're working with that require this? Trying to fully understand your thought process and needs. >>>>>> >>>>>> On Fri, Jan 2, 2015 at 4:04 PM, Thomas Auge >>>>> > wrote: >>>>>> >>>>>> Digital ISDN calls, so I think as far as freeswitch is concerned, voice calls, yes. >>>>>> >>>>>> >>>>>> On 02.01.2015 18:52, Brian West wrote: >>>>>>> In what context? playing sound files, voice calls what? >>>>>>> >>>>>>> On Fri, Jan 2, 2015 at 3:49 PM, Thomas Auge >>>>>> >> >>>>>>> wrote: >>>>>>> >>>>>>> Hello list, >>>>>>> >>>>>>> would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for it? >>>>>>> >>>>>>> Cheers! >>>>>>> >>>>>>> Thomas >>>> >>> > > > _________________________________________________________________________ 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 > From lucibunea at gmail.com Mon Jan 5 23:32:25 2015 From: lucibunea at gmail.com (Bunea Lucian) Date: Mon, 05 Jan 2015 22:32:25 +0200 Subject: [Freeswitch-users] Detecting multiple tones with freeswitch In-Reply-To: <012A2476-B5CC-4F5A-BDF9-E13AE726319E@jerris.com> References: <54A7E4A8.6090702@gmail.com> <012A2476-B5CC-4F5A-BDF9-E13AE726319E@jerris.com> Message-ID: <54AAF4D9.2030904@gmail.com> I have a dedicated short number and multiple DIDs. I also have a database with customer phone numbers and their corresponding DIDs. Each field agent has a mobile phone (from the same provider). When a call comes in freeswitch automatically routes the call to the designated field agent via the corresponding DID. I want to monitor what happens with the call after that, to be able to decide what to do with the call next. So basically, the call never leaves the provider's network. Btw, I did asked the provider to switch back the settings but the feedback was: "The behavior is normal because for calls made inside Vodafone network, we generate tones. Thus, for Busy, we play specific tone and then disconnect the call, signaling with SIP 480 (corresponding to the end of announcement). If a call comes from another network, User Busy is signaled on ISUP and SIP returned as such. The settings can not be changed per customer. " Unofficially, they acknowledged that their PBX behaved differently before because it was not correctly configured. PS: My provider sends the description of the tones using a reason header inside the SIP/183 message. But I was unable to find a way to extract it: http://lists.freeswitch.org/pipermail/freeswitch-users/2014-March/103645.html -------- Original Message -------- *Subject: *Re: [Freeswitch-users] Detecting multiple tones with freeswitch *From: *Michael Jerris *To: *FreeSWITCH Users Help *Date: *05.01.2015 20:02 > You can never reliably depend on the tones, particularly with international traffic. This is never going to work reliably unless you get useful sip signaling. I would be demanding the carrier change settings back. > > >> On Jan 3, 2015, at 7:46 AM, Bunea Lucian wrote: >> >> Hello, >> >> My SIP provider (Vodafone) decided to make me a Christmas present: they >> change the configuration of their SIP server. >> >> Using the old configuration they were sending SIP codes for decline, >> busy and unavailable. >> Now they are sending early media with tones (for decline and busy) and a >> one minute message for unavailable. >> Each status is followed by a SIP/480. >> >> Since I need to be able to distinguish between different statuses, I >> have determined the following: >> - if I don't receive a ring tone within 7 second, the called number is >> unavailable; >> - if I receive a busy tone within 4 seconds the called number is busy; >> - if I receive a busy tone after 4 second the call number has declined >> the call; >> >> This is how I tried to implement it: >> - after receiving early media (SIP/180 or SIP/183) a timer is started >> for call hangup (sched_hangup) >> - if I receive a ring tone within 7 seconds the timer is canceled >> (sched_cancel); >> - if I receive a busy tone a lua script is called; >> >> >> >> >> >> >> >> >> > data="sofia/external/${destination_number}@XXX.XXX.XXX.XXX"/> >> >> But, for some reason, the tones are detected, but the applications are >> never called. >> >> >> If I add one of the following lines to the dialplan the corresponding >> application is called: >> >> >> >> or >> >> >> >> but I can't tell which tone was detected... >> >> What am I missing? >> >> PS: Tested with FreeSWITCH Version 1.4.14-1~64bit. >> >> Regards, >> Lucian > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/0d5fd03f/attachment.html From auge at virtues.net Mon Jan 5 23:51:59 2015 From: auge at virtues.net (Thomas Auge) Date: Mon, 05 Jan 2015 17:51:59 -0300 Subject: [Freeswitch-users] MP2 In-Reply-To: References: <54A71282.9060307@virtues.net> <54A71601.1070403@virtues.net> <54A719B6.2050609@virtues.net> <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> <18E4692B-3BD9-4E61-AEEA-21F2A4CCA79A@jerris.com> <54AAD929.5070809@virtues.net> Message-ID: <54AAF96F.9020803@virtues.net> After a bit of reading I cannot find anything codec specific about the unrestricted digital bearer capability type FreeTDM uses. There does seem to be a G722 type (7khz audio), but from their documentation that's not what FreeTDM does, and since G722 worked with such a hardware codec, it doesn't seem to be what they do either. In an email Sangoma support once said: ... "we only support G.722 as an HD codec because it fits in the 64kbps stream (in fact any codec that fits 64kbps can be used with proper configuration)" ... That sounds like only MP2 support in Freeswitch would be needed. Or am I missing something? On 05.01.2015 15:38, Michael Jerris wrote: > For any hope to achieve this we are going to needs some more info... preferably in the way of specifications. It's maybe possible if you can capture the isdn traffic from working equipment using this that you could decode it (Wireshark does a decent job of this) and see what exactly its signaling. > > > > >> On Jan 5, 2015, at 1:34 PM, Thomas Auge wrote: >> >>> To understand this a little better. Is MP2 over data channel a standard that needs to be interoperated with? Is there >>> details how this is signalled to the other side or are both sides just configured to support it? >> >> Unfortunately I don't know. I didn't know there were multiple approaches to this. The currently implemented G722 method >> works with these boxes, so the assumption was it's all the same with other codecs. If a data channel is not a common >> method to achieve this, then it's most likely not what they are doing. >> >> >>>> On Jan 5, 2015, at 1:06 PM, Michael Jerris > wrote: >>>> >>>> When g722 is used over isdn, it is a voice channel marked as g722, NOT a data channel with voice codec data. >>>> >>>>> On Jan 2, 2015, at 5:20 PM, Thomas Auge > wrote: >>>>> >>>>> There are ISDN devices which establish a digital (data) ISDN connection and then use all kinds of codecs on top of that. >>>>> Think FreeTDM with G722 - same thing. As far as I know FreeTDM doesn't care which codec you choose for a digital >>>>> connection. G722 is just the most common use case. >>>>> >>>>> MP2 is still a widely used standard in broadcast, i.e. for remote contributions. >>>>> >>>>> On 02.01.2015 19:08, Brian West wrote: >>>>>> I'm still puzzled as to why mp2, since ISDN is native PCM (ulaw/alaw), Do you have some legacy files or systems you're >>>>>> working with that require this? Trying to fully understand your thought process and needs. >>>>>> >>>>>> On Fri, Jan 2, 2015 at 4:04 PM, Thomas Auge > wrote: >>>>>> >>>>>> Digital ISDN calls, so I think as far as freeswitch is concerned, voice calls, yes. >>>>>> >>>>>> >>>>>> On 02.01.2015 18:52, Brian West wrote: >>>>>>> In what context? playing sound files, voice calls what? >>>>>>> >>>>>>> On Fri, Jan 2, 2015 at 3:49 PM, Thomas Auge >>>>>>> >> wrote: >>>>>>> >>>>>>> Hello list, >>>>>>> >>>>>>> would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for it? >>>>>>> >>>>>>> Cheers! >>>>>>> >>>>>>> Thomas >>>> >>> > > > _________________________________________________________________________ > 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 > From mike at jerris.com Mon Jan 5 23:52:14 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2015 15:52:14 -0500 Subject: [Freeswitch-users] MP2 In-Reply-To: <54AADCAF.4050909@virtues.net> References: <54A71282.9060307@virtues.net> <54A71601.1070403@virtues.net> <54A719B6.2050609@virtues.net> <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> <18E4692B-3BD9-4E61-AEEA-21F2A4CCA79A@jerris.com> <54AAD929.5070809@virtues.net> <54AADCAF.4050909@virtues.net> Message-ID: <9235C974-F010-45CD-A383-81D640225AD5@jerris.com> Likely, you wont get full call setup, but you could see the inital setup message. Alternatively, there is a way that you can setup some cards passive and you can tap and listen, depends on the hardware. > On Jan 5, 2015, at 1:49 PM, Thomas Auge wrote: > > These are hardware boxes. Can I capture the data with another ISDN card on the same BRI? > > > On 05.01.2015 15:38, Michael Jerris wrote: >> For any hope to achieve this we are going to needs some more info... preferably in the way of specifications. It's >> maybe possible if you can capture the isdn traffic from working equipment using this that you could decode it >> (Wireshark does a decent job of this) and see what exactly its signaling. >> >> >> >> >>> On Jan 5, 2015, at 1:34 PM, Thomas Auge wrote: >>> >>>> To understand this a little better. Is MP2 over data channel a standard that needs to be interoperated with? Is >>>> there details how this is signalled to the other side or are both sides just configured to support it? >>> >>> Unfortunately I don't know. I didn't know there were multiple approaches to this. The currently implemented G722 >>> method works with these boxes, so the assumption was it's all the same with other codecs. If a data channel is not >>> a common method to achieve this, then it's most likely not what they are doing. >>> >>> >>>>> On Jan 5, 2015, at 1:06 PM, Michael Jerris > wrote: >>>>> >>>>> When g722 is used over isdn, it is a voice channel marked as g722, NOT a data channel with voice codec data. >>>>> >>>>>> On Jan 2, 2015, at 5:20 PM, Thomas Auge > wrote: >>>>>> >>>>>> There are ISDN devices which establish a digital (data) ISDN connection and then use all kinds of codecs on >>>>>> top of that. Think FreeTDM with G722 - same thing. As far as I know FreeTDM doesn't care which codec you >>>>>> choose for a digital connection. G722 is just the most common use case. >>>>>> >>>>>> MP2 is still a widely used standard in broadcast, i.e. for remote contributions. >>>>>> >>>>>> On 02.01.2015 19:08, Brian West wrote: >>>>>>> I'm still puzzled as to why mp2, since ISDN is native PCM (ulaw/alaw), Do you have some legacy files or >>>>>>> systems you're working with that require this? Trying to fully understand your thought process and needs. >>>>>>> >>>>>>> On Fri, Jan 2, 2015 at 4:04 PM, Thomas Auge >>>>>> > wrote: >>>>>>> >>>>>>> Digital ISDN calls, so I think as far as freeswitch is concerned, voice calls, yes. >>>>>>> >>>>>>> >>>>>>> On 02.01.2015 18:52, Brian West wrote: >>>>>>>> In what context? playing sound files, voice calls what? >>>>>>>> >>>>>>>> On Fri, Jan 2, 2015 at 3:49 PM, Thomas Auge >>>>>>> >> >>>>>>>> wrote: >>>>>>>> >>>>>>>> Hello list, >>>>>>>> >>>>>>>> would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for it? >>>>>>>> >>>>>>>> Cheers! >>>>>>>> >>>>>>>> Thomas >>>>> >>>> >> >> >> _________________________________________________________________________ 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 From auge at virtues.net Tue Jan 6 00:04:08 2015 From: auge at virtues.net (Thomas Auge) Date: Mon, 05 Jan 2015 18:04:08 -0300 Subject: [Freeswitch-users] MP2 In-Reply-To: <9235C974-F010-45CD-A383-81D640225AD5@jerris.com> References: <54A71282.9060307@virtues.net> <54A71601.1070403@virtues.net> <54A719B6.2050609@virtues.net> <91707E19-EF35-4290-8C26-228D950200A5@jerris.com> <18E4692B-3BD9-4E61-AEEA-21F2A4CCA79A@jerris.com> <54AAD929.5070809@virtues.net> <54AADCAF.4050909@virtues.net> <9235C974-F010-45CD-A383-81D640225AD5@jerris.com> Message-ID: <54AAFC48.3030105@virtues.net> I'm trying to find some detailed specs on this, but so far no luck. There is quite an array of products of that kind out there. They must have set this up other than with trial and error ... One interesting thing about them is that interoperability depends on selecting the correct codec on both ends, so there doesn't seem to be an SDP type negotiation going on. It really seems as if these codecs are just slapped on top of a digital connection without any extra magic. On 05.01.2015 17:52, Michael Jerris wrote: > Likely, you wont get full call setup, but you could see the inital setup message. Alternatively, there is a way that > you can setup some cards passive and you can tap and listen, depends on the hardware. > >> On Jan 5, 2015, at 1:49 PM, Thomas Auge wrote: >> >> These are hardware boxes. Can I capture the data with another ISDN card on the same BRI? >> >> >> On 05.01.2015 15:38, Michael Jerris wrote: >>> For any hope to achieve this we are going to needs some more info... preferably in the way of specifications. >>> It's maybe possible if you can capture the isdn traffic from working equipment using this that you could decode >>> it (Wireshark does a decent job of this) and see what exactly its signaling. >>> >>> >>> >>> >>>> On Jan 5, 2015, at 1:34 PM, Thomas Auge wrote: >>>> >>>>> To understand this a little better. Is MP2 over data channel a standard that needs to be interoperated with? >>>>> Is there details how this is signalled to the other side or are both sides just configured to support it? >>>> >>>> Unfortunately I don't know. I didn't know there were multiple approaches to this. The currently implemented >>>> G722 method works with these boxes, so the assumption was it's all the same with other codecs. If a data >>>> channel is not a common method to achieve this, then it's most likely not what they are doing. >>>> >>>> >>>>>> On Jan 5, 2015, at 1:06 PM, Michael Jerris > wrote: >>>>>> >>>>>> When g722 is used over isdn, it is a voice channel marked as g722, NOT a data channel with voice codec >>>>>> data. >>>>>> >>>>>>> On Jan 2, 2015, at 5:20 PM, Thomas Auge > wrote: >>>>>>> >>>>>>> There are ISDN devices which establish a digital (data) ISDN connection and then use all kinds of codecs >>>>>>> on top of that. Think FreeTDM with G722 - same thing. As far as I know FreeTDM doesn't care which codec >>>>>>> you choose for a digital connection. G722 is just the most common use case. >>>>>>> >>>>>>> MP2 is still a widely used standard in broadcast, i.e. for remote contributions. >>>>>>> >>>>>>> On 02.01.2015 19:08, Brian West wrote: >>>>>>>> I'm still puzzled as to why mp2, since ISDN is native PCM (ulaw/alaw), Do you have some legacy files >>>>>>>> or systems you're working with that require this? Trying to fully understand your thought process and >>>>>>>> needs. >>>>>>>> >>>>>>>> On Fri, Jan 2, 2015 at 4:04 PM, Thomas Auge >>>>>>> > wrote: >>>>>>>> >>>>>>>> Digital ISDN calls, so I think as far as freeswitch is concerned, voice calls, yes. >>>>>>>> >>>>>>>> >>>>>>>> On 02.01.2015 18:52, Brian West wrote: >>>>>>>>> In what context? playing sound files, voice calls what? >>>>>>>>> >>>>>>>>> On Fri, Jan 2, 2015 at 3:49 PM, Thomas Auge >>>>>>>> >>>>>>>>> >> wrote: >>>>>>>>> >>>>>>>>> Hello list, >>>>>>>>> >>>>>>>>> would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for >>>>>>>>> it? >>>>>>>>> >>>>>>>>> Cheers! >>>>>>>>> >>>>>>>>> Thomas >>>>>> >>>>> >>> >>> >>> _________________________________________________________________________ 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 > > > _________________________________________________________________________ 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 > From Richard.Adams at stentofon.com.au Tue Jan 6 01:16:25 2015 From: Richard.Adams at stentofon.com.au (Richard Adams) Date: Mon, 5 Jan 2015 22:16:25 +0000 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys at gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams > wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" @sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: @sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: >;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" @sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: @sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams > wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams > wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams > wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122 at 192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122 at 192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122 at 192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] _________________________________________________________________________ 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 ... -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/3b2de3b6/attachment-0001.jpg From blasterjr at gmail.com Tue Jan 6 02:59:31 2015 From: blasterjr at gmail.com (Chris Tunbridge) Date: Mon, 5 Jan 2015 16:59:31 -0700 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams < Richard.Adams at stentofon.com.au> wrote: > I tried setting NDLB_force_rport on my mynetfone profile, but the result > is the same. > > > > The SDP still uses the private IP of the PBX. > > > > What do I need to set to re-write the O and S records in the SDP to my > public IP? > > > > o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 > > s=FreeSWITCH > > c=IN IP4 192.168.34.2 > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* Daniel Ivanov [mailto:sertys at gmail.com] > *Sent:* Monday, January 05, 2015 7:59 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Just set the NDLB_force_rport param on your mynetfone profile and be done > with it. > > 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: > > There are some exotic NAT settings on the wiki that will _force_ certain > behaviours, which may get things working and give you a starting point to > work back from. Bear in mind that these settings aren't really fixes, just > ways of telling FS to treat devices as stoopid and do the thinking for > them. > > On 5/01/2015 6:11 PM, Richard Adams wrote: > > I said practically. If they're both broken, then they're practically > the same :) > > > > I have the ext-sip-ip and ext-rtp-ip set to my public address in > internal.xml, external.xml and the mynetfone external profile. > > > > Where else do I need to set the public IP to get the SDP to report > correctly? > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com ] > *Sent:* Monday, January 05, 2015 6:09 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Not identical. > > the FROM's differ > one has X-Serialnumber & P-Key-Flags headers. > > Both of them have bad IP's in the SDP ( 192.168.34.2 ) > this will be key here.. fix this and it may fix all your problems. > > that NEEDS To have your external, REAL IP Address. > > Do you know what that IP does in the SDP ?? > > Thats what tells My Net Fone where to send the audio. > how can they route to that IP address, its not globally routable. > so its either going to fail 100% of the time, or they are going to rely on > hacks and magic to make a guess at where to send the audio. and sometimes > that magic might fail ( like your seeing ) > > Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 > ( XXX Added to protect you from more attacks :P ) > > > > > > On 5 January 2015 at 16:47, Richard Adams > wrote: > > First invite is a successful call from my desk to mobile, and the second > is from external incoming call to my mobile. > > > > The packets are practically identical. > > > > send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: > > ------------------------------------------------------------------------ > > INVITE sip:0404058798 at sip20.mynetfone.com.au SIP/2.0 > > Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr > > Max-Forwards: 33 > > From: "0397296600" >;tag=D7U82j7Qj3pKQ > > To: > > Call-ID: b36af627-0f48-1233-04b8-00188b436d1c > > CSeq: 69883976 INVITE > > Contact: ;transport=udp;gw=MyNetFone> > > User-Agent: > FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > > Supported: timer, path, replaces > > Allow-Events: talk, hold, conference, refer > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 270 > > Diversion: ;reason=unconditional > > X-FS-Support: update_display,send_info > > Remote-Party-ID: "0397296600" >;party=calling;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 > > s=FreeSWITCH > > c=IN IP4 192.168.34.2 > > t=0 0 > > m=audio 17482 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > > > send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: > > ------------------------------------------------------------------------ > > INVITE sip:0404058798 at sip20.mynetfone.com.au SIP/2.0 > > Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p > > Max-Forwards: 69 > > From: "Richard" >;tag=egN14DrUFcD6j > > To: > > Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c > > CSeq: 69884009 INVITE > > Contact: ;transport=udp;gw=MyNetFone> > > User-Agent: > FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > > Supported: timer, path, replaces > > Allow-Events: talk, hold, conference, refer > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 270 > > X-Serialnumber: 0004133887C7 > > P-Key-Flags: keys="3" > > X-FS-Support: update_display,send_info > > Remote-Party-ID: "Richard" >;party=calling;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 > > s=FreeSWITCH > > c=IN IP4 192.168.34.2 > > t=0 0 > > m=audio 17766 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com] > *Sent:* Monday, January 05, 2015 5:35 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > it might just be easier to pastebin your config files , specifically the > SIP profiles for a start. > > > > On 5 January 2015 at 16:16, Richard Adams > wrote: > > .2 is the PBX. > > > > Bypass media is not on. > > > > I'll look at the RTP-IP. > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com] > *Sent:* Monday, January 05, 2015 5:09 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > That is kinda what happens when you post IP Address in a mailing list or > online :( > > You can rest assured MyNetFone will also be getting them to their IP > address. > > > > OK so the SDP imn your 200 OK definitaly tells MNF to send the media > stream to 192.168.34.2 so either your RTP IP is not set, your router is > re-writing it with an ALG, or your telling freeswitch to bypass media or > something. > > > > What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 > or #2 > > if .34.2 is a phone handset then id guess its closer to option #3 > > > > Jay > > > > On 5 January 2015 at 15:56, Richard Adams > wrote: > > I have attached a packet trace from a failed call. > > > > We use TPG. The service is exceptional, and we've had only one downtime > event in 5 years. > > > > We don't have any call issues on a standard ADSL2+ line with all office > internet and phones on the same line. > > > > Some nefarious people are on this mailing list, as I'm now under attack. > Hooray! > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com] > *Sent:* Monday, January 05, 2015 4:48 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Im coming in late to the party here. > > > > Do you have any packet captures of this ??? > > > > It sounds like your trying to bypass the media and get out of the RTP > stream. > > Nobody is going to do that for you :) > > > > Also, from your logs I can see your on a TPG service. > > A quick test here shows 8% packet loss and average of 130ms latency with > 40ms std dev. > > > > I hope that is just the service your using for your testing ?? > > > > Jay Binks > > > > > > > > On 5 January 2015 at 15:25, Richard Adams > wrote: > > UPNP is now off. Same result. > > > > The short version of the call progress is below. For testing, I'm calling > from an internal extension 127 to the DID for the Help Line. It fails the > same with an external first caller. > > > > 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel > sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] > > 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo > <127>->86447200 in context default > > 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] > > 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/61397296600 at 125.213.160.83 > [25584398-949a-11e4-9710-192279674221] > > 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing > 61397296600 <61397296600>->0386447200 in context public > > 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer > sofia/external/61397296600 at 125.213.160.83 to XML[0386447200 at default] > > 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing > 61397296600 <61397296600>->0386447200 in context default > > 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel > sofia/internal/sip:122 at 192.168.34.20:2100 > [25594e96-949a-11e4-9727-192279674221] > > 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ > sip:122 at 192.168.34.20:2100! > > 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready > sofia/external/61397296600 at 125.213.160.83! > > 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready > sofia/external/61397296600 at 125.213.160.83! > > 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready > sofia/external/86447200! > > 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready > sofia/internal/127 at 192.168.34.2! > > 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready > sofia/internal/127 at 192.168.34.2! > > 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] > > 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot > create outgoing channel of type [USER] cause: [NO_ANSWER] > > 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. > Cause: NO_ANSWER > > 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 > (sofia/internal/sip:122 at 192.168.34.20:2100) Ended > > 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close > Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] > > 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] > > 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer > sofia/external/0404058798! > > 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ > 0404058798] > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/c8960a56/attachment-0001.jpg From brian at freeswitch.org Tue Jan 6 03:04:31 2015 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Jan 2015 18:04:31 -0600 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: SHHH don't tell them to look at the box for the credentials, Its a test to see if they are paying attention. :) On Mon, Jan 5, 2015 at 5:59 PM, Chris Tunbridge wrote: > Richard someone else asked this, and i think they were correct in asking, > can you please pastebin your config files, specifically the ones in > conf/sip_profiles/ as that'll give us the best information to help you with. > > you can use the freeswitch pastebin: http://pastebin.freeswitch.org > > Instructions/credentials are included on the box that pops up. > > At this point to me, it sounds like improper (or missing) configuration, > OR its getting messed with by the router. > > On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams < > Richard.Adams at stentofon.com.au> wrote: > >> I tried setting NDLB_force_rport on my mynetfone profile, but the >> result is the same. >> >> >> >> The SDP still uses the private IP of the PBX. >> >> >> >> What do I need to set to re-write the O and S records in the SDP to my >> public IP? >> >> >> >> o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 >> >> s=FreeSWITCH >> >> c=IN IP4 192.168.34.2 >> >> Regards, >> >> >> >> Richard Adams >> >> *Technical Manager* >> >> >> >> *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC >> 3153 ? AUSTRALIA >> >> PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 >> >> Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU >> >> >> WWW.ZENITEL.COM ? WWW.DNH.NO >> >> >> [image: cid:image001.jpg at 01CD7927.5758B850] >> >> >> >> >> >> >> >> >> >> >> >> *From:* Daniel Ivanov [mailto:sertys at gmail.com] >> *Sent:* Monday, January 05, 2015 7:59 PM >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Help required >> >> >> >> Just set the NDLB_force_rport param on your mynetfone profile and be done >> with it. >> >> 5 ???. 2015 ?. 10:30 ???????????? "Francis" >> ???????: >> >> There are some exotic NAT settings on the wiki that will _force_ certain >> behaviours, which may get things working and give you a starting point to >> work back from. Bear in mind that these settings aren't really fixes, just >> ways of telling FS to treat devices as stoopid and do the thinking for >> them. >> >> On 5/01/2015 6:11 PM, Richard Adams wrote: >> >> I said practically. If they're both broken, then they're practically >> the same :) >> >> >> >> I have the ext-sip-ip and ext-rtp-ip set to my public address in >> internal.xml, external.xml and the mynetfone external profile. >> >> >> >> Where else do I need to set the public IP to get the SDP to report >> correctly? >> >> >> >> Regards, >> >> >> >> Richard Adams >> >> *Technical Manager* >> >> >> >> *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC >> 3153 ? AUSTRALIA >> >> PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 >> >> Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU >> >> >> WWW.ZENITEL.COM ? WWW.DNH.NO >> >> >> [image: cid:image001.jpg at 01CD7927.5758B850] >> >> >> >> >> >> >> >> >> >> >> >> *From:* jay binks [mailto:jaybinks at gmail.com ] >> *Sent:* Monday, January 05, 2015 6:09 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Help required >> >> >> >> Not identical. >> >> the FROM's differ >> one has X-Serialnumber & P-Key-Flags headers. >> >> Both of them have bad IP's in the SDP ( 192.168.34.2 ) >> this will be key here.. fix this and it may fix all your problems. >> >> that NEEDS To have your external, REAL IP Address. >> >> Do you know what that IP does in the SDP ?? >> >> Thats what tells My Net Fone where to send the audio. >> how can they route to that IP address, its not globally routable. >> so its either going to fail 100% of the time, or they are going to rely >> on hacks and magic to make a guess at where to send the audio. and >> sometimes that magic might fail ( like your seeing ) >> >> Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 >> ( XXX Added to protect you from more attacks :P ) >> >> >> >> >> >> On 5 January 2015 at 16:47, Richard Adams >> wrote: >> >> First invite is a successful call from my desk to mobile, and the second >> is from external incoming call to my mobile. >> >> >> >> The packets are practically identical. >> >> >> >> send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: >> >> >> ------------------------------------------------------------------------ >> >> INVITE sip:0404058798 at sip20.mynetfone.com.au SIP/2.0 >> >> Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr >> >> Max-Forwards: 33 >> >> From: "0397296600" > >;tag=D7U82j7Qj3pKQ >> >> To: >> >> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c >> >> CSeq: 69883976 INVITE >> >> Contact: > ;transport=udp;gw=MyNetFone> >> >> User-Agent: >> FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >> >> Supported: timer, path, replaces >> >> Allow-Events: talk, hold, conference, refer >> >> Content-Type: application/sdp >> >> Content-Disposition: session >> >> Content-Length: 270 >> >> Diversion: ;reason=unconditional >> >> X-FS-Support: update_display,send_info >> >> Remote-Party-ID: "0397296600" > >;party=calling;screen=yes;privacy=off >> >> >> >> v=0 >> >> o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 >> >> s=FreeSWITCH >> >> c=IN IP4 192.168.34.2 >> >> t=0 0 >> >> m=audio 17482 RTP/AVP 0 8 3 101 13 >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:3 GSM/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=ptime:20 >> >> >> >> send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: >> >> >> ------------------------------------------------------------------------ >> >> INVITE sip:0404058798 at sip20.mynetfone.com.au SIP/2.0 >> >> Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p >> >> Max-Forwards: 69 >> >> From: "Richard" > >;tag=egN14DrUFcD6j >> >> To: >> >> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c >> >> CSeq: 69884009 INVITE >> >> Contact: > ;transport=udp;gw=MyNetFone> >> >> User-Agent: >> FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >> >> Supported: timer, path, replaces >> >> Allow-Events: talk, hold, conference, refer >> >> Content-Type: application/sdp >> >> Content-Disposition: session >> >> Content-Length: 270 >> >> X-Serialnumber: 0004133887C7 >> >> P-Key-Flags: keys="3" >> >> X-FS-Support: update_display,send_info >> >> Remote-Party-ID: "Richard" > >;party=calling;screen=yes;privacy=off >> >> >> >> v=0 >> >> o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 >> >> s=FreeSWITCH >> >> c=IN IP4 192.168.34.2 >> >> t=0 0 >> >> m=audio 17766 RTP/AVP 0 8 3 101 13 >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:3 GSM/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=ptime:20 >> >> >> >> >> >> Regards, >> >> >> >> Richard Adams >> >> *Technical Manager* >> >> >> >> *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC >> 3153 ? AUSTRALIA >> >> PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 >> >> Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU >> >> >> WWW.ZENITEL.COM ? WWW.DNH.NO >> >> >> [image: cid:image001.jpg at 01CD7927.5758B850] >> >> >> >> >> >> >> >> >> >> >> >> *From:* jay binks [mailto:jaybinks at gmail.com] >> *Sent:* Monday, January 05, 2015 5:35 PM >> >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Help required >> >> >> >> it might just be easier to pastebin your config files , specifically the >> SIP profiles for a start. >> >> >> >> On 5 January 2015 at 16:16, Richard Adams >> wrote: >> >> .2 is the PBX. >> >> >> >> Bypass media is not on. >> >> >> >> I'll look at the RTP-IP. >> >> >> >> Regards, >> >> >> >> Richard Adams >> >> *Technical Manager* >> >> >> >> *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC >> 3153 ? AUSTRALIA >> >> PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 >> >> Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU >> >> >> WWW.ZENITEL.COM ? WWW.DNH.NO >> >> >> [image: cid:image001.jpg at 01CD7927.5758B850] >> >> >> >> >> >> >> >> >> >> >> >> *From:* jay binks [mailto:jaybinks at gmail.com] >> *Sent:* Monday, January 05, 2015 5:09 PM >> >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Help required >> >> >> >> That is kinda what happens when you post IP Address in a mailing list or >> online :( >> >> You can rest assured MyNetFone will also be getting them to their IP >> address. >> >> >> >> OK so the SDP imn your 200 OK definitaly tells MNF to send the media >> stream to 192.168.34.2 so either your RTP IP is not set, your router is >> re-writing it with an ALG, or your telling freeswitch to bypass media or >> something. >> >> >> >> What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 >> or #2 >> >> if .34.2 is a phone handset then id guess its closer to option #3 >> >> >> >> Jay >> >> >> >> On 5 January 2015 at 15:56, Richard Adams >> wrote: >> >> I have attached a packet trace from a failed call. >> >> >> >> We use TPG. The service is exceptional, and we've had only one downtime >> event in 5 years. >> >> >> >> We don't have any call issues on a standard ADSL2+ line with all office >> internet and phones on the same line. >> >> >> >> Some nefarious people are on this mailing list, as I'm now under attack. >> Hooray! >> >> >> >> Regards, >> >> >> >> Richard Adams >> >> *Technical Manager* >> >> >> >> *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC >> 3153 ? AUSTRALIA >> >> PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 >> >> Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU >> >> >> WWW.ZENITEL.COM ? WWW.DNH.NO >> >> >> [image: cid:image001.jpg at 01CD7927.5758B850] >> >> >> >> >> >> >> >> >> >> >> >> *From:* jay binks [mailto:jaybinks at gmail.com] >> *Sent:* Monday, January 05, 2015 4:48 PM >> >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Help required >> >> >> >> Im coming in late to the party here. >> >> >> >> Do you have any packet captures of this ??? >> >> >> >> It sounds like your trying to bypass the media and get out of the RTP >> stream. >> >> Nobody is going to do that for you :) >> >> >> >> Also, from your logs I can see your on a TPG service. >> >> A quick test here shows 8% packet loss and average of 130ms latency with >> 40ms std dev. >> >> >> >> I hope that is just the service your using for your testing ?? >> >> >> >> Jay Binks >> >> >> >> >> >> >> >> On 5 January 2015 at 15:25, Richard Adams >> wrote: >> >> UPNP is now off. Same result. >> >> >> >> The short version of the call progress is below. For testing, I'm >> calling from an internal extension 127 to the DID for the Help Line. It >> fails the same with an external first caller. >> >> >> >> 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel >> sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] >> >> 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo >> <127>->86447200 in context default >> >> 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel >> sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] >> >> 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel >> sofia/external/61397296600 at 125.213.160.83 >> [25584398-949a-11e4-9710-192279674221] >> >> 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing >> 61397296600 <61397296600>->0386447200 in context public >> >> 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer >> sofia/external/61397296600 at 125.213.160.83 to XML[0386447200 at default] >> >> 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing >> 61397296600 <61397296600>->0386447200 in context default >> >> 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel >> sofia/internal/sip:122 at 192.168.34.20:2100 >> [25594e96-949a-11e4-9727-192279674221] >> >> 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready >> sofia/internal/sip:122 at 192.168.34.20:2100! >> >> 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready >> sofia/external/61397296600 at 125.213.160.83! >> >> 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready >> sofia/external/61397296600 at 125.213.160.83! >> >> 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready >> sofia/external/86447200! >> >> 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready >> sofia/internal/127 at 192.168.34.2! >> >> 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready >> sofia/internal/127 at 192.168.34.2! >> >> 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup >> sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] >> >> 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot >> create outgoing channel of type [USER] cause: [NO_ANSWER] >> >> 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. >> Cause: NO_ANSWER >> >> 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 >> (sofia/internal/sip:122 at 192.168.34.20:2100) Ended >> >> 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close >> Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] >> >> 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel >> sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] >> >> 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer >> sofia/external/0404058798! >> >> 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ >> 0404058798] >> >> >> _________________________________________________________________________ >> 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 >> > > > _________________________________________________________________________ > 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/20150105/4ee4bd29/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/4ee4bd29/attachment-0001.jpg From Richard.Adams at stentofon.com.au Tue Jan 6 03:15:17 2015 From: Richard.Adams at stentofon.com.au (Richard Adams) Date: Tue, 6 Jan 2015 00:15:17 +0000 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr at gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams > wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys at gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams > wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" @sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: @sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: >;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" @sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: @sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams > wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams > wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams > wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122 at 192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122 at 192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122 at 192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/852a73f1/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/852a73f1/attachment-0001.jpg From Richard.Adams at stentofon.com.au Tue Jan 6 03:28:27 2015 From: Richard.Adams at stentofon.com.au (Richard Adams) Date: Tue, 6 Jan 2015 00:28:27 +0000 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: I'm stupid, but not quite that stupid... Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Brian West [mailto:brian at freeswitch.org] Sent: Tuesday, January 06, 2015 11:05 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required SHHH don't tell them to look at the box for the credentials, Its a test to see if they are paying attention. :) On Mon, Jan 5, 2015 at 5:59 PM, Chris Tunbridge > wrote: Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams > wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys at gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams > wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" @sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: @sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: >;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" @sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: @sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams > wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams > wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams > wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122 at 192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122 at 192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122 at 192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] _________________________________________________________________________ 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 _________________________________________________________________________ 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 [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] 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/20150106/0a0ab511/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/0a0ab511/attachment-0001.jpg From max at nysolutions.com Tue Jan 6 03:30:40 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Tue, 6 Jan 2015 00:30:40 +0000 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr at gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams > wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys at gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams > wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" @sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: @sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: >;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" @sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: @sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams > wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams > wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams > wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122 at 192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122 at 192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122 at 192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/facc4124/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 2424 bytes Desc: image002.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/facc4124/attachment-0002.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 4139 bytes Desc: image003.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/facc4124/attachment-0003.jpg From brian at freeswitch.org Tue Jan 6 03:32:17 2015 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Jan 2015 18:32:17 -0600 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: LOL, Wasn't implying you were... I actually do use that as a test to see if someone is paying attention or trying to go too fast without fully understanding whats going on.... missing that usually means something else was missed along the way. On Mon, Jan 5, 2015 at 6:28 PM, Richard Adams < Richard.Adams at stentofon.com.au> wrote: > I'm stupid, but not quite that stupid... > > > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* Brian West [mailto:brian at freeswitch.org] > *Sent:* Tuesday, January 06, 2015 11:05 AM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > SHHH don't tell them to look at the box for the credentials, Its a test to > see if they are paying attention. :) > > > > On Mon, Jan 5, 2015 at 5:59 PM, Chris Tunbridge > wrote: > > Richard someone else asked this, and i think they were correct in asking, > can you please pastebin your config files, specifically the ones in > conf/sip_profiles/ as that'll give us the best information to help you with. > > you can use the freeswitch pastebin: http://pastebin.freeswitch.org > > Instructions/credentials are included on the box that pops up. > > At this point to me, it sounds like improper (or missing) configuration, > OR its getting messed with by the router. > > > > On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams < > Richard.Adams at stentofon.com.au> wrote: > > I tried setting NDLB_force_rport on my mynetfone profile, but the result > is the same. > > > > The SDP still uses the private IP of the PBX. > > > > What do I need to set to re-write the O and S records in the SDP to my > public IP? > > > > o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 > > s=FreeSWITCH > > c=IN IP4 192.168.34.2 > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* Daniel Ivanov [mailto:sertys at gmail.com] > *Sent:* Monday, January 05, 2015 7:59 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Just set the NDLB_force_rport param on your mynetfone profile and be done > with it. > > 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: > > There are some exotic NAT settings on the wiki that will _force_ certain > behaviours, which may get things working and give you a starting point to > work back from. Bear in mind that these settings aren't really fixes, just > ways of telling FS to treat devices as stoopid and do the thinking for > them. > > On 5/01/2015 6:11 PM, Richard Adams wrote: > > I said practically. If they're both broken, then they're practically > the same :) > > > > I have the ext-sip-ip and ext-rtp-ip set to my public address in > internal.xml, external.xml and the mynetfone external profile. > > > > Where else do I need to set the public IP to get the SDP to report > correctly? > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com ] > *Sent:* Monday, January 05, 2015 6:09 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Not identical. > > the FROM's differ > one has X-Serialnumber & P-Key-Flags headers. > > Both of them have bad IP's in the SDP ( 192.168.34.2 ) > this will be key here.. fix this and it may fix all your problems. > > that NEEDS To have your external, REAL IP Address. > > Do you know what that IP does in the SDP ?? > > Thats what tells My Net Fone where to send the audio. > how can they route to that IP address, its not globally routable. > so its either going to fail 100% of the time, or they are going to rely on > hacks and magic to make a guess at where to send the audio. and sometimes > that magic might fail ( like your seeing ) > > Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 > ( XXX Added to protect you from more attacks :P ) > > > > > > On 5 January 2015 at 16:47, Richard Adams > wrote: > > First invite is a successful call from my desk to mobile, and the second > is from external incoming call to my mobile. > > > > The packets are practically identical. > > > > send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: > > ------------------------------------------------------------------------ > > INVITE sip:0404058798 at sip20.mynetfone.com.au SIP/2.0 > > Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr > > Max-Forwards: 33 > > From: "0397296600" >;tag=D7U82j7Qj3pKQ > > To: > > Call-ID: b36af627-0f48-1233-04b8-00188b436d1c > > CSeq: 69883976 INVITE > > Contact: ;transport=udp;gw=MyNetFone> > > User-Agent: > FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > > Supported: timer, path, replaces > > Allow-Events: talk, hold, conference, refer > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 270 > > Diversion: ;reason=unconditional > > X-FS-Support: update_display,send_info > > Remote-Party-ID: "0397296600" >;party=calling;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 > > s=FreeSWITCH > > c=IN IP4 192.168.34.2 > > t=0 0 > > m=audio 17482 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > > > send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: > > ------------------------------------------------------------------------ > > INVITE sip:0404058798 at sip20.mynetfone.com.au SIP/2.0 > > Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p > > Max-Forwards: 69 > > From: "Richard" >;tag=egN14DrUFcD6j > > To: > > Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c > > CSeq: 69884009 INVITE > > Contact: ;transport=udp;gw=MyNetFone> > > User-Agent: > FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > > Supported: timer, path, replaces > > Allow-Events: talk, hold, conference, refer > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 270 > > X-Serialnumber: 0004133887C7 > > P-Key-Flags: keys="3" > > X-FS-Support: update_display,send_info > > Remote-Party-ID: "Richard" >;party=calling;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 > > s=FreeSWITCH > > c=IN IP4 192.168.34.2 > > t=0 0 > > m=audio 17766 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com] > *Sent:* Monday, January 05, 2015 5:35 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > it might just be easier to pastebin your config files , specifically the > SIP profiles for a start. > > > > On 5 January 2015 at 16:16, Richard Adams > wrote: > > .2 is the PBX. > > > > Bypass media is not on. > > > > I'll look at the RTP-IP. > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com] > *Sent:* Monday, January 05, 2015 5:09 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > That is kinda what happens when you post IP Address in a mailing list or > online :( > > You can rest assured MyNetFone will also be getting them to their IP > address. > > > > OK so the SDP imn your 200 OK definitaly tells MNF to send the media > stream to 192.168.34.2 so either your RTP IP is not set, your router is > re-writing it with an ALG, or your telling freeswitch to bypass media or > something. > > > > What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 > or #2 > > if .34.2 is a phone handset then id guess its closer to option #3 > > > > Jay > > > > On 5 January 2015 at 15:56, Richard Adams > wrote: > > I have attached a packet trace from a failed call. > > > > We use TPG. The service is exceptional, and we've had only one downtime > event in 5 years. > > > > We don't have any call issues on a standard ADSL2+ line with all office > internet and phones on the same line. > > > > Some nefarious people are on this mailing list, as I'm now under attack. > Hooray! > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com] > *Sent:* Monday, January 05, 2015 4:48 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Im coming in late to the party here. > > > > Do you have any packet captures of this ??? > > > > It sounds like your trying to bypass the media and get out of the RTP > stream. > > Nobody is going to do that for you :) > > > > Also, from your logs I can see your on a TPG service. > > A quick test here shows 8% packet loss and average of 130ms latency with > 40ms std dev. > > > > I hope that is just the service your using for your testing ?? > > > > Jay Binks > > > > > > > > On 5 January 2015 at 15:25, Richard Adams > wrote: > > UPNP is now off. Same result. > > > > The short version of the call progress is below. For testing, I'm calling > from an internal extension 127 to the DID for the Help Line. It fails the > same with an external first caller. > > > > 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel > sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] > > 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo > <127>->86447200 in context default > > 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] > > 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/61397296600 at 125.213.160.83 > [25584398-949a-11e4-9710-192279674221] > > 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing > 61397296600 <61397296600>->0386447200 in context public > > 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer > sofia/external/61397296600 at 125.213.160.83 to XML[0386447200 at default] > > 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing > 61397296600 <61397296600>->0386447200 in context default > > 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel > sofia/internal/sip:122 at 192.168.34.20:2100 > [25594e96-949a-11e4-9727-192279674221] > > 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ > sip:122 at 192.168.34.20:2100! > > 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready > sofia/external/61397296600 at 125.213.160.83! > > 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready > sofia/external/61397296600 at 125.213.160.83! > > 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready > sofia/external/86447200! > > 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready > sofia/internal/127 at 192.168.34.2! > > 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready > sofia/internal/127 at 192.168.34.2! > > 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] > > 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot > create outgoing channel of type [USER] cause: [NO_ANSWER] > > 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. > Cause: NO_ANSWER > > 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 > (sofia/internal/sip:122 at 192.168.34.20:2100) Ended > > 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close > Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] > > 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] > > 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer > sofia/external/0404058798! > > 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ > 0404058798] > > > _________________________________________________________________________ > 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 > > > > > _________________________________________________________________________ > 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 > -- *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/b1c75678/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/b1c75678/attachment-0001.jpg From Richard.Adams at stentofon.com.au Tue Jan 6 04:14:06 2015 From: Richard.Adams at stentofon.com.au (Richard Adams) Date: Tue, 6 Jan 2015 01:14:06 +0000 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: I enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Moishe Grunstein [mailto:max at nysolutions.com] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr at gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams > wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys at gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams > wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" @sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: @sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: >;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" @sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: @sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams > wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams > wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams > wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122 at 192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122 at 192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122 at 192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/1b317bd5/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/1b317bd5/attachment-0002.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 2424 bytes Desc: image002.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/1b317bd5/attachment-0003.jpg From max at nysolutions.com Tue Jan 6 04:39:22 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Tue, 6 Jan 2015 01:39:22 +0000 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: You should on your internal profile. It actually is in there just uncomment it. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 8:14 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required I enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Moishe Grunstein [mailto:max at nysolutions.com] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr at gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams > wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys at gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams > wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" @sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: @sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: >;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" @sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: @sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams > wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams > wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams > wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122 at 192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122 at 192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122 at 192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/6577ae60/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 2424 bytes Desc: image002.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/6577ae60/attachment-0002.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 4139 bytes Desc: image003.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/6577ae60/attachment-0003.jpg From wesleyakio at tuntscorp.com Tue Jan 6 04:46:54 2015 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Mon, 5 Jan 2015 23:46:54 -0200 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: Could you please post the result to sofia status? Best, Wesley Akio Sent from mobile Em 05/01/2015 23:14, "Richard Adams" escreveu: > I enabled this on the External.xml to no change. > > > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* Moishe Grunstein [mailto:max at nysolutions.com] > *Sent:* Tuesday, January 06, 2015 11:31 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > belongs on the profile not > sip gateway. > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Richard > Adams > *Sent:* Monday, January 5, 2015 7:15 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > https://pastebin.freeswitch.org/23822 for Internal.xml > > https://pastebin.freeswitch.org/23823 for External.xml > > https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* Chris Tunbridge [mailto:blasterjr at gmail.com ] > > *Sent:* Tuesday, January 06, 2015 11:00 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Richard someone else asked this, and i think they were correct in asking, > can you please pastebin your config files, specifically the ones in > conf/sip_profiles/ as that'll give us the best information to help you with. > > you can use the freeswitch pastebin: http://pastebin.freeswitch.org > > Instructions/credentials are included on the box that pops up. > > At this point to me, it sounds like improper (or missing) configuration, > OR its getting messed with by the router. > > > > On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams < > Richard.Adams at stentofon.com.au> wrote: > > I tried setting NDLB_force_rport on my mynetfone profile, but the result > is the same. > > > > The SDP still uses the private IP of the PBX. > > > > What do I need to set to re-write the O and S records in the SDP to my > public IP? > > > > o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 > > s=FreeSWITCH > > c=IN IP4 192.168.34.2 > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* Daniel Ivanov [mailto:sertys at gmail.com] > *Sent:* Monday, January 05, 2015 7:59 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Just set the NDLB_force_rport param on your mynetfone profile and be done > with it. > > 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: > > There are some exotic NAT settings on the wiki that will _force_ certain > behaviours, which may get things working and give you a starting point to > work back from. Bear in mind that these settings aren't really fixes, just > ways of telling FS to treat devices as stoopid and do the thinking for > them. > > On 5/01/2015 6:11 PM, Richard Adams wrote: > > I said practically. If they're both broken, then they're practically > the same :) > > > > I have the ext-sip-ip and ext-rtp-ip set to my public address in > internal.xml, external.xml and the mynetfone external profile. > > > > Where else do I need to set the public IP to get the SDP to report > correctly? > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com ] > *Sent:* Monday, January 05, 2015 6:09 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Not identical. > > the FROM's differ > one has X-Serialnumber & P-Key-Flags headers. > > Both of them have bad IP's in the SDP ( 192.168.34.2 ) > this will be key here.. fix this and it may fix all your problems. > > that NEEDS To have your external, REAL IP Address. > > Do you know what that IP does in the SDP ?? > > Thats what tells My Net Fone where to send the audio. > how can they route to that IP address, its not globally routable. > so its either going to fail 100% of the time, or they are going to rely on > hacks and magic to make a guess at where to send the audio. and sometimes > that magic might fail ( like your seeing ) > > Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 > ( XXX Added to protect you from more attacks :P ) > > > > > > On 5 January 2015 at 16:47, Richard Adams > wrote: > > First invite is a successful call from my desk to mobile, and the second > is from external incoming call to my mobile. > > > > The packets are practically identical. > > > > send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: > > ------------------------------------------------------------------------ > > INVITE sip:0404058798 at sip20.mynetfone.com.au SIP/2.0 > > Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr > > Max-Forwards: 33 > > From: "0397296600" >;tag=D7U82j7Qj3pKQ > > To: > > Call-ID: b36af627-0f48-1233-04b8-00188b436d1c > > CSeq: 69883976 INVITE > > Contact: ;transport=udp;gw=MyNetFone> > > User-Agent: > FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > > Supported: timer, path, replaces > > Allow-Events: talk, hold, conference, refer > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 270 > > Diversion: ;reason=unconditional > > X-FS-Support: update_display,send_info > > Remote-Party-ID: "0397296600" >;party=calling;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 > > s=FreeSWITCH > > c=IN IP4 192.168.34.2 > > t=0 0 > > m=audio 17482 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > > > send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: > > ------------------------------------------------------------------------ > > INVITE sip:0404058798 at sip20.mynetfone.com.au SIP/2.0 > > Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p > > Max-Forwards: 69 > > From: "Richard" >;tag=egN14DrUFcD6j > > To: > > Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c > > CSeq: 69884009 INVITE > > Contact: ;transport=udp;gw=MyNetFone> > > User-Agent: > FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > > Supported: timer, path, replaces > > Allow-Events: talk, hold, conference, refer > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 270 > > X-Serialnumber: 0004133887C7 > > P-Key-Flags: keys="3" > > X-FS-Support: update_display,send_info > > Remote-Party-ID: "Richard" >;party=calling;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 > > s=FreeSWITCH > > c=IN IP4 192.168.34.2 > > t=0 0 > > m=audio 17766 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com] > *Sent:* Monday, January 05, 2015 5:35 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > it might just be easier to pastebin your config files , specifically the > SIP profiles for a start. > > > > On 5 January 2015 at 16:16, Richard Adams > wrote: > > .2 is the PBX. > > > > Bypass media is not on. > > > > I'll look at the RTP-IP. > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com] > *Sent:* Monday, January 05, 2015 5:09 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > That is kinda what happens when you post IP Address in a mailing list or > online :( > > You can rest assured MyNetFone will also be getting them to their IP > address. > > > > OK so the SDP imn your 200 OK definitaly tells MNF to send the media > stream to 192.168.34.2 so either your RTP IP is not set, your router is > re-writing it with an ALG, or your telling freeswitch to bypass media or > something. > > > > What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 > or #2 > > if .34.2 is a phone handset then id guess its closer to option #3 > > > > Jay > > > > On 5 January 2015 at 15:56, Richard Adams > wrote: > > I have attached a packet trace from a failed call. > > > > We use TPG. The service is exceptional, and we've had only one downtime > event in 5 years. > > > > We don't have any call issues on a standard ADSL2+ line with all office > internet and phones on the same line. > > > > Some nefarious people are on this mailing list, as I'm now under attack. > Hooray! > > > > Regards, > > > > Richard Adams > > *Technical Manager* > > > > *STENTOFON AUSTRALIA* ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC > 3153 ? AUSTRALIA > > PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 > > Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU > > > WWW.ZENITEL.COM ? WWW.DNH.NO > > > [image: cid:image001.jpg at 01CD7927.5758B850] > > > > > > > > > > > > *From:* jay binks [mailto:jaybinks at gmail.com] > *Sent:* Monday, January 05, 2015 4:48 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help required > > > > Im coming in late to the party here. > > > > Do you have any packet captures of this ??? > > > > It sounds like your trying to bypass the media and get out of the RTP > stream. > > Nobody is going to do that for you :) > > > > Also, from your logs I can see your on a TPG service. > > A quick test here shows 8% packet loss and average of 130ms latency with > 40ms std dev. > > > > I hope that is just the service your using for your testing ?? > > > > Jay Binks > > > > > > > > On 5 January 2015 at 15:25, Richard Adams > wrote: > > UPNP is now off. Same result. > > > > The short version of the call progress is below. For testing, I'm calling > from an internal extension 127 to the DID for the Help Line. It fails the > same with an external first caller. > > > > 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel > sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] > > 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo > <127>->86447200 in context default > > 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] > > 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/61397296600 at 125.213.160.83 > [25584398-949a-11e4-9710-192279674221] > > 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing > 61397296600 <61397296600>->0386447200 in context public > > 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer > sofia/external/61397296600 at 125.213.160.83 to XML[0386447200 at default] > > 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing > 61397296600 <61397296600>->0386447200 in context default > > 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel > sofia/internal/sip:122 at 192.168.34.20:2100 > [25594e96-949a-11e4-9727-192279674221] > > 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ > sip:122 at 192.168.34.20:2100! > > 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready > sofia/external/61397296600 at 125.213.160.83! > > 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready > sofia/external/61397296600 at 125.213.160.83! > > 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready > sofia/external/86447200! > > 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready > sofia/internal/127 at 192.168.34.2! > > 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready > sofia/internal/127 at 192.168.34.2! > > 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] > > 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot > create outgoing channel of type [USER] cause: [NO_ANSWER] > > 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. > Cause: NO_ANSWER > > 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 > (sofia/internal/sip:122 at 192.168.34.20:2100) Ended > > 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close > Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] > > 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] > > 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer > sofia/external/0404058798! > > 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ > 0404058798] > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150105/6dd2ef77/attachment-0003.jpg From Richard.Adams at stentofon.com.au Tue Jan 6 04:53:15 2015 From: Richard.Adams at stentofon.com.au (Richard Adams) Date: Tue, 6 Jan 2015 01:53:15 +0000 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia at xxx.243.120.79:5080 RUNNING (0) external::dialalphacomcompletenetwork gateway sip:STENTOFON at 192.168.34.5:5080 NOREG external::dialalphacom gateway sip:STENTOFON at 192.168.34.4:5080 NOREG external::MyNetFone gateway sip:0397296600 at sip20.mynetfone.com.au NOREG 192.168.34.2 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia at xxx.243.120.79:5060 RUNNING (0) ================================================================================================= 3 profiles 1 alias Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Wesley Akio [mailto:wesleyakio at tuntscorp.com] Sent: Tuesday, January 06, 2015 12:47 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Could you please post the result to sofia status? Best, Wesley Akio Sent from mobile Em 05/01/2015 23:14, "Richard Adams" > escreveu: I enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Moishe Grunstein [mailto:max at nysolutions.com] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr at gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams > wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys at gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams > wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" @sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: @sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: >;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" @sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: @sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams > wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams > wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams > wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122 at 192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122 at 192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122 at 192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] _________________________________________________________________________ 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/5dca876e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/5dca876e/attachment-0002.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 2424 bytes Desc: image002.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/5dca876e/attachment-0003.jpg From max at nysolutions.com Tue Jan 6 05:00:06 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Tue, 6 Jan 2015 02:00:06 +0000 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: Are you on IRC? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 8:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia at xxx.243.120.79:5080 RUNNING (0) external::dialalphacomcompletenetwork gateway sip:STENTOFON at 192.168.34.5:5080 NOREG external::dialalphacom gateway sip:STENTOFON at 192.168.34.4:5080 NOREG external::MyNetFone gateway sip:0397296600 at sip20.mynetfone.com.au NOREG 192.168.34.2 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia at xxx.243.120.79:5060 RUNNING (0) ================================================================================================= 3 profiles 1 alias Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Wesley Akio [mailto:wesleyakio at tuntscorp.com] Sent: Tuesday, January 06, 2015 12:47 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Could you please post the result to sofia status? Best, Wesley Akio Sent from mobile Em 05/01/2015 23:14, "Richard Adams" > escreveu: I enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Moishe Grunstein [mailto:max at nysolutions.com] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr at gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams > wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys at gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams > wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" @sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: @sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: >;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" @sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: @sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams > wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams > wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams > wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122 at 192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122 at 192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122 at 192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] _________________________________________________________________________ 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/371b0646/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 2424 bytes Desc: image002.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/371b0646/attachment-0002.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 4139 bytes Desc: image003.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/371b0646/attachment-0003.jpg From Richard.Adams at stentofon.com.au Tue Jan 6 05:02:19 2015 From: Richard.Adams at stentofon.com.au (Richard Adams) Date: Tue, 6 Jan 2015 02:02:19 +0000 Subject: [Freeswitch-users] Help required In-Reply-To: References: <54AA1A81.1060302@icefire.qza.net.au> <54AA4B87.5000800@icefire.qza.net.au> Message-ID: Yes, am now as STENTOFON Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Moishe Grunstein [mailto:max at nysolutions.com] Sent: Tuesday, January 06, 2015 1:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Are you on IRC? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 8:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia at xxx.243.120.79:5080 RUNNING (0) external::dialalphacomcompletenetwork gateway sip:STENTOFON at 192.168.34.5:5080 NOREG external::dialalphacom gateway sip:STENTOFON at 192.168.34.4:5080 NOREG external::MyNetFone gateway sip:0397296600 at sip20.mynetfone.com.au NOREG 192.168.34.2 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia at xxx.243.120.79:5060 RUNNING (0) ================================================================================================= 3 profiles 1 alias Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Wesley Akio [mailto:wesleyakio at tuntscorp.com] Sent: Tuesday, January 06, 2015 12:47 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Could you please post the result to sofia status? Best, Wesley Akio Sent from mobile Em 05/01/2015 23:14, "Richard Adams" > escreveu: I enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Moishe Grunstein [mailto:max at nysolutions.com] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr at gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams > wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys at gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 ???. 2015 ?. 10:30 ???????????? "Francis" > ???????: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams > wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" @sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: @sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: >;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" @sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: @sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: @123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" @sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams > wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams > wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798 Richard.Adams at STENTOFON.COM.AU ? WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ? WWW.DNH.NO [cid:image001.jpg at 01CD7927.5758B850] From: jay binks [mailto:jaybinks at gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams > wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127 at 192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122 at 192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122 at 192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127 at 192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122 at 192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122 at 192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122 at 192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] _________________________________________________________________________ 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/b96fbee7/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 4139 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/b96fbee7/attachment-0002.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 2424 bytes Desc: image002.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/b96fbee7/attachment-0003.jpg From GeorgePhelps at gfphelps.com Tue Jan 6 14:41:26 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 6 Jan 2015 06:41:26 -0500 Subject: [Freeswitch-users] Google Voice Support? Message-ID: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> Now that Google has de-supported its XMPP connectivity to Google Voice, what is the Freeswitch configuration for interfacing to Google Voice? Thanks, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/65cdbfae/attachment.html From sertys at gmail.com Tue Jan 6 15:06:02 2015 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 6 Jan 2015 13:06:02 +0100 Subject: [Freeswitch-users] Google Voice Support? In-Reply-To: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> References: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> Message-ID: None? On Tue, Jan 6, 2015 at 12:41 PM, George F. Phelps wrote: > Now that Google has de-supported its XMPP connectivity to Google Voice, > what is the Freeswitch configuration for interfacing to Google Voice? > > > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/dafdfcf2/attachment.html From GeorgePhelps at gfphelps.com Tue Jan 6 15:23:32 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 6 Jan 2015 07:23:32 -0500 Subject: [Freeswitch-users] Google Voice Support? In-Reply-To: References: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> Message-ID: <091301d029ab$965b59e0$c3120da0$@gfphelps.com> Are there plans to add Freeswitch support the new Google Voice interface? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel Ivanov Sent: Tuesday, January 06, 2015 7:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Google Voice Support? None? On Tue, Jan 6, 2015 at 12:41 PM, George F. Phelps wrote: Now that Google has de-supported its XMPP connectivity to Google Voice, what is the Freeswitch configuration for interfacing to Google Voice? 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/3e54fbd3/attachment.html From GeorgePhelps at gfphelps.com Tue Jan 6 16:57:45 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 6 Jan 2015 08:57:45 -0500 Subject: [Freeswitch-users] Multiple P-Asserted-Identity Headers In-Reply-To: <7CA4DE25-0C25-4AFA-B4C2-6D887A3A328F@jerris.com> References: <076a01d028de$96357180$c2a05480$@gfphelps.com> <7CA4DE25-0C25-4AFA-B4C2-6D887A3A328F@jerris.com> Message-ID: <093d01d029b8$bfe63200$3fb29600$@gfphelps.com> Michael Jerris, Thanks for pointing out the comma delimited syntax. However? I am still trying to get setting my outbound caller ID to work. I noticed, in this Freeswitch SIP trace? send 1202 bytes to udp/[66.33.147.150]:5060 at 07:12:09.474756: ------------------------------------------------------------------------ INVITE sip:1770XXXXXXX at 66.33.147.150 SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKDN5acD6Zcp2g Max-Forwards: 69 From: "George F Phelps" ;tag=gpmpmpp142SFH To: Call-ID: 146d208a-1040-1233-1bbe-0a1aa9c1784d CSeq: 69937100 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 251 P-Asserted-ID: "George F Phelps", , tel:+1404XXXXXXX X-FS-Support: update_display,send_info P-Asserted-Identity: "George F Phelps" v=0 o=FreeSWITCH 1420520731 1420520732 IN IP4 54.174.255.168 s=FreeSWITCH c=IN IP4 54.174.255.168 t=0 0 m=audio 25598 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ ?that there is apparently both a ?P-Asserted-Identity? and a ?P-Asserted-ID? header. What?s up with this? Are there two different headers? Is the Freeswitch SIP trace a formatted dump of the SIP packet, or is it a series of debug printf statements from where the data is added to the packet? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, January 05, 2015 1:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple P-Asserted-Identity Headers put them together as a single header separated by a comma. http://www.ietf.org/mail-archive/web/sip/current/msg17485.html has more info. On Jan 5, 2015, at 6:56 AM, George F. Phelps wrote: How do I create two (2) unique, P-Asserted-Identity headers? As required by RFC 3325: A P-Asserted-Identity header field value MUST consist of exactly one name-addr or addr-spec. There may be one or two P-Asserted-Identity values. If there is one value, it MUST be a sip, sips, or tel URI. If there are two values, one value MUST be a sip or sips URI and the other MUST be a tel URI. A valid, multiple P-Asserted-Identity headers example, taken from RFC 3325: INVITE sip:+14085551212 at proxy.pstn.net SIP/2.0 Via: SIP/2.0/TCP useragent.cisco.com;branch=z9hG4bK-124 Via: SIP/2.0/TCP proxy.cisco.com;branch=z9hG4bK-abc To: < sip:+14085551212 at cisco.com> From: "Anonymous" < sip:anonymous at anonymous.invalid>;tag=9802748 Call-ID: 245780247857024504 CSeq: 2 INVITE Max-Forwards: 69 P-Asserted-Identity: "Cullen Jennings" < sip:fluffy at cisco.com> P-Asserted-Identity: tel:+14085264000 Privacy: id When I use the following two (2) statements in a Freeswitch dialplan, the second ?set,? of course, overwrites the data stored by the first ?set.? 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/f42521ed/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/20150106/f42521ed/attachment-0001.bin From brian at freeswitch.org Tue Jan 6 17:37:42 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2015 08:37:42 -0600 Subject: [Freeswitch-users] Google Voice Support? In-Reply-To: <091301d029ab$965b59e0$c3120da0$@gfphelps.com> References: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> <091301d029ab$965b59e0$c3120da0$@gfphelps.com> Message-ID: Is the protocol documented or been reversed yet? If not then I doubt it. On Tue, Jan 6, 2015 at 6:23 AM, George F. Phelps wrote: > Are there plans to add Freeswitch support the new Google Voice interface? > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Daniel > Ivanov > *Sent:* Tuesday, January 06, 2015 7:06 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Google Voice Support? > > > > None? > > > > On Tue, Jan 6, 2015 at 12:41 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > Now that Google has de-supported its XMPP connectivity to Google Voice, > what is the Freeswitch configuration for interfacing to Google Voice? > > > > 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 > > > > _________________________________________________________________________ > 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/20150106/0a82cdac/attachment.html From sertys at gmail.com Tue Jan 6 17:45:28 2015 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 6 Jan 2015 15:45:28 +0100 Subject: [Freeswitch-users] Google Voice Support? In-Reply-To: References: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> <091301d029ab$965b59e0$c3120da0$@gfphelps.com> Message-ID: Given the efforts the community is putting on keeping integration with Skype via mod_skypopen, i doubt someone is gonna be trying to maintain another closed protocol endpoint. On Tue, Jan 6, 2015 at 3:37 PM, Brian West wrote: > Is the protocol documented or been reversed yet? If not then I doubt it. > > On Tue, Jan 6, 2015 at 6:23 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > >> Are there plans to add Freeswitch support the new Google Voice interface? >> >> >> >> Thanks, >> >> >> >> George >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Daniel >> Ivanov >> *Sent:* Tuesday, January 06, 2015 7:06 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Google Voice Support? >> >> >> >> None? >> >> >> >> On Tue, Jan 6, 2015 at 12:41 PM, George F. Phelps < >> GeorgePhelps at gfphelps.com> wrote: >> >> Now that Google has de-supported its XMPP connectivity to Google Voice, >> what is the Freeswitch configuration for interfacing to Google Voice? >> >> >> >> 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 >> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/fd14080f/attachment.html From nickolayr at gmail.com Tue Jan 6 18:05:35 2015 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Tue, 6 Jan 2015 10:05:35 -0500 Subject: [Freeswitch-users] Google Voice Support? In-Reply-To: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> References: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> Message-ID: Still use mod_dingaling for Google Voice as my free home phone, works fine. -- Best regards Rogoshchenkov Nikolay On Tue, Jan 6, 2015 at 6:41 AM, George F. Phelps wrote: > Now that Google has de-supported its XMPP connectivity to Google Voice, > what is the Freeswitch configuration for interfacing to Google Voice? > > > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/500dc365/attachment-0001.html From sertys at gmail.com Tue Jan 6 18:20:15 2015 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 6 Jan 2015 16:20:15 +0100 Subject: [Freeswitch-users] Google Voice Support? In-Reply-To: References: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> Message-ID: Oh (facepalm). On Tue, Jan 6, 2015 at 4:05 PM, Nikolay Rogoshchenkov wrote: > Still use mod_dingaling > for Google Voice as my free home phone, works fine. > > > -- > Best regards > Rogoshchenkov Nikolay > > On Tue, Jan 6, 2015 at 6:41 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > >> Now that Google has de-supported its XMPP connectivity to Google Voice, >> what is the Freeswitch configuration for interfacing to Google Voice? >> >> >> >> 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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/c170fc9d/attachment.html From brian at freeswitch.org Tue Jan 6 18:51:05 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2015 09:51:05 -0600 Subject: [Freeswitch-users] Control of IPv6 vs. IPv4 In-Reply-To: <079101d028e8$0a47b660$1ed72320$@gfphelps.com> References: <042501d026a5$74d7e670$5e87b350$@gfphelps.com> <046a01d026b7$a6d67e90$f4837bb0$@gfphelps.com> <079101d028e8$0a47b660$1ed72320$@gfphelps.com> Message-ID: Its not a bug, FreeSWITCH did exactly what you told it to do. On Mon, Jan 5, 2015 at 7:03 AM, George F. Phelps wrote: > RESOLVED > > > > The specific problem was, that I was defining my gateway in directory ? > /usr/local/freeswitch/conf/directory/default?. This was causing > Freeswitch to attempt registrations for both IPv4 and IPv6 ? which my VoIP > provider supports. A Freeswitch bug, IMHO, due to unexpected, > nondeterministic behavior. > > > > I was mistakenly doing this (i.e., defining my gateway in directory ? > /usr/local/freeswitch/conf/directory/default?) due to incorrect (i.e., > out of date, misleading, confusing) Freeswitch documentation, comments, and > examples. *Not the least of which* was an existing Freeswitch example > file ? ?/usr/local/freeswitch/conf/directory/default/example.com.xml? ? > and multiple references to it. IMHO, this file should be deleted in the > next Freeswitch release. And all other XML includes/comments for the file > should be deleted. It all makes sense now, but as a new Freeswitch user, I > thought I was just following the recommended configuration guidelines. For > example, from Freeswitch configuration file ? > /usr/local/freeswitch/conf/vars.xml?: > > > > > > > > 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 < > GeorgePhelps at gfphelps.com> 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 > > _________________________________________________________________________ > 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/20150106/714c16e7/attachment-0001.html From brian at freeswitch.org Tue Jan 6 18:56:36 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2015 09:56:36 -0600 Subject: [Freeswitch-users] Control of IPv6 vs. IPv4 In-Reply-To: <079101d028e8$0a47b660$1ed72320$@gfphelps.com> References: <042501d026a5$74d7e670$5e87b350$@gfphelps.com> <046a01d026b7$a6d67e90$f4837bb0$@gfphelps.com> <079101d028e8$0a47b660$1ed72320$@gfphelps.com> Message-ID: And to clarify on my previous terse response... This is a community, If you encounter a problem such as this, you can submit updates to the configs via JIRA, or join the DOCS team to help improve the existing documentation and examples on Confluence. At the very least join the docs list and report your findings and someone there could take a look at the docs and take your input and fix it. On Mon, Jan 5, 2015 at 7:03 AM, George F. Phelps wrote: > RESOLVED > > > > The specific problem was, that I was defining my gateway in directory ? > /usr/local/freeswitch/conf/directory/default?. This was causing > Freeswitch to attempt registrations for both IPv4 and IPv6 ? which my VoIP > provider supports. A Freeswitch bug, IMHO, due to unexpected, > nondeterministic behavior. > > > > I was mistakenly doing this (i.e., defining my gateway in directory ? > /usr/local/freeswitch/conf/directory/default?) due to incorrect (i.e., > out of date, misleading, confusing) Freeswitch documentation, comments, and > examples. *Not the least of which* was an existing Freeswitch example > file ? ?/usr/local/freeswitch/conf/directory/default/example.com.xml? ? > and multiple references to it. IMHO, this file should be deleted in the > next Freeswitch release. And all other XML includes/comments for the file > should be deleted. It all makes sense now, but as a new Freeswitch user, I > thought I was just following the recommended configuration guidelines. For > example, from Freeswitch configuration file ? > /usr/local/freeswitch/conf/vars.xml?: > > > > > > > > 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 < > GeorgePhelps at gfphelps.com> 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 > > _________________________________________________________________________ > 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/20150106/d214cfef/attachment-0001.html From brian at freeswitch.org Tue Jan 6 19:36:50 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2015 10:36:50 -0600 Subject: [Freeswitch-users] Google Voice Support? In-Reply-To: References: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> Message-ID: The XMPP interface will die off... I wouldn't bet the farm on it. On Tue, Jan 6, 2015 at 9:20 AM, Daniel Ivanov wrote: > Oh (facepalm). > > On Tue, Jan 6, 2015 at 4:05 PM, Nikolay Rogoshchenkov > wrote: > >> Still use mod_dingaling >> for Google Voice as my free home phone, works fine. >> >> >> -- >> Best regards >> Rogoshchenkov Nikolay >> >> On Tue, Jan 6, 2015 at 6:41 AM, George F. Phelps < >> GeorgePhelps at gfphelps.com> wrote: >> >>> Now that Google has de-supported its XMPP connectivity to Google Voice, >>> what is the Freeswitch configuration for interfacing to Google Voice? >>> >>> >>> >>> 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 >>> >> >> >> _________________________________________________________________________ >> 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/20150106/967916aa/attachment.html From msc at freeswitch.org Tue Jan 6 19:42:33 2015 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Jan 2015 08:42:33 -0800 Subject: [Freeswitch-users] Call back through SIP trunk In-Reply-To: References: <0000014aa16b2a0b-ee3e05bc-9993-4da0-8da1-2edff71f346b-000000@email.amazonses.com> <0000014aa59db0b0-5b7762c1-dc93-4407-acf9-a05d6d8543ea-000000@email.amazonses.com> Message-ID: Hi Sina, Were you able to figure this one out? Also, I didn't see anywhere in this thread what the outbound leg would be connected to. For example, when you call out to the caller ID number received on the initial call, what will the other leg be connected to? Do you have a user or an IVR or what? Just curious. You can't make an outbound call without connecting that leg to something. -MC On Fri, Jan 2, 2015 at 4:22 PM, Sina Owolabi wrote: > Hi Avi > > I'm really sorry to bother you offlist again, but I am really, really > stumped. > This is my first time with lua or any kind of coding and I am > scrambling to make amends for that. > > Could you please help me with some kind of example script I can use for > this? > > On Thu, Jan 1, 2015 at 2:09 PM, Avi Marcus wrote: > > Once you hangup, unless you have zombie exec, the call ends and it won't > > transfer nor execute the lua script. > > > > Also, if you have the lua do the hangup, it can directly access all the > > channel variables itself. > > > > Alternatively, you can set a hangup hook and pass everything: > > > > > > -Avi > > > > On Thu, Jan 1, 2015 at 2:12 PM, Sina Owolabi > wrote: > >> > >> Avi, > >> > >> thanks and happy new year! > >> I'm trying to using a public extension to receive the call, hangup > >> and transfer to the extension that is actually doing the lua'ing (so > >> to speak): > >> > >> > >> >> expression="^0(\d{10})$"require-nested="false"> > >> >> data="effective_caller_id_number=+123${1}"/> > >> >> data="effective_caller_id_name=+123${1}"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> >> > >> Please is this the proper way to collect data from the first leg to > >> pass to the lua script? > >> > >> On Wed, Dec 31, 2014 at 6:36 PM, Avi Marcus wrote: > >> > I've used a lua script to grab the information (and make sure it's a > >> > valid > >> > callback), hangup, and then run: > >> > > >> > freeswitch.msleep(2000); --wait 2 seconds to make sure their side will > >> > actually have the call over > >> > api = freeswitch.API() > >> > api:execute("originate", your-dialstring) > >> > > >> > -Avi > >> > > >> > On Wed, Dec 31, 2014 at 11:37 AM, Sina Owolabi > > >> > wrote: > >> >> > >> >> Hi List! > >> >> > >> >> > >> >> FreeSWITCHNewbie here. > >> >> Please can I have some guidance on how to setup call back? > >> >> > >> >> I would like to be able to dial the DID attached to the SIP trunk > >> >> Freeswitch is registered to, and then have freeSWITCH hang up the > call > >> >> and dial the caller id number back through any other SIP trunk > >> >> FreeSWITCH Is registered with, but with the origination number set to > >> >> the DID that the first call came through in the first place. > >> >> > >> >> Please is this possible just through the dial plan? > >> >> > >> >> Thanks for any help! > >> >> > >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/eab1be82/attachment.html From aqsyounas at gmail.com Tue Jan 6 19:54:35 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 6 Jan 2015 21:54:35 +0500 Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout. In-Reply-To: References: <5495C388.2020902@quentustech.com> <5495E2C8.40604@quentustech.com> Message-ID: Thanks guys i am able to play a stream with mod_vlc. But when i hangup, i see my fs_cli flooding with these logs. Is it ok to have these in your logs.? EXECUTE sofia/internal/1010 at 192.168.1.28 playback(shout:// online.radiodifusion.net:8024/) 2015-01-06 21:49:21.841076 [DEBUG] mod_vlc.c:354 VLC waiting to close the files: 34132 2015-01-06 21:49:22.341221 [DEBUG] mod_vlc.c:354 VLC waiting to close the files: 34132 freeswitch at internal> freeswitch at internal> 2015-01-06 21:49:22.841088 [DEBUG] mod_vlc.c:354 VLC waiting to close the files: 50456 2015-01-06 21:49:23.341098 [DEBUG] mod_vlc.c:354 VLC waiting to close the files: 50456 2015-01-06 21:49:23.841108 [DEBUG] mod_vlc.c:354 VLC waiting to close the files: 66038 2015-01-06 21:49:24.341107 [DEBUG] mod_vlc.c:354 VLC waiting to close the files: 66038 2015-01-06 21:49:24.841066 [DEBUG] mod_vlc.c:354 VLC waiting to close the files: 83104 2015-01-06 21:49:25.341114 [DEBUG] mod_vlc.c:354 VLC waiting to close the files: 83104 2015-01-06 21:49:25.841237 [WARNING] mod_vlc.c:357 Giving up waiting for client to empty the audio buffer 2015-01-06 21:49:25.841237 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:26.341107 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:26.841098 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:27.341065 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:27.841100 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:28.341099 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:28.841099 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:29.341125 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:29.841215 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:30.341078 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:30.841089 [DEBUG] mod_vlc.c:365 VLC waiting for clients: 3 2015-01-06 21:49:31.341078 [WARNING] mod_vlc.c:368 Giving up waiting for client to get the last of the audio stream On 22 December 2014 at 21:22, Aqs Younas wrote: > Many thanks for you reply. I will try this and get back to you if i face > any problem. > > Really Appreciate your Help. > > On 21 December 2014 at 01:57, William King > wrote: > >> Comment out this line: >> https://github.com/videolan/vlc/blob/master/modules/access/http.c#L366 >> >> once you install the updated http access module it'll no longer hang. >> >> William King >> Senior Engineer >> Quentus Technologies, INC >> 1037 NE 65th St Suite 273 >> Seattle, WA 98115 >> Main: (877) 211-9337 >> Office: (206) 388-4772 >> Cell: (253) 686-5518 >> william.king at quentustech.com >> >> On 12/20/14 11:03 AM, Aqs Younas wrote: >> > I installed vlc 2.1.5 from source code using instructions provided in >> > wiki. https://wiki.freeswitch.org/wiki/Mod_vlc >> > I am currently using Debian 7.6. >> > >> > Thanks for you reply. >> > Would really appreciate your help. >> > >> > On 20 December 2014 at 23:44, William King < >> william.king at quentustech.com >> > > wrote: >> > >> > I wrote mod_vlc and have been working with the libvlc author who >> added >> > the patch that is causing the http streaming to hang. >> > >> > Which OS and version of vlc are you using? If it is a debian distro >> I >> > can provide a patch you can apply while rebuilding libvlc. >> > >> > William King >> > Senior Engineer >> > Quentus Technologies, INC >> > 1037 NE 65th St Suite 273 >> > Seattle, WA 98115 >> > Main: (877) 211-9337 >> > Office: (206) 388-4772 >> > Cell: (253) 686-5518 >> > william.king at quentustech.com >> > >> > On 12/20/14 9:50 AM, Aqs Younas wrote: >> > > Thanks for your reply and time. >> > > >> > > Could you tell me how long its gonna take or when this feature >> will be >> > > available.? >> > > >> > > Many thanks for your reply. >> > > >> > > On 20 December 2014 at 22:45, Anthony Minessale >> > > >> > > > >> wrote: >> > > >> > > I think there is an issue playing urls with vlc because vlc >> has some >> > > code in it that uses signal handlers which are blocked by FS. >> > > The developer who made this function in vlc is not very >> receptive to >> > > a solution so its being worked on. >> > > >> > > >> > > >> > > On Sat, Dec 20, 2014 at 6:02 AM, Aqs Younas < >> aqsyounas at gmail.com >> > > >> >> wrote: >> > > >> > > Hi, Seven >> > > >> > > When i try to play a stream it doesn't give any audio. >> Also >> > > session created when i try to play a stream with mod_vlc >> does >> > > not terminate even with "hupall". >> > > >> > > [0x7f69b00025d8] main input debug: Creating an input for >> > > 'http://s9.voscast.com:7584' >> > > [0x7f69b00025d8] main input debug: using timeshift >> granularity >> > > of 50 MiB, in path '/tmp' >> > > [0x7f69b00025d8] main input debug: ` >> http://s9.voscast.com:7584' >> > > gives access `http' demux `' path `s9.voscast.com:7584 < >> http://s9.voscast.com:7584> >> > > ' >> > > [0x7f69b00025d8] main input debug: creating demux: >> access='http' >> > > demux='' location='s9.voscast.com:7584 < >> http://s9.voscast.com:7584> >> > > ' file='(null)' >> > > [0x26cbd48] main demux debug: looking for access_demux >> module >> > > matching "http": 11 candidates >> > > [0x26cbd48] main demux debug: no access_demux modules >> matched >> > > [0x7f69b00025d8] main input debug: creating access 'http' >> > > location='s9.voscast.com:7584 > > >> > ' >> > > , path='(null)' >> > > [0x26cbc68] main access debug: looking for access module >> > > matching "http": 19 candidates >> > > [0x26cbc68] access_http access debug: querying proxy for >> > > http://s9.voscast.com:7584 >> > > >> > > This is what i see in my logs. Can you provide me with an >> > > example or guide me that i am doing wrong? >> > > >> > > Thanks >> > > >> > > >> > > On 20 December 2014 at 06:15, Seven Du < >> dujinfang at gmail.com >> > > >> >> wrote: >> > > >> > > mod_vlc is the best answer to the original question. >> > > >> > > On Wednesday, December 17, 2014 at 10:26 PM, Brian >> West wrote: >> > > >> > >> Don't think mod_rtmp can actually do this, from >> loading it: >> > >> >> > >> 2014-12-17 08:25:39.983446 [NOTICE] >> > >> switch_loadable_module.c:149 Adding Endpoint 'rtmp' >> > >> >> > >> 2014-12-17 08:25:39.983446 [NOTICE] >> > >> switch_loadable_module.c:315 Adding API Function >> 'rtmp' >> > >> >> > >> 2014-12-17 08:25:39.983446 [NOTICE] >> > >> switch_loadable_module.c:315 Adding API Function >> > >> 'rtmp_contact' >> > >> >> > >> >> > >> This is all that gets registered. >> > >> >> > >> >> > >> On Sun, Dec 14, 2014 at 1:37 PM, Danny Gershman >> > >> > danny.gershman at gmail.com> >> > >> > danny.gershman at gmail.com>>> wrote: >> > >>> Also mod_rtmp lets you play from an FMS server. >> > >>> >> > >>> >> > >>> On Friday, December 12, 2014, Aqs Younas >> > >>> >> > >> wrote: >> > >>>> Hi, All >> > >>>> >> > >>>> How can i play a live stream other than mp3 with >> > >>>> mod_shout or any module.? Is there any way to >> buffer the >> > >>>> stream before playing it with mod_shout. >> > >>>> >> > >>>> >> > >>>> Currently i have a list of streams and when i play >> them >> > >>>> with mod_shout some work fine but others give >> (time out) >> > >>>> error. >> > >>>> >> > >>>> How can i play mostly stream in freeswitch? >> > >>>> >> > >>>> Thanks >> > >>> >> > >>> >> _________________________________________________________________________ >> > >>> Professional FreeSWITCH Consulting Services: >> > >>> consulting at freeswitch.org >> > > 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 >> > > 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 >> > > 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 >> > > 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 >> > > >> > > >> > > >> > > >> > > -- >> > > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> > > >> > > ? http://freeswitch.org/ ? http://cluecon.com/ ? >> > > http://twitter.com/FreeSWITCH >> > > ? irc.freenode.net >> > #freeswitch ? >> > > _http://freeswitch.org/g+_ >> > > >> > > ClueCon Weekly Development Call >> > > ? sip:888 at conference.freeswitch.org >> > >> > > > > > ? +19193869900 >> > >> > > >> > > >> > > >> > > >> _________________________________________________________________________ >> > > 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 >> > > >> > >> > >> _________________________________________________________________________ >> > 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 >> > >> >> _________________________________________________________________________ >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/32937762/attachment-0001.html From msc at freeswitch.org Tue Jan 6 20:09:22 2015 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Jan 2015 09:09:22 -0800 Subject: [Freeswitch-users] Canceling sched_hangup and UUID lookup In-Reply-To: References: Message-ID: For posterity's sake I thought it would be good to drop a few tidbits here. #1 - You *can* delete a scheduled hangup. You just need to know the task number. When you issue the sched_hangup you'll see the task id printed on the console. You can also get a list of the scheduled tasks by issuing the "show tasks" API. Once you know the task id you can simply do "sched_del ". Piece of cake, no? #2 - If you aren't using the presence_data field for anything else, which in your case you probably are not, you can set info there. Just do a set or export prior to the bridge or add "{presence_data=foo}" to the beginning of the dialstring. From there you can just do "show channels like foo" and you'll get only those channels. Just be sure to have your presence_data value be very specific so that you don't accidentally grab unwanted channels. Hope this helps. -MC On Fri, Jan 2, 2015 at 3:00 PM, Kurtis Heimerl wrote: > I think I understand how it could be done through ESL, but there's no > mechanism to look up sessions? That seems really weird. > > On Fri, Jan 2, 2015 at 2:48 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> Well, if you don't have a lot of traffic, you can just use nibblebill as >> suggested. You can just use account balances * 100 if you can't use floats >> or decimals. >> >> Else, you need to use ESL and keep track of all calls, store states >> somewhere like a db, so that when you need to change timeouts you know >> which ones to change. >> >> But again, if you don't have too much traffic, you might look into >> nibblebill, I used it some time ago and it works ok with small traffic. >> On Jan 2, 2015 5:42 PM, "Kurtis Heimerl" >> wrote: >> >>> Here's the use case. >>> >>> A new call comes in. I have to find all calls from that user account and >>> reset their timeouts (as a new call will change the total billable time). >>> So I need a mechanism from the dialplan that can search for all calls (and >>> their uuids) with a certain variable set. Does that make sense? >>> >>> On Fri, Jan 2, 2015 at 2:37 PM, David Villasmil Govea < >>> david.villasmil at gmail.com> wrote: >>> >>>> I'm not sure how you intent to use it, but if you set a variable, when >>>> cdrs are saved, the variables are saved just like any other channel >>>> variable, of course along uuids, etc. >>>> On Jan 2, 2015 5:34 PM, "Kurtis Heimerl" >>>> wrote: >>>> >>>>> >>>>> >>>>> On Fri, Jan 2, 2015 at 2:30 PM, David Villasmil Govea < >>>>> david.villasmil at gmail.com> wrote: >>>>> >>>>>> I don't think you can cancel a sched_hangup, but you can afaik >>>>>> re-issue the command with a new time. >>>>>> >>>>> >>>>> If that'll erase the old one, that's totally sufficient. Great! >>>>> >>>>> >>>>>> Regarding the tag, if you need it as a cdr field, you just need to >>>>>> set a custom variable, they get saved automatically when saving the cdr. >>>>>> >>>>> >>>>> Is there a way to look up a uuid via these variables? >>>>> >>>>> >>>>>> On Jan 2, 2015 5:17 PM, "Kurtis Heimerl" >>>>>> wrote: >>>>>> >>>>>>> No! I hadn't seen it. My brief reading says it may be able to fit >>>>>>> our needs. There are some issues though (for instance, it's commonly known >>>>>>> to never use a float/double for currencies and we do bill per second). So >>>>>>> I'll follow up with that and see if it's a solution. >>>>>>> >>>>>>> Barring that, is there any other way to do the two things I'm >>>>>>> looking for? >>>>>>> >>>>>>> On Fri, Jan 2, 2015 at 1:57 PM, Brian West >>>>>>> wrote: >>>>>>> >>>>>>>> This is why mod_nibblebill exists, have you looked at it? >>>>>>>> >>>>>>>> On Fri, Jan 2, 2015 at 3:53 PM, Kurtis Heimerl < >>>>>>>> kheimerl at cs.berkeley.edu> wrote: >>>>>>>> >>>>>>>>> Hello Freeswitch-users, >>>>>>>>> >>>>>>>>> I'm building a group billing solution for our freeswitch system. >>>>>>>>> Our account management is external, so I need to make something custom. I >>>>>>>>> have two questions about how to do some stuff in FS. >>>>>>>>> >>>>>>>>> 1) I know how to limit call length (sched_hangup). I can query an >>>>>>>>> external account via REST to see how many credits they have and limit the >>>>>>>>> call with that. However, when a second user on the same account calls in, I >>>>>>>>> need to change that timeout. Is there a way to cancel a sched_hangup? I >>>>>>>>> found sched_cancel, but that seems to be for broadcasts and transfers, >>>>>>>>> neither of which I'm doing. I *guess* I could transfer to a hangup >>>>>>>>> extension instead of hanging up... but that seems awful ugly. >>>>>>>>> >>>>>>>>> 2) Is there a way to "tag" a call (uuid) with an arbitrary text >>>>>>>>> string and then look it up by that? That would allow me to tag each call >>>>>>>>> with the account name and later get all calls under that account. >>>>>>>>> >>>>>>>>> Thanks! >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> 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 >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/307c44f4/attachment-0001.html From krice at freeswitch.org Tue Jan 6 20:10:32 2015 From: krice at freeswitch.org (Ken Rice) Date: Tue, 06 Jan 2015 17:10:32 +0000 Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) December 28th-January 4th Message-ID: <54ac1708aff51_34c1551338327c@ip-10-33-162-187.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/17g3uGI FreeSWITCH Week in Review (Master Branch) December 28th-January 4th Hello, again. This week in the FreeSWITCH master branch we had 10 commits. The features for this week are improvements to socket handling and the implementation of grunt for javascript management in mod_verto. Grunt is a task runner for javascript designed to automate small mundane tasks like minification, compilation, unit testing, and linting. New features that were added: bf5210b Improved mod_verto socket handling code 0db3b1e FS-7127 Add grunt to mod_verto [Jira: http://ift.tt/17g3xlC] Improvements to documentation: 7a517ee FS-7127 Update README and sync re-generate verto-min [Jira: http://ift.tt/17g3xlC] f0ec193 FS-7127 Update the README for mod_verto [Jira: http://ift.tt/17g3xlC] The following bugs were squashed: 65631ed Revert FS-7004 pending updated fixed for that issue from the original author [Jira: http://ift.tt/1pgbVcN] a067a49 FS-7046 fix warning introduced from b341ff7 properly [Jira: http://ift.tt/1yJc4rg] a961b0e Fix return size in mod_verto 1965b3b FS-7106 #resolve Fix concurrency issue in mod_httapi [Jira: http://ift.tt/1BEamXd] 0bec209 Fix fsapi in verto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/34a410f2/attachment.html From msc at freeswitch.org Tue Jan 6 20:23:48 2015 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Jan 2015 09:23:48 -0800 Subject: [Freeswitch-users] Detecting multiple tones with freeswitch In-Reply-To: <54AAF4D9.2030904@gmail.com> References: <54A7E4A8.6090702@gmail.com> <012A2476-B5CC-4F5A-BDF9-E13AE726319E@jerris.com> <54AAF4D9.2030904@gmail.com> Message-ID: I'm with Jerris on the reliability issue, but if you can't switch carriers and they won't budge on sending the audio inline then you're pretty much stuck with what you've got. I do know that the tone_detect app works well. It's been years since I've done anything with it so I'm a bit rusty. In your case what I would do is use the execute_on_tone_detect to issue an info app. I think you might have a value in there somewhere that will help you differentiate between whether busy or ring was detected. I wish you well. This is not a fun endeavor... -MC On Mon, Jan 5, 2015 at 12:32 PM, Bunea Lucian wrote: > I have a dedicated short number and multiple DIDs. I also have a > database with customer phone numbers and their corresponding DIDs. > Each field agent has a mobile phone (from the same provider). > When a call comes in freeswitch automatically routes the call to the > designated field agent via the corresponding DID. > I want to monitor what happens with the call after that, to be able to > decide what to do with the call next. > So basically, the call never leaves the provider's network. > > Btw, I did asked the provider to switch back the settings but the feedback > was: > "The behavior is normal because for calls made inside Vodafone network, we > generate tones. Thus, for Busy, we play specific tone and then disconnect > the call, signaling with SIP 480 (corresponding to the end of > announcement). If a call comes from another network, User Busy is signaled > on ISUP and SIP returned as such. The settings can not be changed per > customer. " > Unofficially, they acknowledged that their PBX behaved differently before > because it was not correctly configured. > > PS: My provider sends the description of the tones using a reason header > inside the SIP/183 message. But I was unable to find a way to extract it: > > > http://lists.freeswitch.org/pipermail/freeswitch-users/2014-March/103645.html > > > > -------- Original Message -------- > *Subject: *Re: [Freeswitch-users] Detecting multiple tones with freeswitch > *From: *Michael Jerris > *To: *FreeSWITCH Users Help > > *Date: *05.01.2015 20:02 > > You can never reliably depend on the tones, particularly with international traffic. This is never going to work reliably unless you get useful sip signaling. I would be demanding the carrier change settings back. > > > > On Jan 3, 2015, at 7:46 AM, Bunea Lucian wrote: > > Hello, > > My SIP provider (Vodafone) decided to make me a Christmas present: they > change the configuration of their SIP server. > > Using the old configuration they were sending SIP codes for decline, > busy and unavailable. > Now they are sending early media with tones (for decline and busy) and a > one minute message for unavailable. > Each status is followed by a SIP/480. > > Since I need to be able to distinguish between different statuses, I > have determined the following: > - if I don't receive a ring tone within 7 second, the called number is > unavailable; > - if I receive a busy tone within 4 seconds the called number is busy; > - if I receive a busy tone after 4 second the call number has declined > the call; > > This is how I tried to implement it: > - after receiving early media (SIP/180 or SIP/183) a timer is started > for call hangup (sched_hangup) > - if I receive a ring tone within 7 seconds the timer is canceled > (sched_cancel); > - if I receive a busy tone a lua script is called; > > > > > > > > > data="sofia/external/${destination_number}@XXX.XXX.XXX.XXX" /> > > But, for some reason, the tones are detected, but the applications are > never called. > > > If I add one of the following lines to the dialplan the corresponding > application is called: > > > > or > > > > but I can't tell which tone was detected... > > What am I missing? > > PS: Tested with FreeSWITCH Version 1.4.14-1~64bit. > > Regards, > Lucian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/d096f1d3/attachment.html From msc at freeswitch.org Tue Jan 6 20:31:02 2015 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Jan 2015 09:31:02 -0800 Subject: [Freeswitch-users] lua custom variables In-Reply-To: References: Message-ID: Without seeing logs and all that it's hard to know exactly what's going on, however could you try doing an export as opposed to a set and see if your custom variable shows up where you want? -MC On Sat, Jan 3, 2015 at 10:12 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Update: > > I execute 2 scripts. > > I set an execute_on_answer lua script (lua2), and execute another (lua1). > like so: > > > > > > > > > > > > > > Now, the lua1 executes, i set a variable "custom_dur" which is the > duration for the call. > When lua3 executes on answer, the variable custom_duration is still there, > no problems. Then I transfer the call to "9999", which is: > > > > > > > > > Now, on lua3 I try to get the variable and it's not there anymore! > > I also tried the following: On lua1 I create a new variable, which is seen > on ALL scripts, including lua3, but I reset it to a new value on lua2, and > on lua3 I don't see the change!! The variable is still the value I set on > lua1!!! wth?? > > Am I missing something on variables?? > > Thanks a lot, > > David > > > > On Sun, Jan 4, 2015 at 3:30 AM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> Hello Guys, >> >> I have this lus script, which sets a custom variable, this is ok. >> The the call is transferred to another extension, and the variable is not >> there anymore, is this by design? >> >> Thanks! >> >> David >> >> -- >> DVG >> >> -- >> Imagination is more important than knowledge >> Albert Einstein >> > > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/47eca9d2/attachment-0001.html From david.villasmil at gmail.com Tue Jan 6 21:07:03 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 6 Jan 2015 19:07:03 +0100 Subject: [Freeswitch-users] lua custom variables In-Reply-To: References: Message-ID: Is it possible to prevent this? On Jan 5, 2015 7:26 PM, "Michael Jerris" wrote: > sounds like the second time its executing on a different session > > On Jan 4, 2015, at 9:01 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Hello guys, > > Any help with this? > > Thanks > > David > > On Sun, Jan 4, 2015 at 7:12 AM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> Update: >> >> I execute 2 scripts. >> >> I set an execute_on_answer lua script (lua2), and execute another (lua1). >> like so: >> >> >> >> >> >> >> >> >> >> >> >> >> >> Now, the lua1 executes, i set a variable "custom_dur" which is the >> duration for the call. >> When lua3 executes on answer, the variable custom_duration is still >> there, no problems. Then I transfer the call to "9999", which is: >> >> >> >> >> >> >> >> >> Now, on lua3 I try to get the variable and it's not there anymore! >> >> I also tried the following: On lua1 I create a new variable, which is >> seen on ALL scripts, including lua3, but I reset it to a new value on lua2, >> and on lua3 I don't see the change!! The variable is still the value I set >> on lua1!!! wth?? >> >> Am I missing something on variables?? >> On Sun, Jan 4, 2015 at 3:30 AM, David Villasmil Govea < >> david.villasmil at gmail.com> wrote: >> >>> Hello Guys, >>> >>> I have this lus script, which sets a custom variable, this is ok. >>> The the call is transferred to another extension, and the variable is >>> not there anymore, is this by design? >>> >> > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/cb803d75/attachment.html From lucibunea at gmail.com Tue Jan 6 21:15:11 2015 From: lucibunea at gmail.com (Bunea Lucian) Date: Tue, 06 Jan 2015 20:15:11 +0200 Subject: [Freeswitch-users] Detecting multiple tones with freeswitch In-Reply-To: References: <54A7E4A8.6090702@gmail.com> <012A2476-B5CC-4F5A-BDF9-E13AE726319E@jerris.com> <54AAF4D9.2030904@gmail.com> Message-ID: <54AC262F.1000309@gmail.com> I did managed to solve my problem by using tone_detect to handle "busy"/"decline" and using an event hook script that listens for "CHANNEL_CALLSTATE" events (for "unavailable"). So, if: Channel-Call-State = RINGING, Call-Direction = outbound, Answer-State = early, Original-Channel-Call-State = EARLY, Caller-ANI = "one_of_my_DIDs" and (Event-Date-Timestamp - Caller-Channel-Created-Time) is lower than 7 seconds then I execute a sched_cancel on the session ( to delete the sched_hangup). -------- Original Message -------- *Subject: *Re: [Freeswitch-users] Detecting multiple tones with freeswitch *From: *Michael Collins *To: *FreeSWITCH Users Help *Date: *06.01.2015 19:23 > I'm with Jerris on the reliability issue, but if you can't switch > carriers and they won't budge on sending the audio inline then you're > pretty much stuck with what you've got. I do know that the tone_detect > app works well. It's been years since I've done anything with it so > I'm a bit rusty. In your case what I would do is use the > execute_on_tone_detect to issue an info app. I think you might have a > value in there somewhere that will help you differentiate between > whether busy or ring was detected. > > I wish you well. This is not a fun endeavor... > > -MC > > > On Mon, Jan 5, 2015 at 12:32 PM, Bunea Lucian > wrote: > > I have a dedicated short number and multiple DIDs. I also have a > database with customer phone numbers and their corresponding DIDs. > Each field agent has a mobile phone (from the same provider). > When a call comes in freeswitch automatically routes the call to > the designated field agent via the corresponding DID. > I want to monitor what happens with the call after that, to be > able to decide what to do with the call next. > So basically, the call never leaves the provider's network. > > Btw, I did asked the provider to switch back the settings but the > feedback was: > "The behavior is normal because for calls made inside Vodafone > network, we generate tones. Thus, for Busy, we play specific tone > and then disconnect the call, signaling with SIP 480 > (corresponding to the end of announcement). If a call comes from > another network, User Busy is signaled on ISUP and SIP returned as > such. The settings can not be changed per customer. " > Unofficially, they acknowledged that their PBX behaved differently > before because it was not correctly configured. > > PS: My provider sends the description of the tones using a reason > header inside the SIP/183 message. But I was unable to find a way > to extract it: > > http://lists.freeswitch.org/pipermail/freeswitch-users/2014-March/103645.html > > > > > -------- Original Message -------- > *Subject: *Re: [Freeswitch-users] Detecting multiple tones with > freeswitch > *From: *Michael Jerris > *To: *FreeSWITCH Users Help > > > *Date: *05.01.2015 20:02 >> You can never reliably depend on the tones, particularly with international traffic. This is never going to work reliably unless you get useful sip signaling. I would be demanding the carrier change settings back. >> >> >>> On Jan 3, 2015, at 7:46 AM, Bunea Lucian wrote: >>> >>> Hello, >>> >>> My SIP provider (Vodafone) decided to make me a Christmas present: they >>> change the configuration of their SIP server. >>> >>> Using the old configuration they were sending SIP codes for decline, >>> busy and unavailable. >>> Now they are sending early media with tones (for decline and busy) and a >>> one minute message for unavailable. >>> Each status is followed by a SIP/480. >>> >>> Since I need to be able to distinguish between different statuses, I >>> have determined the following: >>> - if I don't receive a ring tone within 7 second, the called number is >>> unavailable; >>> - if I receive a busy tone within 4 seconds the called number is busy; >>> - if I receive a busy tone after 4 second the call number has declined >>> the call; >>> >>> This is how I tried to implement it: >>> - after receiving early media (SIP/180 or SIP/183) a timer is started >>> for call hangup (sched_hangup) >>> - if I receive a ring tone within 7 seconds the timer is canceled >>> (sched_cancel); >>> - if I receive a busy tone a lua script is called; >>> >>> >>> >>> >>> >>> >>> >>> >>> >> data="sofia/external/${destination_number}@XXX.XXX.XXX.XXX" /> >>> >>> But, for some reason, the tones are detected, but the applications are >>> never called. >>> >>> >>> If I add one of the following lines to the dialplan the corresponding >>> application is called: >>> >>> >>> >>> or >>> >>> >>> >>> but I can't tell which tone was detected... >>> >>> What am I missing? >>> >>> PS: Tested with FreeSWITCH Version 1.4.14-1~64bit. >>> >>> Regards, >>> Lucian >> _________________________________________________________________________ >> 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 > > > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/3e0be840/attachment-0001.html From sertys at gmail.com Tue Jan 6 21:22:35 2015 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 6 Jan 2015 19:22:35 +0100 Subject: [Freeswitch-users] Google Voice Support? In-Reply-To: References: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> Message-ID: But dingaling is not xmpp based, right? 6 ???. 2015 ?. 18:38 ???????????? "Brian West" ???????: > The XMPP interface will die off... I wouldn't bet the farm on it. > > On Tue, Jan 6, 2015 at 9:20 AM, Daniel Ivanov wrote: > >> Oh (facepalm). >> >> On Tue, Jan 6, 2015 at 4:05 PM, Nikolay Rogoshchenkov < >> nickolayr at gmail.com> wrote: >> >>> Still use mod_dingaling >>> for Google Voice as my free home phone, works fine. >>> >>> >>> -- >>> Best regards >>> Rogoshchenkov Nikolay >>> >>> On Tue, Jan 6, 2015 at 6:41 AM, George F. Phelps < >>> GeorgePhelps at gfphelps.com> wrote: >>> >>>> Now that Google has de-supported its XMPP connectivity to Google Voice, >>>> what is the Freeswitch configuration for interfacing to Google Voice? >>>> >>>> >>>> >>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/b2ef09e5/attachment.html From steveayre at gmail.com Tue Jan 6 21:37:35 2015 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 6 Jan 2015 18:37:35 +0000 Subject: [Freeswitch-users] Control of IPv6 vs. IPv4 In-Reply-To: <079101d028e8$0a47b660$1ed72320$@gfphelps.com> References: <042501d026a5$74d7e670$5e87b350$@gfphelps.com> <046a01d026b7$a6d67e90$f4837bb0$@gfphelps.com> <079101d028e8$0a47b660$1ed72320$@gfphelps.com> Message-ID: "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?" Not necessarily. Firstly those configuration files are meant as an example. You're probably referring to the vanilla example. It's entirely possible to have the config files laid out in another manner, even to the point of a single flat (but very large) file. You can have as many or few profiles are you require, and 'external' is just a name and doesn't mean anything special. You could have your external profile listening on IPv6 and then those gateways would be IPv6. The key point is that the IP that the gateways use will depend upon the settings of the profile the gateways are defined upon. On 5 January 2015 at 13:03, George F. Phelps wrote: > RESOLVED > > > > The specific problem was, that I was defining my gateway in directory ? > /usr/local/freeswitch/conf/directory/default?. This was causing > Freeswitch to attempt registrations for both IPv4 and IPv6 ? which my VoIP > provider supports. A Freeswitch bug, IMHO, due to unexpected, > nondeterministic behavior. > > > > I was mistakenly doing this (i.e., defining my gateway in directory ? > /usr/local/freeswitch/conf/directory/default?) due to incorrect (i.e., > out of date, misleading, confusing) Freeswitch documentation, comments, and > examples. *Not the least of which* was an existing Freeswitch example > file ? ?/usr/local/freeswitch/conf/directory/default/example.com.xml? ? > and multiple references to it. IMHO, this file should be deleted in the > next Freeswitch release. And all other XML includes/comments for the file > should be deleted. It all makes sense now, but as a new Freeswitch user, I > thought I was just following the recommended configuration guidelines. For > example, from Freeswitch configuration file ? > /usr/local/freeswitch/conf/vars.xml?: > > > > > > > > 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 < > GeorgePhelps at gfphelps.com> 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/840ab8b0/attachment-0001.html From cmrienzo at gmail.com Tue Jan 6 22:26:46 2015 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 6 Jan 2015 14:26:46 -0500 Subject: [Freeswitch-users] Google Voice Support? In-Reply-To: References: <08ff01d029a5$b4b72af0$1e2580d0$@gfphelps.com> Message-ID: mod_dingaling and the jingle protocol are XMPP based On Tue, Jan 6, 2015 at 1:22 PM, Daniel Ivanov wrote: > But dingaling is not xmpp based, right? > 6 ???. 2015 ?. 18:38 ???????????? "Brian West" > ???????: > > The XMPP interface will die off... I wouldn't bet the farm on it. >> >> On Tue, Jan 6, 2015 at 9:20 AM, Daniel Ivanov wrote: >> >>> Oh (facepalm). >>> >>> On Tue, Jan 6, 2015 at 4:05 PM, Nikolay Rogoshchenkov < >>> nickolayr at gmail.com> wrote: >>> >>>> Still use mod_dingaling >>>> for Google Voice as >>>> my free home phone, works fine. >>>> >>>> >>>> -- >>>> Best regards >>>> Rogoshchenkov Nikolay >>>> >>>> On Tue, Jan 6, 2015 at 6:41 AM, George F. Phelps < >>>> GeorgePhelps at gfphelps.com> wrote: >>>> >>>>> Now that Google has de-supported its XMPP connectivity to Google >>>>> Voice, what is the Freeswitch configuration for interfacing to Google Voice? >>>>> >>>>> >>>>> >>>>> 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 >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/54afb720/attachment.html From mike at jerris.com Tue Jan 6 22:37:53 2015 From: mike at jerris.com (Michael Jerris) Date: Tue, 6 Jan 2015 14:37:53 -0500 Subject: [Freeswitch-users] lua custom variables In-Reply-To: References: Message-ID: <923983FF-B2B9-4B18-829E-EDD8C9422B2A@jerris.com> Prevent different sessions being different? > On Jan 6, 2015, at 1:07 PM, David Villasmil Govea wrote: > > Is it possible to prevent this? > > On Jan 5, 2015 7:26 PM, "Michael Jerris" > wrote: > sounds like the second time its executing on a different session > >> On Jan 4, 2015, at 9:01 PM, David Villasmil Govea > wrote: >> >> Hello guys, >> >> Any help with this? >> >> Thanks >> >> David >> >> On Sun, Jan 4, 2015 at 7:12 AM, David Villasmil Govea > wrote: >> Update: >> >> I execute 2 scripts. >> >> I set an execute_on_answer lua script (lua2), and execute another (lua1). >> like so: >> >> >> >> >> >> >> >> >> >> >> >> >> >> Now, the lua1 executes, i set a variable "custom_dur" which is the duration for the call. >> When lua3 executes on answer, the variable custom_duration is still there, no problems. Then I transfer the call to "9999", which is: >> >> >> >> >> >> >> >> >> Now, on lua3 I try to get the variable and it's not there anymore! >> >> I also tried the following: On lua1 I create a new variable, which is seen on ALL scripts, including lua3, but I reset it to a new value on lua2, and on lua3 I don't see the change!! The variable is still the value I set on lua1!!! wth?? >> >> Am I missing something on variables?? >> On Sun, Jan 4, 2015 at 3:30 AM, David Villasmil Govea > wrote: >> Hello Guys, >> >> I have this lus script, which sets a custom variable, this is ok. >> The the call is transferred to another extension, and the variable is not there anymore, is this by design? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/28822a5b/attachment-0001.html From david.villasmil at gmail.com Tue Jan 6 22:40:37 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 6 Jan 2015 20:40:37 +0100 Subject: [Freeswitch-users] lua custom variables In-Reply-To: References: Message-ID: Yes, why are sessions different? Because they are different scripts?? Do I need to keep everything in one script? On Jan 5, 2015 7:26 PM, "Michael Jerris" wrote: > sounds like the second time its executing on a different session > > On Jan 4, 2015, at 9:01 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Hello guys, > > Any help with this? > > Thanks > > David > > On Sun, Jan 4, 2015 at 7:12 AM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> Update: >> >> I execute 2 scripts. >> >> I set an execute_on_answer lua script (lua2), and execute another (lua1). >> like so: >> >> >> >> >> >> >> >> >> >> >> >> >> >> Now, the lua1 executes, i set a variable "custom_dur" which is the >> duration for the call. >> When lua3 executes on answer, the variable custom_duration is still >> there, no problems. Then I transfer the call to "9999", which is: >> >> >> >> >> >> >> >> >> Now, on lua3 I try to get the variable and it's not there anymore! >> >> I also tried the following: On lua1 I create a new variable, which is >> seen on ALL scripts, including lua3, but I reset it to a new value on lua2, >> and on lua3 I don't see the change!! The variable is still the value I set >> on lua1!!! wth?? >> >> Am I missing something on variables?? >> On Sun, Jan 4, 2015 at 3:30 AM, David Villasmil Govea < >> david.villasmil at gmail.com> wrote: >> >>> Hello Guys, >>> >>> I have this lus script, which sets a custom variable, this is ok. >>> The the call is transferred to another extension, and the variable is >>> not there anymore, is this by design? >>> >> > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/f450f61d/attachment.html From mario_fs at mgtech.com Tue Jan 6 23:02:56 2015 From: mario_fs at mgtech.com (Mario G) Date: Tue, 6 Jan 2015 12:02:56 -0800 Subject: [Freeswitch-users] IPV6 phone question about dial_string Message-ID: <37F1EEB4-A343-4EF8-8A6D-7C0064BE4729@mgtech.com> I got calls to go to an IPV6 phone by copying this line from default.xml: to the user extension.xml and changed it to: (this works fine) ;tag=FQag8jSNpac9c To: Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: . . . I am already setting the following values (in my gateway definition): Thanks, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/4c755827/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/20150106/4c755827/attachment-0001.bin From blefko5361 at gmail.com Wed Jan 7 02:34:09 2015 From: blefko5361 at gmail.com (Bruce Lefko) Date: Tue, 6 Jan 2015 17:34:09 -0600 Subject: [Freeswitch-users] getting call duration with event socket Message-ID: Hi there. I'm trying to get the call duration in seconds using ESL, but it looks like the channel variables are not being set correctly. In CHANNEL_HANGUP events, I see the following variables: 'Caller-Profile-Created-Time' => '1420586926995761', 'Caller-Channel-Created-Time' => '1420586926995761', 'Caller-Channel-Answered-Time' => '1420586929503733', 'Caller-Channel-Progress-Time' => '0', 'Caller-Channel-Progress-Media-Time' => '1420586929203735', 'Caller-Channel-Hangup-Time' => '0', I would expect to generate the duration of the call by subtracting Caller-Channel-Answered-Time from Caller-Channel-Hangup-Time, but the hangup time is 0. The actual call in this case was 45 seconds long. What can I do use Freeswitch events to compute call duration? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/b7a2cba0/attachment.html From notify.sina at gmail.com Wed Jan 7 02:52:33 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Wed, 7 Jan 2015 00:52:33 +0100 Subject: [Freeswitch-users] Help Needed Debugging Lua Script Message-ID: Hi List, FreeSWITCH newbie here again. I am trying to cobble togther a lua callback script, my first attempt was successful, but I am trying to make it slightly more elegant. I don't see any errors when I try to run this but the callback isnt happening. This is my very second attempt trying to write in lua, so I would be very grateful for any help. The user is expected to dial in, have the call hangup and FreeSWITCH call back. I'm passing a modified $effective_caller_id_number and $destination_number to the lua script: The script itself: api = freeswitch.API(); call_string = "bagpi originate {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" freeswitch.msleep(5000); if (session:ready()) then caller_id_number = session:getVariable("destination_number"); caller_id_name = session:getVariable("destination_number"); number_to_call = session:getVariable("effective_caller_id_number"); api:executeString(call_string); freeswitch.msleep(2000); session:streamFile("/tmp/get_off_my_lawn.wav"); session:hangup("NORMAL_CLEARING"); end From brian at freeswitch.org Wed Jan 7 03:38:10 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2015 18:38:10 -0600 Subject: [Freeswitch-users] Register->From->Caller ID In-Reply-To: <0aa001d02a04$620588b0$26109a10$@gfphelps.com> References: <0aa001d02a04$620588b0$26109a10$@gfphelps.com> Message-ID: gateway->register_from = switch_core_sprintf(gateway->pool, "", from_user, !zstr(from_domain) ? from_domain : proxy); This is how the from is built in the code. Do you have a use case that Requires it? I can't think of a reason it would matter. On Tue, Jan 6, 2015 at 4:59 PM, George F. Phelps wrote: > My current gateway REGISTER packet has a certain FROM header, as shown > below: > > > > REGISTER sip:66.33.147.150;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp > > Max-Forwards: 70 > > From: ;tag=FQag8jSNpac9c > > To: > > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce > > CSeq: 69955649 REGISTER > > Contact: switch2voip.us> > > . . . > > > > What do I set/change in my gateway configuration to add a character string > ?1404XXXXXXX? to the FROM header? As shown in red below: > > > > REGISTER sip:66.33.147.150;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp > > Max-Forwards: 70 > > From: ?1404XXXXXXX? ;tag=FQag8jSNpac9c > > To: > > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce > > CSeq: 69955649 REGISTER > > Contact: switch2voip.us> > > . . . > > > > I am already setting the following values (in my gateway definition): > > > > > > > > > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/720561f4/attachment.html From brian at freeswitch.org Wed Jan 7 03:42:14 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2015 18:42:14 -0600 Subject: [Freeswitch-users] getting call duration with event socket In-Reply-To: References: Message-ID: Use the CHANNEL_HANGUP_COMPLETE event for this purpose. On Tue, Jan 6, 2015 at 5:34 PM, Bruce Lefko wrote: > Hi there. > > I'm trying to get the call duration in seconds using ESL, but it looks > like the channel variables are not being set correctly. > > In CHANNEL_HANGUP events, I see the following variables: > > 'Caller-Profile-Created-Time' => '1420586926995761', > 'Caller-Channel-Created-Time' => '1420586926995761', > 'Caller-Channel-Answered-Time' => '1420586929503733', > 'Caller-Channel-Progress-Time' => '0', > 'Caller-Channel-Progress-Media-Time' => '1420586929203735', > 'Caller-Channel-Hangup-Time' => '0', > > I would expect to generate the duration of the call by subtracting > Caller-Channel-Answered-Time from Caller-Channel-Hangup-Time, but the > hangup time is 0. The actual call in this case was 45 seconds long. > > What can I do use Freeswitch events to compute call duration? > > Thanks! > > > > _________________________________________________________________________ > 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/20150106/3370a391/attachment.html From brian at freeswitch.org Wed Jan 7 03:47:54 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2015 18:47:54 -0600 Subject: [Freeswitch-users] getting call duration with event socket In-Reply-To: References: Message-ID: Also if you use CHANNEL_HANGUP_COMPLETE the data is already calculated for you: variable_billsec: 17 variable_billmsec: 16660 variable_billusec: 16660036 Need more granularity? :) On Tue, Jan 6, 2015 at 6:42 PM, Brian West wrote: > Use the CHANNEL_HANGUP_COMPLETE event for this purpose. > > On Tue, Jan 6, 2015 at 5:34 PM, Bruce Lefko wrote: > >> Hi there. >> >> I'm trying to get the call duration in seconds using ESL, but it looks >> like the channel variables are not being set correctly. >> >> In CHANNEL_HANGUP events, I see the following variables: >> >> 'Caller-Profile-Created-Time' => '1420586926995761', >> 'Caller-Channel-Created-Time' => '1420586926995761', >> 'Caller-Channel-Answered-Time' => '1420586929503733', >> 'Caller-Channel-Progress-Time' => '0', >> 'Caller-Channel-Progress-Media-Time' => '1420586929203735', >> 'Caller-Channel-Hangup-Time' => '0', >> >> I would expect to generate the duration of the call by subtracting >> Caller-Channel-Answered-Time from Caller-Channel-Hangup-Time, but the >> hangup time is 0. The actual call in this case was 45 seconds long. >> >> What can I do use Freeswitch events to compute call duration? >> >> Thanks! >> >> >> >> _________________________________________________________________________ >> 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 > -- *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/20150106/096ac4b9/attachment-0001.html From greg at jazzmessengers.com Wed Jan 7 03:23:23 2015 From: greg at jazzmessengers.com (greg at jazzmessengers.com) Date: Wed, 07 Jan 2015 00:23:23 +0000 Subject: [Freeswitch-users] Webrtc no video Message-ID: <20150107002323.Horde.whKZOxcHSq_rXKkYo1XZHQ1@webmail.jazzmessengers.com> Hi folks, I have just installed freeswitch with a basic configuration. Actually, i trying to get to work a call with video. I'm using sipjs to set up my client. Each call has audio but no video. After some days checking the docs and some test, i don't really know where could come from this problem. Maybe the codec. Here, from fs_cli, the codecs list i have installed: === codec,ADPCM (IMA),mod_spandsp codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,G.729,mod_g729 codec,GSM,mod_spandsp codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_spandsp codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,CORE_SPEEX_MODULE 27 total. ==== I tried to call to soft sip (x-lite on my computer, zoiper on my iphone 4s). here some log call accepted: ============== 2015-01-07 01:14:43.534992 [NOTICE] sofia.c:7475 Channel [sofia/internal/sip:1002 at 62.57.238.211:36178] has been answered 2015-01-07 01:14:43.534992 [DEBUG] switch_channel.c:3689 (sofia/internal/sip:1002 at 62.57.238.211:36178) Callstate Change RINGING -> ACTIVE 2015-01-07 01:14:43.554994 [DEBUG] switch_core_codec.c:246 sofia/internal/1000 at 37.187.113.94 Restore previous codec PCMA:8. 2015-01-07 01:14:43.554994 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1000 at 37.187.113.94: v=0 o=FreeSWITCH 1420564044 1420564046 IN IP4 37.187.113.94 s=FreeSWITCH c=IN IP4 37.187.113.94 t=0 0 m=audio 25628 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv =============== when i active the video from one of the client: ==== 2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1002 at 62.57.238.211:36178 entering state [received][100] 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=- 13065063286487275 4 IN IP4 62.57.238.211 s=X-Lite release 4.7.1 stamp 74247 c=IN IP4 62.57.238.211 t=0 0 m=audio 55746 RTP/AVP 8 0 101 125 100 9 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 m=video 58218 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3543 Set telephone-event payload to 101 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:2]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:2]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3881 Set 2833 dtmf send payload to 101 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:5124 Audio params are unchanged for sofia/internal/sip:1002 at 62.57.238.211:36178. 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:7259 Processing updated SDP 2015-01-07 01:18:23.114995 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.134997 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1002 at 62.57.238.211:36178 entering state [completed][200] 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.334992 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1002 at 62.57.238.211:36178 entering state [ready][200] ==== Thank in advance for your help Greg From GeorgePhelps at gfphelps.com Wed Jan 7 04:24:22 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 6 Jan 2015 20:24:22 -0500 Subject: [Freeswitch-users] Register->From->Caller ID In-Reply-To: References: <0aa001d02a04$620588b0$26109a10$@gfphelps.com> Message-ID: <0ad701d02a18$aba19720$02e4c560$@gfphelps.com> Brian West, I?m trying different tests to find a way to set the outgoing caller ID when using switch2voip.us. I?m attempting to emulate the results of their Asterisk configuration example (with Freeswitch configuration). They told me just to set ?fromuser? as shown below, in the SIP REGISTRATION message. [Switch2Voip] username={USERNAME} type=peer secret={PASSWORD} progressinband=never port=5060 nat=auto insecure=very ignoresdpversion=yes host=sip.switch2voip.us dtmfmode=rfc2833 context=from-trunk canreinvite=no allow=g729&g711&g723 fromuser=+{CALLER ID} Taken from this URL: http://switch2voip.us/index.php/customer-support/byod/asterisk-sip-trunk-configuration I?m not convinced that they want to exactly see? From: ?1404XXXXXXX? >;tag=FQag8jSNpac9c ?but I wanted to try. I can originate calls, just not override the outgoing caller ID. RFC 3261 does allow for an optional display-name field, along with the required URI field, in the From header. From RFC 3261: From: "Bob" ;tag=a48s From: sip:+12125551212 at phone2net.com;tag=887s From: Anonymous ;tag=hyh8 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, January 06, 2015 7:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Register->From->Caller ID gateway->register_from = switch_core_sprintf(gateway->pool, " >", from_user, !zstr(from_domain) ? from_domain : proxy); This is how the from is built in the code. Do you have a use case that Requires it? I can't think of a reason it would matter. On Tue, Jan 6, 2015 at 4:59 PM, George F. Phelps wrote: My current gateway REGISTER packet has a certain FROM header, as shown below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: >;tag=FQag8jSNpac9c To: > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: . . . What do I set/change in my gateway configuration to add a character string ?1404XXXXXXX? to the FROM header? As shown in red below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: ?1404XXXXXXX? >;tag=FQag8jSNpac9c To: > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: . . . I am already setting the following values (in my gateway definition): 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/c29a3f3f/attachment-0001.html From brian at freeswitch.org Wed Jan 7 04:42:35 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2015 19:42:35 -0600 Subject: [Freeswitch-users] Register->From->Caller ID In-Reply-To: <0ad701d02a18$aba19720$02e4c560$@gfphelps.com> References: <0aa001d02a04$620588b0$26109a10$@gfphelps.com> <0ad701d02a18$aba19720$02e4c560$@gfphelps.com> Message-ID: Having it on a register won't affect the CID, Can you show me the INVITE packet that Asterisk sends that works for you? On Tue, Jan 6, 2015 at 7:24 PM, George F. Phelps wrote: > Brian West, > > > > I?m trying different tests to find a way to set the outgoing caller ID > when using switch2voip.us. I?m attempting to emulate the results of > their Asterisk configuration example (with Freeswitch configuration). They > told me just to set ?fromuser? as shown below, in the SIP REGISTRATION > message. > > > > [Switch2Voip] > > username={USERNAME} > > type=peer > > secret={PASSWORD} > > progressinband=never > > port=5060 > > nat=auto > > insecure=very > > ignoresdpversion=yes > > host=sip.switch2voip.us > > dtmfmode=rfc2833 > > context=from-trunk > > canreinvite=no > > allow=g729&g711&g723 > > fromuser=+{CALLER ID} > > > > Taken from this URL: > > > > > http://switch2voip.us/index.php/customer-support/byod/asterisk-sip-trunk-configuration > > > > I?m not convinced that they want to exactly see? > > > > From: ?1404XXXXXXX? ;tag=FQag8jSNpac9c > > > > ?but I wanted to try. I can originate calls, just not override the > outgoing caller ID. > > > > RFC 3261 does allow for an optional display-name field, along with the > required URI field, in the From header. From RFC 3261: > > > > From: "Bob" ;tag=a48s > > From: sip:+12125551212 at phone2net.com;tag=887s > > From: Anonymous ;tag=hyh8 > > > > 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, January 06, 2015 7:38 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Register->From->Caller ID > > > > > > gateway->register_from = switch_core_sprintf(gateway->pool, "", from_user, > !zstr(from_domain) ? from_domain : proxy); > > This is how the from is built in the code. Do you have a use case that > Requires it? I can't think of a reason it would matter. > > > > > > > > On Tue, Jan 6, 2015 at 4:59 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > My current gateway REGISTER packet has a certain FROM header, as shown > below: > > > > REGISTER sip:66.33.147.150;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp > > Max-Forwards: 70 > > From: ;tag=FQag8jSNpac9c > > To: > > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce > > CSeq: 69955649 REGISTER > > Contact: switch2voip.us> > > . . . > > > > What do I set/change in my gateway configuration to add a character string > ?1404XXXXXXX? to the FROM header? As shown in red below: > > > > REGISTER sip:66.33.147.150;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp > > Max-Forwards: 70 > > From: ?1404XXXXXXX? ;tag=FQag8jSNpac9c > > To: > > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce > > CSeq: 69955649 REGISTER > > Contact: switch2voip.us> > > . . . > > > > I am already setting the following values (in my gateway definition): > > > > > > > > > > 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 > -- *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/20150106/794f5351/attachment.html From Paul at timefortitan.com Wed Jan 7 04:53:50 2015 From: Paul at timefortitan.com (Paul Nebb) Date: Wed, 7 Jan 2015 01:53:50 +0000 Subject: [Freeswitch-users] BLF on Cisco 303/504/525 Message-ID: Does anyone have templates or articles that would assist in setting up BLF on Cisco 303/504/525 Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/532be06a/attachment-0001.html From GeorgePhelps at gfphelps.com Wed Jan 7 05:09:38 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 6 Jan 2015 21:09:38 -0500 Subject: [Freeswitch-users] Register->From->Caller ID In-Reply-To: References: <0aa001d02a04$620588b0$26109a10$@gfphelps.com> <0ad701d02a18$aba19720$02e4c560$@gfphelps.com> Message-ID: <0afb01d02a1e$fe442a50$facc7ef0$@gfphelps.com> I wish I could, but no. I was previously using pbxes.com and overriding the outgoing caller ID worked. I believe pbxes.com is Asterisk based? Anyway, I don?t have a SIP trace from pbxes.com. Sorry. For what it?s worth, my pbxes.com trunk configuration, using their GUI, was: username=?1404XXXXXXX? password=?password:username? SIP-proxy=?66.33.147.150? And my outgoing caller ID would be set to ?1404XXXXXXX?? 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, January 06, 2015 8:43 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Register->From->Caller ID Having it on a register won't affect the CID, Can you show me the INVITE packet that Asterisk sends that works for you? On Tue, Jan 6, 2015 at 7:24 PM, George F. Phelps wrote: Brian West, I?m trying different tests to find a way to set the outgoing caller ID when using switch2voip.us. I?m attempting to emulate the results of their Asterisk configuration example (with Freeswitch configuration). They told me just to set ?fromuser? as shown below, in the SIP REGISTRATION message. [Switch2Voip] username={USERNAME} type=peer secret={PASSWORD} progressinband=never port=5060 nat=auto insecure=very ignoresdpversion=yes host=sip.switch2voip.us dtmfmode=rfc2833 context=from-trunk canreinvite=no allow=g729&g711&g723 fromuser=+{CALLER ID} Taken from this URL: http://switch2voip.us/index.php/customer-support/byod/asterisk-sip-trunk-configuration I?m not convinced that they want to exactly see? From: ?1404XXXXXXX? >;tag=FQag8jSNpac9c ?but I wanted to try. I can originate calls, just not override the outgoing caller ID. RFC 3261 does allow for an optional display-name field, along with the required URI field, in the From header. From RFC 3261: From: "Bob" >;tag=a48s From: sip:+12125551212 @phone2net.com;tag=887s From: Anonymous >;tag=hyh8 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, January 06, 2015 7:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Register->From->Caller ID gateway->register_from = switch_core_sprintf(gateway->pool, " >", from_user, !zstr(from_domain) ? from_domain : proxy); This is how the from is built in the code. Do you have a use case that Requires it? I can't think of a reason it would matter. On Tue, Jan 6, 2015 at 4:59 PM, George F. Phelps wrote: My current gateway REGISTER packet has a certain FROM header, as shown below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: >;tag=FQag8jSNpac9c To: > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: . . . What do I set/change in my gateway configuration to add a character string ?1404XXXXXXX? to the FROM header? As shown in red below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: ?1404XXXXXXX? >;tag=FQag8jSNpac9c To: > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: . . . I am already setting the following values (in my gateway definition): 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 -- 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/20150106/f4b1d8ea/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/20150106/f4b1d8ea/attachment-0001.bin From mike at jerris.com Wed Jan 7 05:10:40 2015 From: mike at jerris.com (Michael Jerris) Date: Tue, 6 Jan 2015 21:10:40 -0500 Subject: [Freeswitch-users] lua custom variables In-Reply-To: References: Message-ID: <856A21E3-7584-4C84-AD59-C1639F9EB35C@jerris.com> They are different calls to different destinations? > On Jan 6, 2015, at 2:40 PM, David Villasmil Govea wrote: > > Yes, why are sessions different? Because they are different scripts?? Do I need to keep everything in one script? > On Jan 5, 2015 7:26 PM, "Michael Jerris" > wrote: > sounds like the second time its executing on a different session > >> On Jan 4, 2015, at 9:01 PM, David Villasmil Govea > wrote: >> >> Hello guys, >> >> Any help with this? >> >> Thanks >> >> David >> >> On Sun, Jan 4, 2015 at 7:12 AM, David Villasmil Govea > wrote: >> Update: >> >> I execute 2 scripts. >> >> I set an execute_on_answer lua script (lua2), and execute another (lua1). >> like so: >> >> >> >> >> >> >> >> >> >> >> >> >> >> Now, the lua1 executes, i set a variable "custom_dur" which is the duration for the call. >> When lua3 executes on answer, the variable custom_duration is still there, no problems. Then I transfer the call to "9999", which is: >> >> >> >> >> >> >> >> >> Now, on lua3 I try to get the variable and it's not there anymore! >> >> I also tried the following: On lua1 I create a new variable, which is seen on ALL scripts, including lua3, but I reset it to a new value on lua2, and on lua3 I don't see the change!! The variable is still the value I set on lua1!!! wth?? >> >> Am I missing something on variables?? >> On Sun, Jan 4, 2015 at 3:30 AM, David Villasmil Govea > wrote: >> Hello Guys, >> >> I have this lus script, which sets a custom variable, this is ok. >> The the call is transferred to another extension, and the variable is not there anymore, is this by design? > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/a6bebab6/attachment.html From anthony.minessale at gmail.com Wed Jan 7 05:25:16 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jan 2015 20:25:16 -0600 Subject: [Freeswitch-users] Webrtc no video In-Reply-To: <20150107002323.Horde.whKZOxcHSq_rXKkYo1XZHQ1@webmail.jazzmessengers.com> References: <20150107002323.Horde.whKZOxcHSq_rXKkYo1XZHQ1@webmail.jazzmessengers.com> Message-ID: The video support in FS is undergoing massive modification. Support is currently limited to same codecs and in limited setups. When 1.6 beta emerges, give it a try. On Tue, Jan 6, 2015 at 6:23 PM, wrote: > Hi folks, > > I have just installed freeswitch with a basic configuration. Actually, > i trying to get to work a call with video. I'm using sipjs to set up > my client. Each call has audio but no video. > > After some days checking the docs and some test, i don't really know > where could come from this problem. > Maybe the codec. > > Here, from fs_cli, the codecs list i have installed: > > === > codec,ADPCM (IMA),mod_spandsp > codec,AMR,mod_amr > codec,G.711 alaw,CORE_PCM_MODULE > codec,G.711 ulaw,CORE_PCM_MODULE > codec,G.722,mod_spandsp > codec,G.723.1 6.3k,mod_g723_1 > codec,G.726 16k,mod_spandsp > codec,G.726 16k (AAL2),mod_spandsp > codec,G.726 24k,mod_spandsp > codec,G.726 24k (AAL2),mod_spandsp > codec,G.726 32k,mod_spandsp > codec,G.726 32k (AAL2),mod_spandsp > codec,G.726 40k,mod_spandsp > codec,G.726 40k (AAL2),mod_spandsp > codec,G.729,mod_g729 > codec,GSM,mod_spandsp > codec,H.261 Video (passthru),mod_h26x > codec,H.263 Video (passthru),mod_h26x > codec,H.263+ Video (passthru),mod_h26x > codec,H.263++ Video (passthru),mod_h26x > codec,H.264 Video (passthru),mod_h26x > codec,LPC-10,mod_spandsp > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE > codec,Speex,CORE_SPEEX_MODULE > 27 total. > ==== > > > I tried to call to soft sip (x-lite on my computer, zoiper on my iphone > 4s). > here some log > > call accepted: > > ============== > 2015-01-07 01:14:43.534992 [NOTICE] sofia.c:7475 Channel > [sofia/internal/sip:1002 at 62.57.238.211:36178] has been answered > 2015-01-07 01:14:43.534992 [DEBUG] switch_channel.c:3689 > (sofia/internal/sip:1002 at 62.57.238.211:36178) Callstate Change RINGING > -> ACTIVE > 2015-01-07 01:14:43.554994 [DEBUG] switch_core_codec.c:246 > sofia/internal/1000 at 37.187.113.94 Restore previous codec PCMA:8. > 2015-01-07 01:14:43.554994 [DEBUG] mod_sofia.c:780 Local SDP > sofia/internal/1000 at 37.187.113.94: > v=0 > o=FreeSWITCH 1420564044 1420564046 IN IP4 37.187.113.94 > s=FreeSWITCH > c=IN IP4 37.187.113.94 > t=0 0 > m=audio 25628 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > =============== > > when i active the video from one of the client: > > > ==== > > 2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] > 2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] > 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6614 Channel > sofia/internal/sip:1002 at 62.57.238.211:36178 entering state > [received][100] > 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6624 Remote SDP: > v=0 > o=- 13065063286487275 4 IN IP4 62.57.238.211 > s=X-Lite release 4.7.1 stamp 74247 > c=IN IP4 62.57.238.211 > t=0 0 > m=audio 55746 RTP/AVP 8 0 101 125 100 9 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:125 opus/48000/2 > a=fmtp:125 useinbandfec=1 > a=rtpmap:100 speex/16000 > m=video 58218 RTP/AVP 115 34 > a=rtpmap:115 H263-1998/90000 > a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 > a=rtpmap:34 H263/90000 > a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 > a=rtcp-fb:* nack pli > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio > Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio > Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3543 Set > telephone-event payload to 101 > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [opus:125:48000:20:0:2]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [opus:125:48000:20:0:2]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [speex:100:16000:20:0:1]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [speex:100:16000:20:0:1]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3881 Set 2833 > dtmf send payload to 101 > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:5124 Audio > params are unchanged for sofia/internal/sip:1002 at 62.57.238.211:36178. > 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:7259 Processing updated SDP > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] > 2015-01-07 01:18:23.134997 [DEBUG] sofia.c:6614 Channel > sofia/internal/sip:1002 at 62.57.238.211:36178 entering state > [completed][200] > 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] > 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] > 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 [BREAK] > 2015-01-07 01:18:23.334992 [DEBUG] sofia.c:6614 Channel > sofia/internal/sip:1002 at 62.57.238.211:36178 entering state [ready][200] > > ==== > > > Thank in advance for your help > > Greg > > > _________________________________________________________________________ > 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150106/534fa1ca/attachment-0001.html From greg at jazzmessengers.com Wed Jan 7 11:23:40 2015 From: greg at jazzmessengers.com (Greg) Date: Wed, 07 Jan 2015 09:23:40 +0100 Subject: [Freeswitch-users] Webrtc no video In-Reply-To: References: <20150107002323.Horde.whKZOxcHSq_rXKkYo1XZHQ1@webmail.jazzmessengers.com> Message-ID: <54ACED0C.2060705@jazzmessengers.com> Thank you for your answer ! I will try to talk with the FS consulting service. El 07/01/2015 a las 3:25, Anthony Minessale escribi?: > The video support in FS is undergoing massive modification. > Support is currently limited to same codecs and in limited setups. > > When 1.6 beta emerges, give it a try. > > On Tue, Jan 6, 2015 at 6:23 PM, > wrote: > > Hi folks, > > I have just installed freeswitch with a basic configuration. Actually, > i trying to get to work a call with video. I'm using sipjs to set up > my client. Each call has audio but no video. > > After some days checking the docs and some test, i don't really know > where could come from this problem. > Maybe the codec. > > Here, from fs_cli, the codecs list i have installed: > > === > codec,ADPCM (IMA),mod_spandsp > codec,AMR,mod_amr > codec,G.711 alaw,CORE_PCM_MODULE > codec,G.711 ulaw,CORE_PCM_MODULE > codec,G.722,mod_spandsp > codec,G.723.1 6.3k,mod_g723_1 > codec,G.726 16k,mod_spandsp > codec,G.726 16k (AAL2),mod_spandsp > codec,G.726 24k,mod_spandsp > codec,G.726 24k (AAL2),mod_spandsp > codec,G.726 32k,mod_spandsp > codec,G.726 32k (AAL2),mod_spandsp > codec,G.726 40k,mod_spandsp > codec,G.726 40k (AAL2),mod_spandsp > codec,G.729,mod_g729 > codec,GSM,mod_spandsp > codec,H.261 Video (passthru),mod_h26x > codec,H.263 Video (passthru),mod_h26x > codec,H.263+ Video (passthru),mod_h26x > codec,H.263++ Video (passthru),mod_h26x > codec,H.264 Video (passthru),mod_h26x > codec,LPC-10,mod_spandsp > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE > codec,Speex,CORE_SPEEX_MODULE > 27 total. > ==== > > > I tried to call to soft sip (x-lite on my computer, zoiper on my > iphone 4s). > here some log > > call accepted: > > ============== > 2015-01-07 01:14:43.534992 [NOTICE] sofia.c:7475 Channel > [sofia/internal/sip:1002 at 62.57.238.211:36178 > ] has been answered > 2015-01-07 01:14:43.534992 [DEBUG] switch_channel.c:3689 > (sofia/internal/sip:1002 at 62.57.238.211:36178 > ) Callstate Change RINGING > -> ACTIVE > 2015-01-07 01:14:43.554994 [DEBUG] switch_core_codec.c:246 > sofia/internal/1000 at 37.187.113.94 > Restore previous codec PCMA:8. > 2015-01-07 01:14:43.554994 [DEBUG] mod_sofia.c:780 Local SDP > sofia/internal/1000 at 37.187.113.94 : > v=0 > o=FreeSWITCH 1420564044 1420564046 IN IP4 37.187.113.94 > s=FreeSWITCH > c=IN IP4 37.187.113.94 > t=0 0 > m=audio 25628 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > =============== > > when i active the video from one of the client: > > > ==== > > 2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 > [BREAK] > 2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 > [BREAK] > 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6614 Channel > sofia/internal/sip:1002 at 62.57.238.211:36178 > entering state > [received][100] > 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6624 Remote SDP: > v=0 > o=- 13065063286487275 4 IN IP4 62.57.238.211 > s=X-Lite release 4.7.1 stamp 74247 > c=IN IP4 62.57.238.211 > t=0 0 > m=audio 55746 RTP/AVP 8 0 101 125 100 9 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:125 opus/48000/2 > a=fmtp:125 useinbandfec=1 > a=rtpmap:100 speex/16000 > m=video 58218 RTP/AVP 115 34 > a=rtpmap:115 H263-1998/90000 > a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 > a=rtpmap:34 H263/90000 > a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 > a=rtcp-fb:* nack pli > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio > Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio > Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3543 Set > telephone-event payload to 101 > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [opus:125:48000:20:0:2]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [opus:125:48000:20:0:2]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [speex:100:16000:20:0:1]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [speex:100:16000:20:0:1]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio > Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3881 Set 2833 > dtmf send payload to 101 > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:5124 Audio > params are unchanged for > sofia/internal/sip:1002 at 62.57.238.211:36178 > . > 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:7259 Processing updated SDP > 2015-01-07 01:18:23.114995 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 > [BREAK] > 2015-01-07 01:18:23.134997 [DEBUG] sofia.c:6614 Channel > sofia/internal/sip:1002 at 62.57.238.211:36178 > entering state > [completed][200] > 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 > [BREAK] > 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 > [BREAK] > 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/sip:1002 at 62.57.238.211:36178 > [BREAK] > 2015-01-07 01:18:23.334992 [DEBUG] sofia.c:6614 Channel > sofia/internal/sip:1002 at 62.57.238.211:36178 > entering state [ready][200] > > ==== > > > Thank in advance for your help > > Greg > > > _________________________________________________________________________ > 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 > > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? > _http://freeswitch.org/g+_ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org > ? +19193869900 > > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/18bf065c/attachment.html From david.villasmil at gmail.com Wed Jan 7 12:45:08 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 7 Jan 2015 10:45:08 +0100 Subject: [Freeswitch-users] lua custom variables In-Reply-To: References: Message-ID: Hello, Not really, no. I receive a call, send it to the b side with a script. When answered another script is launched which does a few checks and transfers it to another extension which then sets the hangup time with a variable set previously with the script launched at answer. Problem is I can't see the variable set with the answer script. On Jan 5, 2015 7:26 PM, "Michael Jerris" wrote: > sounds like the second time its executing on a different session > > On Jan 4, 2015, at 9:01 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Hello guys, > > Any help with this? > > Thanks > > David > > On Sun, Jan 4, 2015 at 7:12 AM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> Update: >> >> I execute 2 scripts. >> >> I set an execute_on_answer lua script (lua2), and execute another (lua1). >> like so: >> >> >> >> >> >> >> >> >> >> >> >> >> >> Now, the lua1 executes, i set a variable "custom_dur" which is the >> duration for the call. >> When lua3 executes on answer, the variable custom_duration is still >> there, no problems. Then I transfer the call to "9999", which is: >> >> >> >> >> >> >> >> >> Now, on lua3 I try to get the variable and it's not there anymore! >> >> I also tried the following: On lua1 I create a new variable, which is >> seen on ALL scripts, including lua3, but I reset it to a new value on lua2, >> and on lua3 I don't see the change!! The variable is still the value I set >> on lua1!!! wth?? >> >> Am I missing something on variables?? >> On Sun, Jan 4, 2015 at 3:30 AM, David Villasmil Govea < >> david.villasmil at gmail.com> wrote: >> >>> Hello Guys, >>> >>> I have this lus script, which sets a custom variable, this is ok. >>> The the call is transferred to another extension, and the variable is >>> not there anymore, is this by design? >>> >> > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/5e74f50e/attachment-0001.html From raphael.lechner at gmail.com Wed Jan 7 13:28:55 2015 From: raphael.lechner at gmail.com (Raphael Lechner) Date: Wed, 7 Jan 2015 11:28:55 +0100 Subject: [Freeswitch-users] BLF status not always correct Message-ID: <6BBF07DE-9E58-402D-A6E0-D905F9565922@gmail.com> Hi, Sometimes we have the issue that the status for the BLF linekeys don?t have the correct status. We have the same issue on different installations (different FreeSWITCH 1.4.X versions and different phone vendor/types, snom/yealink). Today we found the issue on a Installation with FreeSWITCH 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit). sip_subscriptions and sip_presence tables(at that moment the phone don?t have the right status) https://pastebin.freeswitch.org/23834 Is that a known issue? How can I debug that and is there a workaround/solution for that? After rebooting the phones the status is working again and after some days the issue is back. On newer versions > 1.4.8 the problem is don?t occur so frequently. Thank you, Raphael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/71dc5f44/attachment.html From vipkilla at gmail.com Wed Jan 7 16:09:58 2015 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 7 Jan 2015 08:09:58 -0500 Subject: [Freeswitch-users] BLF on Cisco 303/504/525 In-Reply-To: References: Message-ID: Hi Paul, I have BLF and SLA setup on a 525G Here is the template the phone downloads via tftp: tftp://192.168.0.158:69/Cisco/SPA504G/$MA.xml 1 0.us.pool.ntp.org 1.us.pool.ntp.org GMT-05:00 Yes Yes 0.020 *1 Yes Yes RFC3265_4235 ** 1 Ext 1001 private private 3.local 192.168.0.100 Yes Yes 1001 ************** No [x*]. 1001 at 3.local 300 300 300 2 SLA 1002 shared shared 3.local sipproxy.voip.demo.i-evolve.net Yes Yes 1002 ************** No [x*]. 1002 at 3.local 300 300 300 Disabled Parking private private 3.local 192.168.0.100 Yes Yes fnc=sd+blf;sub=park+701 at 3.local ;ext=park+701 at 3.local;usr=park+701 at 3.local 300 300 300 4 SLA 1004 shared shared 3.local 192.168.0.100 Yes Yes 1004 ************** No [x*]. 1004 at 3.local 300 300 300 Disabled On Tue, Jan 6, 2015 at 8:53 PM, Paul Nebb wrote: > Does anyone have templates or articles that would assist in setting up > BLF on Cisco 303/504/525 > > > > Paul > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/06835df9/attachment-0001.html From avi at avimarcus.net Wed Jan 7 16:41:28 2015 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 7 Jan 2015 13:41:28 +0000 Subject: [Freeswitch-users] Help Needed Debugging Lua Script In-Reply-To: References: Message-ID: <0000014ac4a0ec5e-81630ae2-7f78-4b3a-8659-6ba76a3c9b96-000000@email.amazonses.com> Two things: 1) You aren't grabbing the arg, but the channel variable.. try this in your script: caller_id_number = argv[1] number_to_call = argv[2] 2) I don't think you're managing your hangup/callback originate properly. I don't think you want to use bgapi... or maybe you just need a destination. It's "originate sofia/A endpoint" -- you need to specify where it goes to, the lua script can't "receive" the call. You can have it received by e.g: &lua(pickup.lua) api = freeswitch.API() api:execute("originate", DialString.." &lua(pickup.lua)"); Also: Maybe you want to use it as a hangup hook. Instead of: Do: -Avi On Wed, Jan 7, 2015 at 1:52 AM, Sina Owolabi wrote: > Hi List, > > FreeSWITCH newbie here again. > I am trying to cobble togther a lua callback script, my first attempt > was successful, but I am trying to make it slightly more elegant. > I don't see any errors when I try to run this but the callback isnt > happening. > This is my very second attempt trying to write in lua, so I would be > very grateful for any help. > > The user is expected to dial in, have the call hangup and FreeSWITCH call > back. > > I'm passing a modified $effective_caller_id_number and > $destination_number to the lua script: > > > expression="^1(\d{10})$"require-nested="false"> > data="effective_caller_id_number=+234${1}"/> > > > > data="destination_number=+12312345${1}${2}" /> > > > > > > > The script itself: > > api = freeswitch.API(); > call_string = "bagpi originate > > {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" > > freeswitch.msleep(5000); > if (session:ready()) then > caller_id_number = session:getVariable("destination_number"); > caller_id_name = session:getVariable("destination_number"); > number_to_call = session:getVariable("effective_caller_id_number"); > > api:executeString(call_string); > freeswitch.msleep(2000); > session:streamFile("/tmp/get_off_my_lawn.wav"); > session:hangup("NORMAL_CLEARING"); > end > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/6fc56d2e/attachment.html From brian at freeswitch.org Wed Jan 7 17:50:02 2015 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jan 2015 08:50:02 -0600 Subject: [Freeswitch-users] Register->From->Caller ID In-Reply-To: <0afb01d02a1e$fe442a50$facc7ef0$@gfphelps.com> References: <0aa001d02a04$620588b0$26109a10$@gfphelps.com> <0ad701d02a18$aba19720$02e4c560$@gfphelps.com> <0afb01d02a1e$fe442a50$facc7ef0$@gfphelps.com> Message-ID: https://wiki.freeswitch.org/wiki/Variable_sip_cid_type https://wiki.freeswitch.org/wiki/Variable_effective_caller_id_name https://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number There are usually only three ways to send CID, From, RPID, PID Those links should give you an idea of what you need to try. On Tue, Jan 6, 2015 at 8:09 PM, George F. Phelps wrote: > I wish I could, but no. > > > > I was previously using pbxes.com and overriding the outgoing caller ID > worked. I believe pbxes.com is Asterisk based? Anyway, I don?t have a > SIP trace from pbxes.com. Sorry. > > > > For what it?s worth, my pbxes.com trunk configuration, using their GUI, > was: > > > > username=?1404XXXXXXX? > > password=?password:username? > > SIP-proxy=?66.33.147.150? > > > > And my outgoing caller ID would be set to ?1404XXXXXXX?? > > > > 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, January 06, 2015 8:43 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Register->From->Caller ID > > > > Having it on a register won't affect the CID, Can you show me the INVITE > packet that Asterisk sends that works for you? > > > > On Tue, Jan 6, 2015 at 7:24 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > Brian West, > > > > I?m trying different tests to find a way to set the outgoing caller ID > when using switch2voip.us. I?m attempting to emulate the results of > their Asterisk configuration example (with Freeswitch configuration). They > told me just to set ?fromuser? as shown below, in the SIP REGISTRATION > message. > > > > [Switch2Voip] > > username={USERNAME} > > type=peer > > secret={PASSWORD} > > progressinband=never > > port=5060 > > nat=auto > > insecure=very > > ignoresdpversion=yes > > host=sip.switch2voip.us > > dtmfmode=rfc2833 > > context=from-trunk > > canreinvite=no > > allow=g729&g711&g723 > > fromuser=+{CALLER ID} > > > > Taken from this URL: > > > > > http://switch2voip.us/index.php/customer-support/byod/asterisk-sip-trunk-configuration > > > > I?m not convinced that they want to exactly see? > > > > From: ?1404XXXXXXX? ;tag=FQag8jSNpac9c > > > > ?but I wanted to try. I can originate calls, just not override the > outgoing caller ID. > > > > RFC 3261 does allow for an optional display-name field, along with the > required URI field, in the From header. From RFC 3261: > > > > From: "Bob" ;tag=a48s > > From: sip:+12125551212 at phone2net.com;tag=887s > > From: Anonymous ;tag=hyh8 > > > > 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, January 06, 2015 7:38 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Register->From->Caller ID > > > > > > gateway->register_from = switch_core_sprintf(gateway->pool, "", from_user, > !zstr(from_domain) ? from_domain : proxy); > > This is how the from is built in the code. Do you have a use case that > Requires it? I can't think of a reason it would matter. > > > > > > > > On Tue, Jan 6, 2015 at 4:59 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > My current gateway REGISTER packet has a certain FROM header, as shown > below: > > > > REGISTER sip:66.33.147.150;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp > > Max-Forwards: 70 > > From: ;tag=FQag8jSNpac9c > > To: > > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce > > CSeq: 69955649 REGISTER > > Contact: switch2voip.us> > > . . . > > > > What do I set/change in my gateway configuration to add a character string > ?1404XXXXXXX? to the FROM header? As shown in red below: > > > > REGISTER sip:66.33.147.150;transport=udp SIP/2.0 > > Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp > > Max-Forwards: 70 > > From: ?1404XXXXXXX? ;tag=FQag8jSNpac9c > > To: > > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce > > CSeq: 69955649 REGISTER > > Contact: switch2voip.us> > > . . . > > > > I am already setting the following values (in my gateway definition): > > > > > > > > > > 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 > > > > > > -- > > *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 > -- *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/20150107/a723b7c2/attachment-0001.html From brian at freeswitch.org Wed Jan 7 17:51:48 2015 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jan 2015 08:51:48 -0600 Subject: [Freeswitch-users] lua custom variables In-Reply-To: References: Message-ID: Why are you doing anything at all to set variables with hangup times, FreeSWITCH does all this for you already, maybe can you step back and tell me exactly what you're trying to accomplish? On Wed, Jan 7, 2015 at 3:45 AM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Hello, > > Not really, no. > > I receive a call, send it to the b side with a script. When answered > another script is launched which does a few checks and transfers it to > another extension which then sets the hangup time with a variable set > previously with the script launched at answer. Problem is I can't see the > variable set with the answer script. > On Jan 5, 2015 7:26 PM, "Michael Jerris" wrote: > >> sounds like the second time its executing on a different session >> >> On Jan 4, 2015, at 9:01 PM, David Villasmil Govea < >> david.villasmil at gmail.com> wrote: >> >> Hello guys, >> >> Any help with this? >> >> Thanks >> >> David >> >> On Sun, Jan 4, 2015 at 7:12 AM, David Villasmil Govea < >> david.villasmil at gmail.com> wrote: >> >>> Update: >>> >>> I execute 2 scripts. >>> >>> I set an execute_on_answer lua script (lua2), and execute another (lua1). >>> like so: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Now, the lua1 executes, i set a variable "custom_dur" which is the >>> duration for the call. >>> When lua3 executes on answer, the variable custom_duration is still >>> there, no problems. Then I transfer the call to "9999", which is: >>> >>> >>> >>> >>> >>> >>> >>> >>> Now, on lua3 I try to get the variable and it's not there anymore! >>> >>> I also tried the following: On lua1 I create a new variable, which is >>> seen on ALL scripts, including lua3, but I reset it to a new value on lua2, >>> and on lua3 I don't see the change!! The variable is still the value I set >>> on lua1!!! wth?? >>> >>> Am I missing something on variables?? >>> On Sun, Jan 4, 2015 at 3:30 AM, David Villasmil Govea < >>> david.villasmil at gmail.com> wrote: >>> >>>> Hello Guys, >>>> >>>> I have this lus script, which sets a custom variable, this is ok. >>>> The the call is transferred to another extension, and the variable is >>>> not there anymore, is this by design? >>>> >>> >> >> _________________________________________________________________________ >> 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/20150107/67fa0729/attachment.html From david.villasmil at gmail.com Wed Jan 7 18:18:42 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 7 Jan 2015 16:18:42 +0100 Subject: [Freeswitch-users] lua custom variables In-Reply-To: References: Message-ID: What I'm doing is the following: - Receive from A side - Launch a lua to get a destination number, and send the call to the B-side. - When the call is answered, I launch another lua which make a few checks and set a variable with the maximum call duration. It is then transferred, only the b side to an extension. - said extension launches a lua which should use the max call duration to set a sched_hangup and then add the b side to a queue (mod_callcenter) Thanks for your help David On Jan 5, 2015 7:26 PM, "Michael Jerris" wrote: > sounds like the second time its executing on a different session > > On Jan 4, 2015, at 9:01 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Hello guys, > > Any help with this? > > Thanks > > David > > On Sun, Jan 4, 2015 at 7:12 AM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> Update: >> >> I execute 2 scripts. >> >> I set an execute_on_answer lua script (lua2), and execute another (lua1). >> like so: >> >> >> >> >> >> >> >> >> >> >> >> >> >> Now, the lua1 executes, i set a variable "custom_dur" which is the >> duration for the call. >> When lua3 executes on answer, the variable custom_duration is still >> there, no problems. Then I transfer the call to "9999", which is: >> >> >> >> >> >> >> >> >> Now, on lua3 I try to get the variable and it's not there anymore! >> >> I also tried the following: On lua1 I create a new variable, which is >> seen on ALL scripts, including lua3, but I reset it to a new value on lua2, >> and on lua3 I don't see the change!! The variable is still the value I set >> on lua1!!! wth?? >> >> Am I missing something on variables?? >> On Sun, Jan 4, 2015 at 3:30 AM, David Villasmil Govea < >> david.villasmil at gmail.com> wrote: >> >>> Hello Guys, >>> >>> I have this lus script, which sets a custom variable, this is ok. >>> The the call is transferred to another extension, and the variable is >>> not there anymore, is this by design? >>> >> > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/99a6e656/attachment-0001.html From shisheer at tifr.res.in Wed Jan 7 18:34:54 2015 From: shisheer at tifr.res.in (Shisheer Teli) Date: Wed, 7 Jan 2015 21:04:54 +0530 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch Message-ID: Hi Team, I don't know what happen , but when I start video call it disconnected after every 30 seconds. e.g. x-lite to x-lite call : video call disconnect after 30 seconds X-lite to Zoiper : video call continue, but no video sending. Regards Shisheer T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/8831c20f/attachment.html From nneul at mst.edu Wed Jan 7 19:20:37 2015 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 07 Jan 2015 10:20:37 -0600 Subject: [Freeswitch-users] Testing inbound calls from LOTS of origin points? Message-ID: <54AD5CD5.3030900@mst.edu> Anyone know of any good way to test inbound calls from lots of different origins? Similar to what you do with things like 'global dns propagation checker'? I'd like to be able to periodically validate inbound calls from a bunch of different sources - ideally taking different paths inbound. (We've had a number of occasions lately where we suspect a routing problem external to our sip carrier only affecting certain inbound call sources.) I realize I can probably do this by hand, just by getting SIP trunks from a bunch of different providers, but is there any way that is a bit less manual/time consuming/third party to do the heavy lifting/etc? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From anthony.minessale at gmail.com Wed Jan 7 19:22:31 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jan 2015 10:22:31 -0600 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: Try latest master or 1.4.15 On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli wrote: > Hi Team, > > I don't know what happen , but when I start video call it disconnected > after every 30 seconds. > > e.g. > x-lite to x-lite call : video call disconnect after 30 seconds > > X-lite to Zoiper : video call continue, but no video sending. > > > Regards > Shisheer T > > _________________________________________________________________________ > 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/77623482/attachment.html From kris at kriskinc.com Wed Jan 7 20:00:11 2015 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 7 Jan 2015 12:00:11 -0500 Subject: [Freeswitch-users] Testing inbound calls from LOTS of origin points? In-Reply-To: <54AD5CD5.3030900@mst.edu> References: <54AD5CD5.3030900@mst.edu> Message-ID: Hi Nathan, Justin Newman proposed something along these lines about a month ago on Voiceops. He wrote: --- Greetings, I'm currently building out daily testing of a large set of outbound providers against a wide array of destinations. I'm looking for other facilities-based LECs who would offer up a telephone number that either: - answers the phone then plays a recording we provide (consisting of test tones, etc.) - forwards incoming calls to a toll free number (which will answer & play a similar recording) Estimated call volume is 100-200 calls per day, max. of 1 concurrent call. ACD of 15-30s. If you work for a company who would be willing, please drop me an email off list and we can coordinate. The perks of participating include knowing someone's doing a fairly substantial test of a wide array of carriers a couple of times a day, and opening tickets with the carriers when there are problems. We'll also try to provide some visibility into problems, particularly if we see them from the big carriers. Many thanks, -jbn --- There were no replies on list but he may have received some directly... On Wed, Jan 7, 2015 at 11:20 AM, Nathan Neulinger wrote: > Anyone know of any good way to test inbound calls from lots of different origins? Similar to what you do with things > like 'global dns propagation checker'? > > I'd like to be able to periodically validate inbound calls from a bunch of different sources - ideally taking different > paths inbound. (We've had a number of occasions lately where we suspect a routing problem external to our sip carrier > only affecting certain inbound call sources.) > > I realize I can probably do this by hand, just by getting SIP trunks from a bunch of different providers, but is there > any way that is a bit less manual/time consuming/third party to do the heavy lifting/etc? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > 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 -- Kristian Kielhofner From mike at jerris.com Wed Jan 7 20:56:39 2015 From: mike at jerris.com (Michael Jerris) Date: Wed, 7 Jan 2015 12:56:39 -0500 Subject: [Freeswitch-users] BLF status not always correct In-Reply-To: <6BBF07DE-9E58-402D-A6E0-D905F9565922@gmail.com> References: <6BBF07DE-9E58-402D-A6E0-D905F9565922@gmail.com> Message-ID: <3268BFEE-147B-422B-891B-991A159ADBBE@jerris.com> We have fixed a number of issues in this area since 1.4.13. I would try 1.4.15 and see if you still have the issues or not. > On Jan 7, 2015, at 5:28 AM, Raphael Lechner wrote: > > Hi, > > Sometimes we have the issue that the status for the BLF linekeys don?t have the correct status. We have the same issue on different installations (different FreeSWITCH 1.4.X versions and different phone vendor/types, snom/yealink). > Today we found the issue on a Installation with FreeSWITCH 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit). > > sip_subscriptions and sip_presence tables(at that moment the phone don?t have the right status) > https://pastebin.freeswitch.org/23834 > > Is that a known issue? > How can I debug that and is there a workaround/solution for that? > > After rebooting the phones the status is working again and after some days the issue is back. On newer versions > 1.4.8 the problem is don?t occur so frequently. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/bee244f9/attachment.html From juanito1982 at gmail.com Wed Jan 7 21:02:07 2015 From: juanito1982 at gmail.com (=?UTF-8?Q?Juan_Antonio_Iba=C3=B1ez_Santorum?=) Date: Wed, 7 Jan 2015 19:02:07 +0100 Subject: [Freeswitch-users] Adding X-CID custom header for sipcapture integration Message-ID: Hello, I am trying to add custom X-CID header to copy call id from a-leg to b-leg in order to make work call flow in sipcapture. I set channel variable sip_h_X-CID to add it to b-leg, I can see it is setted up but outgoing b-leg request does not contain the header. If I set up sip_h_B-CID it works but sip_h_X-CID not. Do you know why? Any other way to add the header? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/36447372/attachment.html From nneul at mst.edu Wed Jan 7 21:17:09 2015 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 07 Jan 2015 12:17:09 -0600 Subject: [Freeswitch-users] Testing inbound calls from LOTS of origin points? In-Reply-To: References: <54AD5CD5.3030900@mst.edu> Message-ID: <54AD7825.904@mst.edu> Thank you, I will inquire with him to see if he got anywhere with it. On 01/07/2015 11:00 AM, Kristian Kielhofner wrote: > Hi Nathan, > > Justin Newman proposed something along these lines about a month ago > on Voiceops. He wrote: > > --- > Greetings, > > I'm currently building out daily testing of a large set of outbound > providers against a wide array of destinations. I'm looking for other > facilities-based LECs who would offer up a telephone number that > either: > - answers the phone then plays a recording we provide (consisting of > test tones, etc.) > - forwards incoming calls to a toll free number (which will answer & > play a similar recording) > > Estimated call volume is 100-200 calls per day, max. of 1 concurrent > call. ACD of 15-30s. > > If you work for a company who would be willing, please drop me an > email off list and we can coordinate. The perks of participating > include knowing someone's doing a fairly substantial test of a wide > array of carriers a couple of times a day, and opening tickets with > the carriers when there are problems. We'll also try to provide some > visibility into problems, particularly if we see them from the big > carriers. > > Many thanks, > > -jbn > --- > > There were no replies on list but he may have received some directly... > > > On Wed, Jan 7, 2015 at 11:20 AM, Nathan Neulinger wrote: >> Anyone know of any good way to test inbound calls from lots of different origins? Similar to what you do with things >> like 'global dns propagation checker'? >> >> I'd like to be able to periodically validate inbound calls from a bunch of different sources - ideally taking different >> paths inbound. (We've had a number of occasions lately where we suspect a routing problem external to our sip carrier >> only affecting certain inbound call sources.) >> >> I realize I can probably do this by hand, just by getting SIP trunks from a bunch of different providers, but is there >> any way that is a bit less manual/time consuming/third party to do the heavy lifting/etc? >> >> -- Nathan >> >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> 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 > > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From cmrienzo at gmail.com Wed Jan 7 21:49:21 2015 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 7 Jan 2015 13:49:21 -0500 Subject: [Freeswitch-users] Adding X-CID custom header for sipcapture integration In-Reply-To: References: Message-ID: It should work the same way. You can either set that value in the dialstring or export the variable. If you can't make it work with X-CID there is: modparam("sipcapture", "callid_aleg_header ", "X-CID") in your kamailio sipcapture config. On Wed, Jan 7, 2015 at 1:02 PM, Juan Antonio Iba?ez Santorum < juanito1982 at gmail.com> wrote: > Hello, > > I am trying to add custom X-CID header to copy call id from a-leg to > b-leg in order to make work call flow in sipcapture. I set channel variable > sip_h_X-CID to add it to b-leg, I can see it is setted up but outgoing > b-leg request does not contain the header. If I set up sip_h_B-CID it works > but sip_h_X-CID not. > > Do you know why? Any other way to add the header? > > Regards > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/a544eb5f/attachment.html From aqsyounas at gmail.com Thu Jan 8 00:21:10 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 8 Jan 2015 02:21:10 +0500 Subject: [Freeswitch-users] how can i do stream piping in freeswitch Message-ID: Currently i am playing a stream with mod_shout and this is my default xml. Every time a user makes a call for stream it opens a separate connection with the stream provider. If 100 users dials this number, there would be 100 connections with stream provider listening to same stream, means more rtp packets containing same data for different users. What i want is, if a user is listening to stream then other users must share the same listening connection that the first user is opened, instead of creating a separate connection with stream provider for same stream. Someone told me this is possible in asterisk, so there must be a way in freeswitch. How can i do this.? Any help would be much appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/d6994d0f/attachment.html From juanito1982 at gmail.com Thu Jan 8 00:28:27 2015 From: juanito1982 at gmail.com (=?UTF-8?Q?Juan_Antonio_Iba=C3=B1ez_Santorum?=) Date: Wed, 7 Jan 2015 22:28:27 +0100 Subject: [Freeswitch-users] Adding X-CID custom header for sipcapture integration In-Reply-To: References: Message-ID: Good solution Christopher! Thank you Regards 2015-01-07 19:49 GMT+01:00 Christopher Rienzo : > It should work the same way. You can either set that value in the > dialstring or export the variable. If you can't make it work with X-CID > there is: > > modparam("sipcapture", "callid_aleg_header ", "X-CID") > > in your kamailio sipcapture config. > > > On Wed, Jan 7, 2015 at 1:02 PM, Juan Antonio Iba?ez Santorum < > juanito1982 at gmail.com> wrote: > >> Hello, >> >> I am trying to add custom X-CID header to copy call id from a-leg to >> b-leg in order to make work call flow in sipcapture. I set channel variable >> sip_h_X-CID to add it to b-leg, I can see it is setted up but outgoing >> b-leg request does not contain the header. If I set up sip_h_B-CID it works >> but sip_h_X-CID not. >> >> Do you know why? Any other way to add the header? >> >> Regards >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/155754ed/attachment.html From rnbrady at gmail.com Thu Jan 8 01:02:31 2015 From: rnbrady at gmail.com (Richard Brady) Date: Wed, 7 Jan 2015 22:02:31 +0000 Subject: [Freeswitch-users] Multiple P-Asserted-Identity Headers In-Reply-To: <093d01d029b8$bfe63200$3fb29600$@gfphelps.com> References: <076a01d028de$96357180$c2a05480$@gfphelps.com> <7CA4DE25-0C25-4AFA-B4C2-6D887A3A328F@jerris.com> <093d01d029b8$bfe63200$3fb29600$@gfphelps.com> Message-ID: I think ... >From a standards point of view there is no such header as P-Asserted-ID so you've effectively made that one up by setting the sip_h_P-Asserted-ID variable. The P-Asserted-*Identity* header is the standards based one which is inserted by FS if you set something like: {sip_cid_type=pid,effective_caller_id_name=George,effective_caller_id_number=1234} I don't think it supports multiple comma delimited values but you could try suppress it and craft the header yourself: {sip_cid_type=none,sip_h_P-Asserted-Identity=', tel:+1234'} You will need to work our the quotation system and escaping if you want to add a name and you might need to use \, instead of , too. No idea if that would work. Good luck! On 6 January 2015 at 13:57, George F. Phelps wrote: > Michael Jerris, > > > > Thanks for pointing out the comma delimited syntax. However? > > > > I am still trying to get setting my outbound caller ID to work. I > noticed, in this Freeswitch SIP trace? > > > > send 1202 bytes to udp/[66.33.147.150]:5060 at 07:12:09.474756: > > ------------------------------------------------------------------------ > > INVITE sip:1770XXXXXXX at 66.33.147.150 SIP/2.0 > > Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKDN5acD6Zcp2g > > Max-Forwards: 69 > > From: "George F Phelps" >;tag=gpmpmpp142SFH > > To: > > Call-ID: 146d208a-1040-1233-1bbe-0a1aa9c1784d > > CSeq: 69937100 INVITE > > Contact: switch2voip.us> > > User-Agent: > FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > > Supported: timer, path, replaces > > Allow-Events: talk, hold, conference, refer > > Privacy: none > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 251 > > P-Asserted-ID: "George F Phelps", , > tel:+1404XXXXXXX > > X-FS-Support: update_display,send_info > > P-Asserted-Identity: "George F Phelps" > > > > v=0 > > o=FreeSWITCH 1420520731 1420520732 IN IP4 54.174.255.168 > > s=FreeSWITCH > > c=IN IP4 54.174.255.168 > > t=0 0 > > m=audio 25598 RTP/AVP 0 8 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > ------------------------------------------------------------------------ > > > > ?that there is apparently both a ?P-Asserted-Identity? and a ?P-Asserted- > ID? header. What?s up with this? Are there two different headers? Is > the Freeswitch SIP trace a formatted dump of the SIP packet, or is it a > series of debug printf statements from where the data is added to the > packet? > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Jerris > *Sent:* Monday, January 05, 2015 1:34 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Multiple P-Asserted-Identity Headers > > > > put them together as a single header separated by a comma. > http://www.ietf.org/mail-archive/web/sip/current/msg17485.html has more > info. > > > > > > On Jan 5, 2015, at 6:56 AM, George F. Phelps > wrote: > > > > How do I create two (2) unique, P-Asserted-Identity headers? As required > by *RFC 3325 *: > > > > A P-Asserted-Identity header field value MUST consist of exactly one > name-addr or addr-spec. There may be one or two P-Asserted-Identity > values. If there is one value, it MUST be a sip, sips, or tel URI. If > there are two values, one value MUST be a sip or sips URI and the other > MUST be a tel URI. > > > > A valid, multiple P-Asserted-Identity headers example, taken from RFC 3325: > > > > INVITE sip:+14085551212 at proxy.pstn.net SIP/2.0 > > Via: SIP/2.0/TCP useragent.cisco.com;branch=z9hG4bK-124 > > Via: SIP/2.0/TCP proxy.cisco.com;branch=z9hG4bK-abc > > To: > > From: "Anonymous" ;tag=9802748 > > Call-ID: 245780247857024504 > > CSeq: 2 INVITE > > Max-Forwards: 69 > > P-Asserted-Identity: "Cullen Jennings" > > P-Asserted-Identity: tel:+14085264000 <+14085264000> > > Privacy: id > > > > When I use the following two (2) statements in a Freeswitch dialplan, the > second ?set,? of course, overwrites the data stored by the first ?set.? > > > > > > > > > > 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 > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/ebc20dd8/attachment-0001.html From GeorgePhelps at gfphelps.com Thu Jan 8 01:30:30 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Wed, 7 Jan 2015 17:30:30 -0500 Subject: [Freeswitch-users] Multiple P-Asserted-Identity Headers In-Reply-To: References: <076a01d028de$96357180$c2a05480$@gfphelps.com> <7CA4DE25-0C25-4AFA-B4C2-6D887A3A328F@jerris.com> <093d01d029b8$bfe63200$3fb29600$@gfphelps.com> Message-ID: <0c1a01d02ac9$8bde5020$a39af060$@gfphelps.com> Richard Brady, Thanks for the input, particularly regarding ??you've effectively made that one up by setting the sip_h_P-Asserted-ID variable??. I copied and pasted someone else?s bad example code. This Freeswitch syntax is working? ?but I?m still working with my VoIP service provider to find out exactly what they expect. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Brady Sent: Wednesday, January 07, 2015 5:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple P-Asserted-Identity Headers I think ... >From a standards point of view there is no such header as P-Asserted-ID so you've effectively made that one up by setting the sip_h_P-Asserted-ID variable. The P-Asserted-Identity header is the standards based one which is inserted by FS if you set something like: {sip_cid_type=pid,effective_caller_id_name=George,effective_caller_id_number=1234} I don't think it supports multiple comma delimited values but you could try suppress it and craft the header yourself: {sip_cid_type=none,sip_h_P-Asserted-Identity=' >, tel:+1234'} You will need to work our the quotation system and escaping if you want to add a name and you might need to use \, instead of , too. No idea if that would work. Good luck! On 6 January 2015 at 13:57, George F. Phelps wrote: Michael Jerris, Thanks for pointing out the comma delimited syntax. However? I am still trying to get setting my outbound caller ID to work. I noticed, in this Freeswitch SIP trace? send 1202 bytes to udp/[66.33.147.150]:5060 at 07:12:09.474756: ------------------------------------------------------------------------ INVITE sip:1770XXXXXXX at 66.33.147.150 SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKDN5acD6Zcp2g Max-Forwards: 69 From: "George F Phelps" >;tag=gpmpmpp142SFH To: > Call-ID: 146d208a-1040-1233-1bbe-0a1aa9c1784d CSeq: 69937100 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 251 P-Asserted-ID: "George F Phelps", >, tel:+1404XXXXXXX X-FS-Support: update_display,send_info P-Asserted-Identity: "George F Phelps" > v=0 o=FreeSWITCH 1420520731 1420520732 IN IP4 54.174.255.168 s=FreeSWITCH c=IN IP4 54.174.255.168 t=0 0 m=audio 25598 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ ?that there is apparently both a ?P-Asserted-Identity? and a ?P-Asserted-ID? header. What?s up with this? Are there two different headers? Is the Freeswitch SIP trace a formatted dump of the SIP packet, or is it a series of debug printf statements from where the data is added to the packet? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, January 05, 2015 1:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple P-Asserted-Identity Headers put them together as a single header separated by a comma. http://www.ietf.org/mail-archive/web/sip/current/msg17485.html has more info. On Jan 5, 2015, at 6:56 AM, George F. Phelps wrote: How do I create two (2) unique, P-Asserted-Identity headers? As required by RFC 3325: A P-Asserted-Identity header field value MUST consist of exactly one name-addr or addr-spec. There may be one or two P-Asserted-Identity values. If there is one value, it MUST be a sip, sips, or tel URI. If there are two values, one value MUST be a sip or sips URI and the other MUST be a tel URI. A valid, multiple P-Asserted-Identity headers example, taken from RFC 3325: INVITE sip:+14085551212 at proxy.pstn.net SIP/2.0 Via: SIP/2.0/TCP useragent.cisco.com;branch=z9hG4bK-124 Via: SIP/2.0/TCP proxy.cisco.com;branch=z9hG4bK-abc To: From: "Anonymous" ;tag=9802748 Call-ID: 245780247857024504 CSeq: 2 INVITE Max-Forwards: 69 P-Asserted-Identity: "Cullen Jennings" P-Asserted-Identity: tel:+14085264000 Privacy: id When I use the following two (2) statements in a Freeswitch dialplan, the second ?set,? of course, overwrites the data stored by the first ?set.? 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/ddeedf9d/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/20150107/ddeedf9d/attachment-0001.bin From adam.ben.ayoun1 at gmail.com Thu Jan 8 01:21:39 2015 From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun) Date: Thu, 8 Jan 2015 00:21:39 +0200 Subject: [Freeswitch-users] Wrong ice candidate (internal ip) when using mod_verto Message-ID: Hi, I am experimenting with mod_verto and using the demo to call to a conference on my server (logs on FreeSwitch seems ok generally), the issue is that I am not receiving/sending any audio (I would expect to hear moh at this point), I suspect this is because FreeSwitch sends an ice host candidate set to the internal ip instead of the external, here is the SDP: o=FreeSWITCH 1420641299 1420641300 IN IP4 10.245.43.195 s=FreeSWITCH c=IN IP4 10.245.43.195 t=0 0 a=msid-semantic: WMS Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br m=audio 26060 RTP/SAVPF 111 126 a=rtpmap:111 opus/48000/2 a=fmtp:111 maxplaybackrate=48000; minptime=10 a=rtpmap:126 telephone-event/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=fingerprint:sha-256 0D:BB:87:E0:B6:8A:AE:0D:32:3F:54:9F:36:FF:9B:89:28:AB:B4:17:BC:95:0D:2E:48:0C:33:30:78:52:5F:24 a=rtcp-mux a=rtcp:26060 IN IP4 10.245.43.195 a=ssrc:1435389439 cname:TfPkXSQ755IUEU1s a=ssrc:1435389439 msid:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br a0 a=ssrc:1435389439 mslabel:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br a=ssrc:1435389439 label:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Bra0 a=ice-ufrag:hQuEgraFWcUo2lGL a=ice-pwd:qmWe5Cb0xJL2tDkAjF4pW5UJ a=candidate:0460306999 1 udp 659136 10.245.43.195 26060 typ host generation 0 Using Wireshark I see that the client is trying to send packets to 10.245.43.195 which obviously fails.. How can I fix this? Thanks, Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/9d7aaddb/attachment.html From nneul at mst.edu Thu Jan 8 04:22:58 2015 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 07 Jan 2015 19:22:58 -0600 Subject: [Freeswitch-users] how can i do stream piping in freeswitch In-Reply-To: References: Message-ID: <54ADDBF2.50106@mst.edu> Look at the Shout stream as MOH section of: https://wiki.freeswitch.org/wiki/Mod_shout You have to define the stream as a local_stream and then reference it via local_stream://moh/whatever I use this to stream a local campus radio station via mp3/icecast as MOH stream. -- Nathan On 01/07/2015 03:21 PM, Aqs Younas wrote: > Currently i am playing a stream with mod_shout and this is my default xml. > > > > > > > > > > > Every time a user makes a call for stream it opens a separate connection with the stream provider. If 100 users dials > this number, there would be 100 connections with stream provider listening to same stream, means more rtp packets > containing same data for different users. > > What i want is, if a user is listening to stream then other users must share the same listening connection that the > first user is opened, instead of creating a separate connection with stream provider for same stream. > > Someone told me this is possible in asterisk, so there must be a way in freeswitch. > > How can i do this.? > Any help would be much appreciated. > > Thanks > > > > _________________________________________________________________________ > 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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From Paul at timefortitan.com Thu Jan 8 06:45:44 2015 From: Paul at timefortitan.com (Paul Nebb) Date: Thu, 8 Jan 2015 03:45:44 +0000 Subject: [Freeswitch-users] BLF on Cisco 303/504/525 In-Reply-To: References: Message-ID: Thank you, Paul Nebb From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vik Killa Sent: Wednesday, January 07, 2015 8:10 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] BLF on Cisco 303/504/525 Hi Paul, I have BLF and SLA setup on a 525G Here is the template the phone downloads via tftp: tftp://192.168.0.158:69/Cisco/SPA504G/$MA.xml 1 0.us.pool.ntp.org 1.us.pool.ntp.org GMT-05:00 Yes Yes 0.020 *1 Yes Yes RFC3265_4235 ** 1 Ext 1001 private private 3.local 192.168.0.100 Yes Yes 1001 ************** No [x*]. 1001 at 3.local> 300 300 300 2 SLA 1002 shared shared 3.local sipproxy.voip.demo.i-evolve.net Yes Yes 1002 ************** No [x*]. 1002 at 3.local> 300 300 300 Disabled Parking private private 3.local 192.168.0.100 Yes Yes fnc=sd+blf;sub=park+701 at 3.local;ext=park+701 at 3.local;usr=park+701 at 3.local 300 300 300 4 SLA 1004 shared shared 3.local 192.168.0.100 Yes Yes 1004 ************** No [x*]. 1004 at 3.local> 300 300 300 Disabled On Tue, Jan 6, 2015 at 8:53 PM, Paul Nebb > wrote: Does anyone have templates or articles that would assist in setting up BLF on Cisco 303/504/525 Paul _________________________________________________________________________ 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 ________________________________ Total Control Panel Login To: paul at timefortitan.com From: freeswitch-users-bounces at lists.freeswitch.org Message Score: 1 High (60): Pass My Spam Blocking Level: Medium Medium (75): Pass Low (90): Pass Block this sender / Block this sender enterprise-wide Block lists.freeswitch.org / Block lists.freeswitch.org enterprise-wide This message was delivered because the content filter score did not exceed your filter level. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/5e516a20/attachment-0001.html From myforums.indra at gmail.com Thu Jan 8 09:41:24 2015 From: myforums.indra at gmail.com (indra sena) Date: Thu, 8 Jan 2015 12:11:24 +0530 Subject: [Freeswitch-users] Hi Reg IVR and codec negotiation. Message-ID: Hi , Happy new year to every body. I have observed in freeswitch that, In case of IVR scenario prior to the bridge FreeSwitch plays IVR with the 1st priority codec then invite to the bridge endpoint (B-leg) with SDP containing only the codec negotiated for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only with IVR negotiated codec. I am having one issue here for example. Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated with G729 and starts playing IVR with G729 and answered(183) to originator with SDP containing only G729. And Invited to termination endpoint with SDP having only G729 and termination gateway having support of only G711, and it is rejecting call after receiving invite with only G729. It should be work like after IVR , invite can be having sdp with all the originator supported codecs and if termination (b-leg) having different priority codecs then it can send re-Invite sanding the same. I have observed one issue in freeswitch Jeera ( https://freeswitch.org/jira/browse/FS-880) , but there is no solution in that page. Do we have any solution for this right now ? Your answers will be very much helpful for me. I appreciate if you can quick response or give some solution for this. Thanks in advance. Thanks & Regards, Indra.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/bd0c8177/attachment.html From myforums.indra at gmail.com Thu Jan 8 10:09:09 2015 From: myforums.indra at gmail.com (indra sena) Date: Thu, 8 Jan 2015 12:39:09 +0530 Subject: [Freeswitch-users] REg IVR and codec negotiation Message-ID: Hi , I have observed in freeswitch that, In case of IVR scenario prior to the bridge FreeSwitch plays IVR with the 1st priority codec then invite to the bridge endpoint (B-leg) with SDP containing only the codec negotiated for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only with IVR negotiated codec. I am having one issue here for example. Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated with G729 and starts playing IVR with G729 and answered(183) to originator with SDP containing only G729. And Invited to termination endpoint with SDP having only G729 and termination gateway having support of only G711, and it is rejecting call after receiving invite with only G729. It should be work like after IVR , invite can be having sdp with all the originator supported codecs and if termination (b-leg) having different priority codecs then it can send re-Invite sanding the same. I have observed one issue in freeswitch Jeera ( https://freeswitch.org/jira/browse/FS-880) , but there is no solution in that page. Do we have any solution for this right now ? Your answers will be very much helpful for me. I appreciate if you can quick response or give some solution for this. Thanks in advance. Thanks & Regards, GISR.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/cf8fb334/attachment.html From myforums.indra at gmail.com Thu Jan 8 10:23:49 2015 From: myforums.indra at gmail.com (indra sena) Date: Thu, 8 Jan 2015 12:53:49 +0530 Subject: [Freeswitch-users] Fwd: OverTime RAM usage keep inceasing In-Reply-To: References: Message-ID: Hi Vitalie, Thanks for your suggestion. I was checking with cache/buffers and mysql alloted RAM memory. Now it is ok. I have one more concern as follows. I have observed in freeswitch that, In case of IVR scenario prior to the bridge FreeSwitch plays IVR with the 1st priority codec then invite to the bridge endpoint (B-leg) with SDP containing only the codec negotiated for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only with IVR negotiated codec. I am having one issue here for example. Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated with G729 and starts playing IVR with G729 and answered(183) to originator with SDP containing only G729. And Invited to termination endpoint with SDP having only G729 and termination gateway having support of only G711, and it is rejecting call after receiving invite with only G729. It should be work like after IVR , invite can be having sdp with all the originator supported codecs and if termination (b-leg) having different priority codecs then it can send re-Invite sanding the same. I have observed one issue in freeswitch Jeera ( https://freeswitch.org/jira/browse/FS-880) , but there is no solution in that page. Do we have any solution for this right now ? Your answers will be very much helpful for me. I appreciate if you can quick response or give some solution for this. Thanks in advance. Thanks & Regards, GISR.. On Sun, Nov 23, 2014 at 1:02 AM, Vitalie Colosov wrote: > Are you checking total RAM utilization or without cache/buffers? > > Can you run command "free" and paste the output? > > 2014-11-20 23:24 GMT-08:00 indra sena : > >> >> Hi Team, >> >> I have observed that overtime RAM utilization/usage is keep increasing >> maybe due to memory leaks ? >> >> when I put load initially it starts with 5GB/12GM RAM, I have kept for 15 >> hours load test and oberserved now it is using almost 11GB/12GB starts swap >> also increasing now. >> >> >> Do you have config changes to fix or to take as work around ? >> How can we reduce RAM utilization overtime ? >> >> >> Could you please provide some solution for this ? >> >> Thanks & Regards, >> Indra. >> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/7cd75b45/attachment.html From telishisheer at gmail.com Thu Jan 8 13:52:54 2015 From: telishisheer at gmail.com (Shisheer Teli) Date: Thu, 8 Jan 2015 16:22:54 +0530 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: i am using FreeSWITCH Version 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 03:40:22Z 64bit) and still video call disconnect after 30 seconds Regards, Shisheer T On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Try latest master or 1.4.15 > > > On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli > wrote: > >> Hi Team, >> >> I don't know what happen , but when I start video call it disconnected >> after every 30 seconds. >> >> e.g. >> x-lite to x-lite call : video call disconnect after 30 seconds >> >> X-lite to Zoiper : video call continue, but no video sending. >> >> >> Regards >> Shisheer T >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > 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 > -- Regards, Shisheer Teli Phone: +91-022 2278 2519 / 2121 shisheer at tifr.res.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/dac620e9/attachment-0001.html From ilya at e-2-m.mobi Thu Jan 8 13:19:24 2015 From: ilya at e-2-m.mobi (ilyakn) Date: Thu, 8 Jan 2015 03:19:24 -0700 (MST) Subject: [Freeswitch-users] 488 Not Acceptable Media during blind call transfer in bypass media mode Message-ID: <1420712364644-7596138.post@n2.nabble.com> Hi All, I've run into a problem during call transfer. I have a CISCO switch calling Freeswitch that set bypass_media to true and makes a bridge to IVR platform (its actually a VXML browser). When IVR platform then tries to transfer a call using REFER the Freeswitch sends INVITE to the gateway in order to perform call transfer but Media part inside SDP is empty - as a result gateway returns 488 Not Acceptable Media and transfer fails. Is it possoble to make Freeswitch to send meaningful SDP inside INVITE? I would expect it to copy SDP from original incoming call - but this does not happen. Thanks, Ilya -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/488-Not-Acceptable-Media-during-blind-call-transfer-in-bypass-media-mode-tp7596138.html Sent from the freeswitch-users mailing list archive at Nabble.com. From GeorgePhelps at gfphelps.com Thu Jan 8 16:47:42 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Thu, 8 Jan 2015 08:47:42 -0500 Subject: [Freeswitch-users] Register->From->Caller ID In-Reply-To: References: <0aa001d02a04$620588b0$26109a10$@gfphelps.com> <0ad701d02a18$aba19720$02e4c560$@gfphelps.com> <0afb01d02a1e$fe442a50$facc7ef0$@gfphelps.com> Message-ID: <0cac01d02b49$ad7ea6f0$087bf4d0$@gfphelps.com> I have (temporarily) given up on trying to get overriding the outgoing Caller ID with either Caller ID-pid or Caller ID-rpid to work. I believe that Freeswitch is creating the correct pid and rpid SIP headers, but overriding the outgoing Caller ID is not working with my VoIP service provider. So going back to my Caller ID-from testing? In my gateway definition, I am now setting: Not quite sure why there are three similar, ?caller id? parameters to be set? Not sure if I am supposed to set one or all of them? And in my dialplan, I am now setting, even though they do not appear to be used: I see this SIP trace ? which is not overriding the outgoing Caller ID: INVITE sip:770XXXXXXX at 172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 50.160.141.159:43998;rport;branch=z9hG4bKPjXhLWMy2C.1IlJT7rpdMR.qay1IM-xAEk Max-Forwards: 70 From: ;tag=NQPRwYaq4mtxQmJlgV.3koDLn4UPBcqy To: sip:770XXXXXXX at 172.31.33.109 Contact: Call-ID: TRQEREtaaVcVT3z4fUIa4s1z5jjegT1. CSeq: 23043 INVITE I expected to see something like what is shown below (but not sure if that would work either), i.e., with the additional "+1404XXXXXXX" field. Or maybe, this: ? INVITE sip:770XXXXXXX at 172.31.33.109 SIP/2.0 Via: SIP/2.0/UDP 50.160.141.159:43998;rport;branch=z9hG4bKPjXhLWMy2C.1IlJT7rpdMR.qay1IM-xAEk Max-Forwards: 70 From: "+1404XXXXXXX" ;tag=NQPRwYaq4mtxQmJlgV.3koDLn4UPBcqy To: sip:770XXXXXXX at 172.31.33.109 Contact: Call-ID: TRQEREtaaVcVT3z4fUIa4s1z5jjegT1. CSeq: 23043 INVITE What should I see in my SIP trace (in the From header) for Caller ID ?From? support to work? And, what Freeswitch configuration do I change to make the From header match what is expected? The requirement, in Asterisk configuration terms, from my VoIP service provider is ?to set fromuser=+1XXXXXXXXXX?. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, January 07, 2015 9:50 AM To: FreeSWITCH Users Help; FreeSWITCH Docs Team Subject: Re: [Freeswitch-users] Register->From->Caller ID https://wiki.freeswitch.org/wiki/Variable_sip_cid_type https://wiki.freeswitch.org/wiki/Variable_effective_caller_id_name https://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number There are usually only three ways to send CID, From, RPID, PID Those links should give you an idea of what you need to try. On Tue, Jan 6, 2015 at 8:09 PM, George F. Phelps wrote: I wish I could, but no. I was previously using pbxes.com and overriding the outgoing caller ID worked. I believe pbxes.com is Asterisk based? Anyway, I don?t have a SIP trace from pbxes.com. Sorry. For what it?s worth, my pbxes.com trunk configuration, using their GUI, was: username=?1404XXXXXXX? password=?password:username? SIP-proxy=?66.33.147.150? And my outgoing caller ID would be set to ?1404XXXXXXX?? 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, January 06, 2015 8:43 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Register->From->Caller ID Having it on a register won't affect the CID, Can you show me the INVITE packet that Asterisk sends that works for you? On Tue, Jan 6, 2015 at 7:24 PM, George F. Phelps wrote: Brian West, I?m trying different tests to find a way to set the outgoing caller ID when using switch2voip.us. I?m attempting to emulate the results of their Asterisk configuration example (with Freeswitch configuration). They told me just to set ?fromuser? as shown below, in the SIP REGISTRATION message. [Switch2Voip] username={USERNAME} type=peer secret={PASSWORD} progressinband=never port=5060 nat=auto insecure=very ignoresdpversion=yes host=sip.switch2voip.us dtmfmode=rfc2833 context=from-trunk canreinvite=no allow=g729&g711&g723 fromuser=+{CALLER ID} Taken from this URL: http://switch2voip.us/index.php/customer-support/byod/asterisk-sip-trunk-configuration I?m not convinced that they want to exactly see? From: ?1404XXXXXXX? >;tag=FQag8jSNpac9c ?but I wanted to try. I can originate calls, just not override the outgoing caller ID. RFC 3261 does allow for an optional display-name field, along with the required URI field, in the From header. From RFC 3261: From: "Bob" >;tag=a48s From: sip:+12125551212 @phone2net.com;tag=887s From: Anonymous >;tag=hyh8 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, January 06, 2015 7:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Register->From->Caller ID gateway->register_from = switch_core_sprintf(gateway->pool, " >", from_user, !zstr(from_domain) ? from_domain : proxy); This is how the from is built in the code. Do you have a use case that Requires it? I can't think of a reason it would matter. On Tue, Jan 6, 2015 at 4:59 PM, George F. Phelps wrote: My current gateway REGISTER packet has a certain FROM header, as shown below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: >;tag=FQag8jSNpac9c To: > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: . . . What do I set/change in my gateway configuration to add a character string ?1404XXXXXXX? to the FROM header? As shown in red below: REGISTER sip:66.33.147.150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 54.174.255.168:5080;rport;branch=z9hG4bKKQayX8pZ5yctp Max-Forwards: 70 From: ?1404XXXXXXX? >;tag=FQag8jSNpac9c To: > Call-ID: d3cd81b1-c9e2-4401-8126-f4e545e3ebce CSeq: 69955649 REGISTER Contact: . . . I am already setting the following values (in my gateway definition): 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 -- 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 -- 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/20150108/d5df41c0/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/20150108/d5df41c0/attachment-0001.bin From brian at freeswitch.org Thu Jan 8 17:36:29 2015 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2015 08:36:29 -0600 Subject: [Freeswitch-users] 488 Not Acceptable Media during blind call transfer in bypass media mode In-Reply-To: <1420712364644-7596138.post@n2.nabble.com> References: <1420712364644-7596138.post@n2.nabble.com> Message-ID: Not something that can be answered with ease, No logs, no info on what rev of FreeSWITCH you're running. No info about endpoints and capabilities ... but if you're in bypass that would usually mean that your trying to bridge two endpoints that don't/can't speak the same codec. On Thu, Jan 8, 2015 at 4:19 AM, ilyakn wrote: > Hi All, > I've run into a problem during call transfer. > I have a CISCO switch calling Freeswitch that set bypass_media to true and > makes a bridge to IVR platform (its actually a VXML browser). > When IVR platform then tries to transfer a call using REFER the Freeswitch > sends INVITE to the gateway in order to perform call transfer but Media > part > inside SDP is empty - as a result gateway returns 488 Not Acceptable Media > and transfer fails. > Is it possoble to make Freeswitch to send meaningful SDP inside INVITE? I > would expect it to copy SDP from original incoming call - but this does not > happen. > > Thanks, > Ilya > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/488-Not-Acceptable-Media-during-blind-call-transfer-in-bypass-media-mode-tp7596138.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > 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/20150108/cf82b05a/attachment.html From brian at freeswitch.org Thu Jan 8 17:42:15 2015 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2015 08:42:15 -0600 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: What does the sip traffic say? On Thu, Jan 8, 2015 at 4:52 AM, Shisheer Teli wrote: > i am using FreeSWITCH Version > 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 > 03:40:22Z 64bit) > > and still video call disconnect after 30 seconds > > Regards, > Shisheer T > > On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Try latest master or 1.4.15 >> >> >> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >> wrote: >> >>> Hi Team, >>> >>> I don't know what happen , but when I start video call it disconnected >>> after every 30 seconds. >>> >>> e.g. >>> x-lite to x-lite call : video call disconnect after 30 seconds >>> >>> X-lite to Zoiper : video call continue, but no video sending. >>> >>> >>> Regards >>> Shisheer T >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Regards, > Shisheer Teli > Phone: +91-022 2278 2519 / 2121 > shisheer at tifr.res.in > > _________________________________________________________________________ > 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/20150108/d20de50b/attachment.html From aqsyounas at gmail.com Thu Jan 8 17:47:02 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 8 Jan 2015 19:47:02 +0500 Subject: [Freeswitch-users] how can i do stream piping in freeswitch In-Reply-To: <54ADDBF2.50106@mst.edu> References: <54ADDBF2.50106@mst.edu> Message-ID: Thanks for your suggestions. It really helped. But i have few questions. What if i have multiple streams actually about 400 streams and a lot of users who can switch between different streams, if i put all stream in a file then users will not be able to switch between desired streams. And if i create separate folder each containing a file with single stream then there will be a lots of directories and for every directory and also i have to manually enter its name in local_stream.conf.xml Because i see this setup suitable for only single stream or in a scenario where users don't want to switch between desired streams by pressing extension. Is there any other way to do so.? Or how can i improve this? Really thankful for your help. Regards. On 8 January 2015 at 06:22, Nathan Neulinger wrote: > Look at the Shout stream as MOH section of: > > https://wiki.freeswitch.org/wiki/Mod_shout > > You have to define the stream as a local_stream and then reference it via > local_stream://moh/whatever > > I use this to stream a local campus radio station via mp3/icecast as MOH > stream. > > -- Nathan > > On 01/07/2015 03:21 PM, Aqs Younas wrote: > > Currently i am playing a stream with mod_shout and this is my default > xml. > > > > > > > > > > > > > > > > > > > > > > Every time a user makes a call for stream it opens a separate connection > with the stream provider. If 100 users dials > > this number, there would be 100 connections with stream provider > listening to same stream, means more rtp packets > > containing same data for different users. > > > > What i want is, if a user is listening to stream then other users must > share the same listening connection that the > > first user is opened, instead of creating a separate connection with > stream provider for same stream. > > > > Someone told me this is possible in asterisk, so there must be a way in > freeswitch. > > > > How can i do this.? > > Any help would be much appreciated. > > > > Thanks > > > > > > > > _________________________________________________________________________ > > 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 > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/11a782c4/attachment-0001.html From telishisheer at gmail.com Thu Jan 8 17:52:23 2015 From: telishisheer at gmail.com (Shisheer Teli) Date: Thu, 8 Jan 2015 20:22:23 +0530 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: Kindly find the attached log details .. Regards, Shisheer T On Thu, Jan 8, 2015 at 8:12 PM, Brian West wrote: > What does the sip traffic say? > > On Thu, Jan 8, 2015 at 4:52 AM, Shisheer Teli > wrote: > >> i am using FreeSWITCH Version >> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >> 03:40:22Z 64bit) >> >> and still video call disconnect after 30 seconds >> >> Regards, >> Shisheer T >> >> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Try latest master or 1.4.15 >>> >>> >>> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >>> wrote: >>> >>>> Hi Team, >>>> >>>> I don't know what happen , but when I start video call it disconnected >>>> after every 30 seconds. >>>> >>>> e.g. >>>> x-lite to x-lite call : video call disconnect after 30 seconds >>>> >>>> X-lite to Zoiper : video call continue, but no video sending. >>>> >>>> >>>> Regards >>>> Shisheer T >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>> >>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>> http://twitter.com/FreeSWITCH >>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> Regards, >> Shisheer Teli >> Phone: +91-022 2278 2519 / 2121 >> shisheer at tifr.res.in >> >> _________________________________________________________________________ >> 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 > -- Regards, Shisheer Teli Phone: +91-022 2278 2519 / 2121 shisheer at tifr.res.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/b73abff7/attachment.html -------------- next part -------------- 2015-01-08 18:52:20.978885 [INFO] mod_dialplan_xml.c:635 Processing 1011 <1011>->1003 in context default 2015-01-08 18:52:20.978885 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2015-01-08 18:52:20.978885 [CRIT] mod_dptools.c:1628 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2015-01-08 18:52:20.978885 [CRIT] mod_dptools.c:1628 Once changed type 'reloadxml' at the console. 2015-01-08 18:52:20.978885 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2015-01-08 18:52:31.018880 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *1 execute_extension::dx XML features 2015-01-08 18:52:31.018880 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1011.2015-01-08-18-52-31.wav 2015-01-08 18:52:31.018880 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *3 execute_extension::cf XML features 2015-01-08 18:52:31.018880 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *4 execute_extension::att_xfer XML features 2015-01-08 18:52:31.018880 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:1003 at 10.1.1.3:30372 [661ff262-9739-11e4-97e9-3136adddb849] 2015-01-08 18:52:31.178927 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:1003 at 10.1.1.3:30372! 2015-01-08 18:52:31.178927 [INFO] switch_ivr_originate.c:1192 Sending early media 2015-01-08 18:52:31.178927 [NOTICE] switch_core_media.c:4405 sofia/internal/1011 at 10.1.1.1 Starting Video thread 2015-01-08 18:52:31.178927 [INFO] switch_core_media.c:5723 Activating VIDEO RTCP PORT 0 mux -1 2015-01-08 18:52:31.178927 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1011 at 10.1.1.1! 2015-01-08 18:52:37.858888 [NOTICE] switch_core_media.c:4405 sofia/internal/sip:1003 at 10.1.1.3:30372 Starting Video thread 2015-01-08 18:52:37.858888 [INFO] switch_core_media.c:5723 Activating VIDEO RTCP PORT 0 mux -1 2015-01-08 18:52:37.858888 [NOTICE] sofia.c:7475 Channel [sofia/internal/sip:1003 at 10.1.1.3:30372] has been answered 2015-01-08 18:52:37.878892 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/1011 at 10.1.1.1] has been answered 2015-01-08 18:52:37.978889 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-08-00 13:22:37.985145] SSRC[719723942]RTT[21600.327164] A[52296754] - DLSR[910709816] - LSR[2020955193] 2015-01-08 18:53:09.898888 [NOTICE] sofia.c:7530 Hangup sofia/internal/1011 at 10.1.1.1 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2015-01-08 18:53:09.918887 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/internal/sip:1003 at 10.1.1.3:30372 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-08 18:53:09.918887 [NOTICE] switch_core_session.c:1633 Session 37 (sofia/internal/1011 at 10.1.1.1) Ended 2015-01-08 18:53:09.918887 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/1011 at 10.1.1.1 [CS_DESTROY] 2015-01-08 18:53:09.918887 [NOTICE] switch_core_session.c:1633 Session 38 (sofia/internal/sip:1003 at 10.1.1.3:30372) Ended 2015-01-08 18:53:09.918887 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:1003 at 10.1.1.3:30372 [CS_DESTROY] From brian at freeswitch.org Thu Jan 8 17:55:34 2015 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2015 08:55:34 -0600 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: do 'sofia global siptrace on', so we can see the sip trace. On Thu, Jan 8, 2015 at 8:52 AM, Shisheer Teli wrote: > Kindly find the attached log details .. > > Regards, > Shisheer T > > On Thu, Jan 8, 2015 at 8:12 PM, Brian West wrote: > >> What does the sip traffic say? >> >> On Thu, Jan 8, 2015 at 4:52 AM, Shisheer Teli >> wrote: >> >>> i am using FreeSWITCH Version >>> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >>> 03:40:22Z 64bit) >>> >>> and still video call disconnect after 30 seconds >>> >>> Regards, >>> Shisheer T >>> >>> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> Try latest master or 1.4.15 >>>> >>>> >>>> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >>>> wrote: >>>> >>>>> Hi Team, >>>>> >>>>> I don't know what happen , but when I start video call it disconnected >>>>> after every 30 seconds. >>>>> >>>>> e.g. >>>>> x-lite to x-lite call : video call disconnect after 30 seconds >>>>> >>>>> X-lite to Zoiper : video call continue, but no video sending. >>>>> >>>>> >>>>> Regards >>>>> Shisheer T >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>> >>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>> http://twitter.com/FreeSWITCH >>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>> >>> >>> >>> -- >>> Regards, >>> Shisheer Teli >>> Phone: +91-022 2278 2519 / 2121 >>> shisheer at tifr.res.in >>> >>> _________________________________________________________________________ >>> 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 >> > > > > -- > Regards, > Shisheer Teli > Phone: +91-022 2278 2519 / 2121 > shisheer at tifr.res.in > > _________________________________________________________________________ > 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/20150108/4cd4acf0/attachment-0001.html From brian at freeswitch.org Thu Jan 8 17:56:06 2015 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2015 08:56:06 -0600 Subject: [Freeswitch-users] how can i do stream piping in freeswitch In-Reply-To: References: <54ADDBF2.50106@mst.edu> Message-ID: You're going to need 400 local streams, or to re-think you approach. On Thu, Jan 8, 2015 at 8:47 AM, Aqs Younas wrote: > Thanks for your suggestions. It really helped. > > But i have few questions. > > What if i have multiple streams actually about 400 streams and a lot of > users who can switch between different streams, if i put all stream in a > file then users will not be able to switch between desired streams. > > And if i create separate folder each containing a file with single stream > then there will be a lots of directories and for every directory and also i > have to manually enter its name in local_stream.conf.xml > > Because i see this setup suitable for only single stream or in a scenario > where users don't want to switch between desired streams by pressing > extension. > > Is there any other way to do so.? Or how can i improve this? > > Really thankful for your help. > > Regards. > > On 8 January 2015 at 06:22, Nathan Neulinger wrote: > >> Look at the Shout stream as MOH section of: >> >> https://wiki.freeswitch.org/wiki/Mod_shout >> >> You have to define the stream as a local_stream and then reference it via >> local_stream://moh/whatever >> >> I use this to stream a local campus radio station via mp3/icecast as MOH >> stream. >> >> -- Nathan >> >> On 01/07/2015 03:21 PM, Aqs Younas wrote: >> > Currently i am playing a stream with mod_shout and this is my default >> xml. >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > Every time a user makes a call for stream it opens a separate >> connection with the stream provider. If 100 users dials >> > this number, there would be 100 connections with stream provider >> listening to same stream, means more rtp packets >> > containing same data for different users. >> > >> > What i want is, if a user is listening to stream then other users must >> share the same listening connection that the >> > first user is opened, instead of creating a separate connection with >> stream provider for same stream. >> > >> > Someone told me this is possible in asterisk, so there must be a way in >> freeswitch. >> > >> > How can i do this.? >> > Any help would be much appreciated. >> > >> > Thanks >> > >> > >> > >> > >> _________________________________________________________________________ >> > 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 >> > >> >> -- >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> 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/20150108/f75144be/attachment.html From avi at avimarcus.net Thu Jan 8 18:07:29 2015 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 8 Jan 2015 15:07:29 +0000 Subject: [Freeswitch-users] how can i do stream piping in freeswitch In-Reply-To: References: <54ADDBF2.50106@mst.edu> Message-ID: <0000014aca160ad0-8c5bd4dd-a7eb-47b9-a9da-03dc36330d3c-000000@email.amazonses.com> You set up.. all 400.. FreeSWITCH mod_shout streams as an MOH stream. Unfortunately, all are always running even if nobody is listening to it. Then, it sounded like you already have a switching mechanism. Instead of switching them to a playback of an http:....mp3, you switch them to a playback of a localstream -Avi On Thu, Jan 8, 2015 at 4:47 PM, Aqs Younas wrote: > Thanks for your suggestions. It really helped. > > But i have few questions. > > What if i have multiple streams actually about 400 streams and a lot of > users who can switch between different streams, if i put all stream in a > file then users will not be able to switch between desired streams. > > And if i create separate folder each containing a file with single stream > then there will be a lots of directories and for every directory and also i > have to manually enter its name in local_stream.conf.xml > > Because i see this setup suitable for only single stream or in a scenario > where users don't want to switch between desired streams by pressing > extension. > > Is there any other way to do so.? Or how can i improve this? > > Really thankful for your help. > > Regards. > > On 8 January 2015 at 06:22, Nathan Neulinger wrote: > >> Look at the Shout stream as MOH section of: >> >> https://wiki.freeswitch.org/wiki/Mod_shout >> >> You have to define the stream as a local_stream and then reference it via >> local_stream://moh/whatever >> >> I use this to stream a local campus radio station via mp3/icecast as MOH >> stream. >> >> -- Nathan >> >> On 01/07/2015 03:21 PM, Aqs Younas wrote: >> > Currently i am playing a stream with mod_shout and this is my default >> xml. >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > Every time a user makes a call for stream it opens a separate >> connection with the stream provider. If 100 users dials >> > this number, there would be 100 connections with stream provider >> listening to same stream, means more rtp packets >> > containing same data for different users. >> > >> > What i want is, if a user is listening to stream then other users must >> share the same listening connection that the >> > first user is opened, instead of creating a separate connection with >> stream provider for same stream. >> > >> > Someone told me this is possible in asterisk, so there must be a way in >> freeswitch. >> > >> > How can i do this.? >> > Any help would be much appreciated. >> > >> > Thanks >> > >> > >> > >> > >> _________________________________________________________________________ >> > 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 >> > >> >> -- >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/aedf4339/attachment-0001.html From nneul at mst.edu Thu Jan 8 18:12:58 2015 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 08 Jan 2015 09:12:58 -0600 Subject: [Freeswitch-users] how can i do stream piping in freeswitch In-Reply-To: <0000014aca160ad0-8c5bd4dd-a7eb-47b9-a9da-03dc36330d3c-000000@email.amazonses.com> References: <54ADDBF2.50106@mst.edu> <0000014aca160ad0-8c5bd4dd-a7eb-47b9-a9da-03dc36330d3c-000000@email.amazonses.com> Message-ID: <54AE9E7A.3020505@mst.edu> Only other approach I can think of would be to try and do some sort of reverse reflector. i.e. the issue is that you're having 100 connections to the external stream for 100 listeners If you could get the 100 connections to a LOCAL server, to where you'd still have the duplication locally on your internal network, but not 100 separate connections to the remote stream provider - that might address it. At that point, you could have each caller do the individual playback - getting the local duplication. I don't know that anything exists to do that function though. -- Nathan On 01/08/2015 09:07 AM, Avi Marcus wrote: > You set up.. all 400.. FreeSWITCH mod_shout streams as an MOH stream. Unfortunately, all are always running even if > nobody is listening to it. > Then, it sounded like you already have a switching mechanism. Instead of switching them to a playback of an > http:....mp3, you switch them to a playback of a localstream > > -Avi > > On Thu, Jan 8, 2015 at 4:47 PM, Aqs Younas > wrote: > > Thanks for your suggestions. It really helped. > > But i have few questions. > > What if i have multiple streams actually about 400 streams and a lot of users who can switch between different > streams, if i put all stream in a file then users will not be able to switch between desired streams. > > And if i create separate folder each containing a file with single stream then there will be a lots of directories > and for every directory and also i have to manually enter its name in local_stream.conf.xml > > Because i see this setup suitable for only single stream or in a scenario where users don't want to switch between > desired streams by pressing extension. > > Is there any other way to do so.? Or how can i improve this? > > Really thankful for your help. > > Regards. > > On 8 January 2015 at 06:22, Nathan Neulinger > wrote: > > Look at the Shout stream as MOH section of: > > https://wiki.freeswitch.org/wiki/Mod_shout > > You have to define the stream as a local_stream and then reference it via local_stream://moh/whatever > > I use this to stream a local campus radio station via mp3/icecast as MOH stream. > > -- Nathan > > On 01/07/2015 03:21 PM, Aqs Younas wrote: > > Currently i am playing a stream with mod_shout and this is my default xml. > > > > > > > > > > > > > > > > > > > > > > Every time a user makes a call for stream it opens a separate connection with the stream provider. If 100 users dials > > this number, there would be 100 connections with stream provider listening to same stream, means more rtp packets > > containing same data for different users. > > > > What i want is, if a user is listening to stream then other users must share the same listening connection that the > > first user is opened, instead of creating a separate connection with stream provider for same stream. > > > > Someone told me this is possible in asterisk, so there must be a way in freeswitch. > > > > How can i do this.? > > Any help would be much appreciated. > > > > Thanks > > > > > > > > _________________________________________________________________________ > > 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 > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > 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 > > > > > _________________________________________________________________________ > 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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From shisheer at tifr.res.in Thu Jan 8 18:14:59 2015 From: shisheer at tifr.res.in (Shisheer Teli) Date: Thu, 8 Jan 2015 20:44:59 +0530 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: Kindly find the attached "sofia global siptrace on" logs .. On Thu, Jan 8, 2015 at 8:25 PM, Brian West wrote: > do 'sofia global siptrace on', so we can see the sip trace. > > On Thu, Jan 8, 2015 at 8:52 AM, Shisheer Teli > wrote: > >> Kindly find the attached log details .. >> >> Regards, >> Shisheer T >> >> On Thu, Jan 8, 2015 at 8:12 PM, Brian West wrote: >> >>> What does the sip traffic say? >>> >>> On Thu, Jan 8, 2015 at 4:52 AM, Shisheer Teli >>> wrote: >>> >>>> i am using FreeSWITCH Version >>>> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >>>> 03:40:22Z 64bit) >>>> >>>> and still video call disconnect after 30 seconds >>>> >>>> Regards, >>>> Shisheer T >>>> >>>> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> Try latest master or 1.4.15 >>>>> >>>>> >>>>> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >>>>> wrote: >>>>> >>>>>> Hi Team, >>>>>> >>>>>> I don't know what happen , but when I start video call it >>>>>> disconnected after every 30 seconds. >>>>>> >>>>>> e.g. >>>>>> x-lite to x-lite call : video call disconnect after 30 seconds >>>>>> >>>>>> X-lite to Zoiper : video call continue, but no video sending. >>>>>> >>>>>> >>>>>> Regards >>>>>> Shisheer T >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>>> >>>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>>> http://twitter.com/FreeSWITCH >>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>>> * >>>>> >>>>> ClueCon Weekly Development Call >>>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> Shisheer Teli >>>> Phone: +91-022 2278 2519 / 2121 >>>> shisheer at tifr.res.in >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> >> >> -- >> Regards, >> Shisheer Teli >> Phone: +91-022 2278 2519 / 2121 >> shisheer at tifr.res.in >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/0961a8e6/attachment-0001.html -------------- next part -------------- freeswitch at internal> sofia global siptrace on +OK Global siptrace on freeswitch at internal> recv 1050 bytes from udp/[10.1.1.3]:21194 at 20:37:10.418033: ------------------------------------------------------------------------ INVITE sip:1001 at 10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-0e061d6b0960fa75-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4.7.0 73589-abcb99a9-W6.1 Content-Length: 508 v=0 o=- 13065203331951171 1 IN IP4 10.1.1.3 s=X-Lite release 4.7.0 stamp 73589 c=IN IP4 10.1.1.3 t=0 0 m=audio 56818 RTP/AVP 125 100 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 51488 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli a=sendrecv ------------------------------------------------------------------------ send 367 bytes to udp/[10.1.1.3]:21194 at 20:37:10.418426: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-0e061d6b0960fa75-1---d8754z-;rport=21194 From: ;tag=fc96497b To: Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:10.398899 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1004 at 10.1.1.1 [04ef4178-9748-11e4-981f-3136adddb849] 2015-01-08 20:37:10.398899 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.398899 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.398899 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004 at 10.1.1.1) Running State Change CS_NEW 2015-01-08 20:37:10.398899 [DEBUG] sofia.c:8834 sofia/internal/1004 at 10.1.1.1 receiving invite from 10.1.1.3:21194 version: 1.5.15b git 1ed290e 2015-01-08 03:40:22Z 64bit 2015-01-08 20:37:10.418893 [DEBUG] sofia.c:9001 IP 10.1.1.3 Rejected by acl "domains". Falling back to Digest auth. 2015-01-08 20:37:10.418893 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1004 at 10.1.1.1) State NEW send 871 bytes to udp/[10.1.1.3]:21194 at 20:37:10.419509: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-0e061d6b0960fa75-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=tSSB99r2Fypaa Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="10.1.1.1", nonce="04ef5302-9748-11e4-9820-3136adddb849", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:10.418893 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.418893 [DEBUG] sofia.c:2067 detaching session 04ef4178-9748-11e4-981f-3136adddb849 recv 330 bytes from udp/[10.1.1.3]:21194 at 20:37:10.423840: ------------------------------------------------------------------------ ACK sip:1001 at 10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-0e061d6b0960fa75-1---d8754z-;rport Max-Forwards: 70 To: ;tag=tSSB99r2Fypaa From: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 1312 bytes from udp/[10.1.1.3]:21194 at 20:37:10.429397: ------------------------------------------------------------------------ INVITE sip:1001 at 10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1004",realm="10.1.1.1",nonce="04ef5302-9748-11e4-9820-3136adddb849",uri="sip:1001 at 10.1.1.1",response="481a9aca4f69b1b4fb3ea4426111efa4",cnonce="5dbdd76fed771a60c49b5a60917f09a4",nc=00000001,qop=auth,algorithm=MD5 Supported: replaces User-Agent: X-Lite 4.7.0 73589-abcb99a9-W6.1 Content-Length: 508 v=0 o=- 13065203331951171 1 IN IP4 10.1.1.3 s=X-Lite release 4.7.0 stamp 73589 c=IN IP4 10.1.1.3 t=0 0 m=audio 56818 RTP/AVP 125 100 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 51488 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli a=sendrecv ------------------------------------------------------------------------ send 367 bytes to udp/[10.1.1.3]:21194 at 20:37:10.429689: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:10.418893 [DEBUG] sofia.c:2175 Re-attaching to session 04ef4178-9748-11e4-981f-3136adddb849 2015-01-08 20:37:10.418893 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.418893 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:8834 sofia/internal/1004 at 10.1.1.1 receiving invite from 10.1.1.3:21194 version: 1.5.15b git 1ed290e 2015-01-08 03:40:22Z 64bit 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:9001 IP 10.1.1.3 Rejected by acl "domains". Falling back to Digest auth. 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:6614 Channel sofia/internal/1004 at 10.1.1.1 entering state [received][100] 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=- 13065203331951171 1 IN IP4 10.1.1.3 s=X-Lite release 4.7.0 stamp 73589 c=IN IP4 10.1.1.3 t=0 0 m=audio 56818 RTP/AVP 125 100 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 51488 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:6890 (sofia/internal/1004 at 10.1.1.1) State Change CS_NEW -> CS_INIT 2015-01-08 20:37:10.438894 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004 at 10.1.1.1) Running State Change CS_INIT 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1004 at 10.1.1.1) State INIT 2015-01-08 20:37:10.438894 [DEBUG] mod_sofia.c:87 sofia/internal/1004 at 10.1.1.1 SOFIA INIT 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1004 at 10.1.1.1 Standard INIT 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1004 at 10.1.1.1) State Change CS_INIT -> CS_ROUTING 2015-01-08 20:37:10.438894 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1004 at 10.1.1.1) State INIT going to sleep 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004 at 10.1.1.1) Running State Change CS_ROUTING 2015-01-08 20:37:10.438894 [DEBUG] switch_channel.c:2184 (sofia/internal/1004 at 10.1.1.1) Callstate Change DOWN -> RINGING 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1004 at 10.1.1.1) State ROUTING 2015-01-08 20:37:10.438894 [DEBUG] mod_sofia.c:123 sofia/internal/1004 at 10.1.1.1 SOFIA ROUTING 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1004 at 10.1.1.1 Standard ROUTING 2015-01-08 20:37:10.438894 [INFO] mod_dialplan_xml.c:635 Processing 1004 <1004>->1001 in context default Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->unloop] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->tod_example] continue=true Dialplan: sofia/internal/1004 at 10.1.1.1 Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1004 at 10.1.1.1 Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->redial] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->global] continue=true Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (PASS) [global] ${default_password}(1234) =~ /^1234$/ break=never Dialplan: sofia/internal/1004 at 10.1.1.1 Action log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) Dialplan: sofia/internal/1004 at 10.1.1.1 Action log(CRIT Open /usr/local/freeswitch/conf/vars.xml and change the default_password.) Dialplan: sofia/internal/1004 at 10.1.1.1 Action log(CRIT Once changed type 'reloadxml' at the console.) Dialplan: sofia/internal/1004 at 10.1.1.1 Action log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) Dialplan: sofia/internal/1004 at 10.1.1.1 Action sleep(10000) Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [global] ${rtp_has_crypto}() =~ /^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/ break=never Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [global] ${switch_r_sdp}(v=0 o=- 13065203331951171 1 IN IP4 10.1.1.3 s=X-Lite release 4.7.0 stamp 73589 c=IN IP4 10.1.1.3 t=0 0 m=audio 56818 RTP/AVP 125 100 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 51488 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never Dialplan: sofia/internal/1004 at 10.1.1.1 Absolute Condition [global] Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->call_return] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->del-group] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->add-group] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 Action export(dialed_extension=1001) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(ringback=${us-ring}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(call_timeout=30) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(continue_on_fail=true) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action answer() Dialplan: sofia/internal/1004 at 10.1.1.1 Action sleep(1000) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1004 at 10.1.1.1) State Change CS_ROUTING -> CS_EXECUTE 2015-01-08 20:37:10.438894 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1004 at 10.1.1.1) State ROUTING going to sleep 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004 at 10.1.1.1) Running State Change CS_EXECUTE 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1004 at 10.1.1.1) State EXECUTE 2015-01-08 20:37:10.438894 [DEBUG] mod_sofia.c:178 sofia/internal/1004 at 10.1.1.1 SOFIA EXECUTE 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1004 at 10.1.1.1 Standard EXECUTE EXECUTE sofia/internal/1004 at 10.1.1.1 log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) 2015-01-08 20:37:10.438894 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING EXECUTE sofia/internal/1004 at 10.1.1.1 log(CRIT Open /usr/local/freeswitch/conf/vars.xml and change the default_password.) 2015-01-08 20:37:10.438894 [CRIT] mod_dptools.c:1628 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. EXECUTE sofia/internal/1004 at 10.1.1.1 log(CRIT Once changed type 'reloadxml' at the console.) 2015-01-08 20:37:10.438894 [CRIT] mod_dptools.c:1628 Once changed type 'reloadxml' at the console. EXECUTE sofia/internal/1004 at 10.1.1.1 log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) 2015-01-08 20:37:10.438894 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING EXECUTE sofia/internal/1004 at 10.1.1.1 sleep(10000) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-spymap/1004/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial/1004/1001) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial/global/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 export(RFC2822_DATE=Thu, 08 Jan 2015 20:37:20 +0530) 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 08 Jan 2015 20:37:20 +0530] EXECUTE sofia/internal/1004 at 10.1.1.1 export(dialed_extension=1001) 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at 10.1.1.1 bind_meta_app(1 b s execute_extension::dx XML features) 2015-01-08 20:37:20.458892 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/1004 at 10.1.1.1 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1004.2015-01-08-20-37-20.wav) 2015-01-08 20:37:20.458892 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1004.2015-01-08-20-37-20.wav EXECUTE sofia/internal/1004 at 10.1.1.1 bind_meta_app(3 b s execute_extension::cf XML features) 2015-01-08 20:37:20.458892 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/1004 at 10.1.1.1 bind_meta_app(4 b s execute_extension::att_xfer XML features) 2015-01-08 20:37:20.458892 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/1004 at 10.1.1.1 set(ringback=%(2000,4000,440,480)) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/1004 at 10.1.1.1 set(transfer_ringback=local_stream://moh) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1004 at 10.1.1.1 set(call_timeout=30) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [call_timeout]=[30] EXECUTE sofia/internal/1004 at 10.1.1.1 set(hangup_after_bridge=true) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1004 at 10.1.1.1 set(continue_on_fail=true) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [continue_on_fail]=[true] EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-call_return/1001/1004) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial_ext/1001/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 set(called_party_callgroup=techsupport) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial_ext/techsupport/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial_ext/global/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial/techsupport/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 bridge(user/1001 at 10.1.1.1) 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1201 sofia/internal/1004 at 10.1.1.1 EXPORTING[export_vars] [RFC2822_DATE]=[Thu, 08 Jan 2015 20:37:20 +0530] to event 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1201 sofia/internal/1004 at 10.1.1.1 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2015-01-08 20:37:20.458892 [DEBUG] switch_ivr_originate.c:2103 Parsing global variables 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1201 sofia/internal/1004 at 10.1.1.1 EXPORTING[export_vars] [RFC2822_DATE]=[Thu, 08 Jan 2015 20:37:20 +0530] to event 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1201 sofia/internal/1004 at 10.1.1.1 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2015-01-08 20:37:20.458892 [DEBUG] switch_ivr_originate.c:2103 Parsing global variables 2015-01-08 20:37:20.458892 [DEBUG] switch_event.c:1688 Parsing variable [sip_invite_domain]=[10.1.1.1] 2015-01-08 20:37:20.458892 [DEBUG] switch_event.c:1688 Parsing variable [presence_id]=[1001 at 10.1.1.1] 2015-01-08 20:37:20.458892 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:1001 at 10.1.1.4:36762 [0aece904-9748-11e4-9842-3136adddb849] 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:4636 (sofia/internal/sip:1001 at 10.1.1.4:36762) State Change CS_NEW -> CS_INIT 2015-01-08 20:37:20.458892 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:4706 [zrtp_passthru] Setting a-leg inherit_codec=true 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:4709 [zrtp_passthru] Setting b-leg absolute_codec_string='opus at 48000h@20i at 2c,PCMU at 8000h@20i at 64000b,G722 at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b,H263 at 90000h@1c,H263-1998 at 90000h@1c' 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1001 at 10.1.1.4:36762) Running State Change CS_INIT 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/sip:1001 at 10.1.1.4:36762) State INIT 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:87 sofia/internal/sip:1001 at 10.1.1.4:36762 SOFIA INIT 2015-01-08 20:37:20.458892 [DEBUG] sofia_glue.c:1232 sofia/internal/sip:1001 at 10.1.1.4:36762 sending invite version: 1.5.15b git 1ed290e 2015-01-08 03:40:22Z 64bit Local SDP: v=0 o=FreeSWITCH 1420703660 1420703661 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 25980 RTP/AVP 125 0 9 8 101 13 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 30320 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:40 sofia/internal/sip:1001 at 10.1.1.4:36762 Standard INIT 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/sip:1001 at 10.1.1.4:36762) State Change CS_INIT -> CS_ROUTING 2015-01-08 20:37:20.458892 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/sip:1001 at 10.1.1.4:36762) State INIT going to sleep 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1001 at 10.1.1.4:36762) Running State Change CS_ROUTING send 1482 bytes to udp/[10.1.1.4]:36762 at 20:37:20.471070: ------------------------------------------------------------------------ INVITE sip:1001 at 10.1.1.4:36762;rinstance=18b07042dd52441e SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKXKKNgevFvQaUQ Max-Forwards: 69 From: "Extension 1004" ;tag=vBcXc0t99F3FH To: Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028756 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 465 X-FS-Support: update_display,send_info Remote-Party-ID: "Extension 1004" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420703660 1420703661 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 25980 RTP/AVP 125 0 9 8 101 13 a=rtpmap:125 opus/48000/2 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/sip:1001 at 10.1.1.4:36762) State ROUTING a=fmtp:125 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:123 sofia/internal/sip:1001 at 10.1.1.4:36762 SOFIA ROUTING a=fmtp:101 0-16 a=ptime:20 m=video 30320 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 2015-01-08 20:37:20.458892 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:1001 at 10.1.1.4:36762) State Change CS_ROUTING -> CS_CONSUME_MEDIA a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 ------------------------------------------------------------------------ 2015-01-08 20:37:20.458892 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/sip:1001 at 10.1.1.4:36762) State ROUTING going to sleep 2015-01-08 20:37:20.458892 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1001 at 10.1.1.4:36762) Running State Change CS_CONSUME_MEDIA 2015-01-08 20:37:20.458892 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1001 at 10.1.1.4:36762 entering state [calling][0] 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/sip:1001 at 10.1.1.4:36762) State CONSUME_MEDIA 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/sip:1001 at 10.1.1.4:36762) State CONSUME_MEDIA going to sleep recv 412 bytes from udp/[10.1.1.4]:36762 at 20:37:20.550219: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKXKKNgevFvQaUQ Contact: To: ;tag=7d1efd53 From: "Extension 1004";tag=vBcXc0t99F3FH Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028756 INVITE User-Agent: X-Lite release 4.7.1 stamp 74247 Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:20.538891 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.538891 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1001 at 10.1.1.4:36762 entering state [proceeding][180] 2015-01-08 20:37:20.538891 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:1001 at 10.1.1.4:36762! 2015-01-08 20:37:20.538891 [DEBUG] switch_channel.c:3277 (sofia/internal/sip:1001 at 10.1.1.4:36762) Callstate Change DOWN -> RINGING 2015-01-08 20:37:20.538891 [INFO] switch_ivr_originate.c:1192 Sending early media 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3543 Set telephone-event payload to 101 2015-01-08 20:37:20.538891 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:20.538891 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/1004 at 10.1.1.1 opus/48000 20 ms 960 samples 0 bits 1 channels 2015-01-08 20:37:20.538891 [DEBUG] switch_core_codec.c:111 sofia/internal/1004 at 10.1.1.1 Original read codec set to opus:116 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3890 Set 2833 dtmf send/recv payload to 101 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H261:31] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263:34] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263-1998:115] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4020 Video Codec Compare [H263-1998:115] +++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263-2000:121] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H264:97] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[VP8:99] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H261:31] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263:34] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4020 Video Codec Compare [H263:34] +++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263-1998:115] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263-2000:121] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H264:97] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[VP8:99] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:2293 Set VIDEO Codec sofia/internal/1004 at 10.1.1.1 H263-1998/90000 0 ms 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/internal/1004 at 10.1.1.1] 10.1.1.1 port 19038 -> 10.1.1.3 port 56818 codec: 125 ms: 20 2015-01-08 20:37:20.538891 [DEBUG] switch_rtp.c:3548 Starting timer [soft] 960 bytes per 20ms 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:5439 Set 2833 dtmf send payload to 101 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:5445 Set 2833 dtmf receive payload to 101 2015-01-08 20:37:20.538891 [DEBUG] switch_rtp.c:3557 Not using a timer 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:5643 VIDEO RTP [sofia/internal/1004 at 10.1.1.1] 10.1.1.3:29642->10.1.1.3:51488 codec: 115 ms: 0 [SUCCESS] 2015-01-08 20:37:20.538891 [NOTICE] switch_core_media.c:4405 sofia/internal/1004 at 10.1.1.1 Starting Video thread 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4329 sofia/internal/1004 at 10.1.1.1 Video thread started. Echo is on 2015-01-08 20:37:20.538891 [INFO] switch_core_media.c:5723 Activating VIDEO RTCP PORT 0 mux -1 2015-01-08 20:37:20.538891 [DEBUG] switch_rtp.c:3898 RTCP send rate is: 10000 and packet rate is: 90000 Remote Port: 51489 2015-01-08 20:37:20.538891 [DEBUG] switch_rtp.c:2354 Setting RTCP remote addr to 10.1.1.3:51489 2015-01-08 20:37:20.538891 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1004 at 10.1.1.1! 2015-01-08 20:37:20.538891 [DEBUG] switch_channel.c:3399 (sofia/internal/1004 at 10.1.1.1) Callstate Change RINGING -> EARLY 2015-01-08 20:37:20.538891 [DEBUG] mod_sofia.c:2268 Ring SDP: v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 2015-01-08 20:37:20.538891 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:20.538891 [DEBUG] switch_ivr_originate.c:1249 Raw Codec Activation Success L16 at 48000hz 1 channel 20ms 2015-01-08 20:37:20.538891 [DEBUG] switch_core_codec.c:221 sofia/internal/1004 at 10.1.1.1 Push codec L16:70 2015-01-08 20:37:20.538891 [DEBUG] switch_ivr_originate.c:1317 Play Ringback Tone [%(2000,4000,440,480)] send 1336 bytes to udp/[10.1.1.3]:21194 at 20:37:20.558402: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "1001" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ 2015-01-08 20:37:20.538891 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:20.558650: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ 2015-01-08 20:37:20.578891 [DEBUG] sofia.c:6614 Channel sofia/internal/1004 at 10.1.1.1 entering state [early][183] 2015-01-08 20:37:20.618894 [DEBUG] switch_rtp.c:5853 Correct ip/port confirmed. send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:21.559269: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 1033 bytes from udp/[10.1.1.4]:36762 at 20:37:22.558754: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKXKKNgevFvQaUQ Contact: To: ;tag=7d1efd53 From: "Extension 1004";tag=vBcXc0t99F3FH Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028756 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.7.1 stamp 74247 Content-Length: 478 v=0 o=- 13065203368350614 3 IN IP4 10.1.1.4 s=X-Lite release 4.7.1 stamp 74247 c=IN IP4 10.1.1.4 t=0 0 m=audio 62698 RTP/AVP 125 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 49936 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=sendrecv a=rtcp-fb:* nack pli ------------------------------------------------------------------------ 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.558880 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1001 at 10.1.1.4:36762 entering state [completing][200] 2015-01-08 20:37:22.558880 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=- 13065203368350614 3 IN IP4 10.1.1.4 s=X-Lite release 4.7.1 stamp 74247 c=IN IP4 10.1.1.4 t=0 0 m=audio 62698 RTP/AVP 125 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 49936 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp-fb:* nack pli send 402 bytes to udp/[10.1.1.4]:36762 at 20:37:22.560661: ------------------------------------------------------------------------ ACK sip:1001 at 10.1.1.4:36762 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKZ556K4Xpp9p0e Max-Forwards: 70 From: "Extension 1004" ;tag=vBcXc0t99F3FH To: ;tag=7d1efd53 Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028756 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.558880 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1001 at 10.1.1.4:36762 entering state [ready][200] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[opus:116:48000:20:0:2] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [opus:116:48000:20:0:2] ++++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:2] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:2] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:2] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3543 Set telephone-event payload to 101 2015-01-08 20:37:22.558880 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:22.558880 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/sip:1001 at 10.1.1.4:36762 opus/48000 20 ms 960 samples 0 bits 1 channels 2015-01-08 20:37:22.558880 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1001 at 10.1.1.4:36762 Original read codec set to opus:116 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3881 Set 2833 dtmf send payload to 101 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263:34] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4020 Video Codec Compare [H263:34] +++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263-1998:115] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263:34] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263-1998:115] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4020 Video Codec Compare [H263-1998:115] +++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:2293 Set VIDEO Codec sofia/internal/sip:1001 at 10.1.1.4:36762 H263/90000 0 ms 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/internal/sip:1001 at 10.1.1.4:36762] 10.1.1.1 port 25980 -> 10.1.1.4 port 62698 codec: 125 ms: 20 2015-01-08 20:37:22.558880 [DEBUG] switch_rtp.c:3548 Starting timer [soft] 960 bytes per 20ms 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:5439 Set 2833 dtmf send payload to 101 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:5445 Set 2833 dtmf receive payload to 101 2015-01-08 20:37:22.558880 [DEBUG] switch_rtp.c:3557 Not using a timer 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:5643 VIDEO RTP [sofia/internal/sip:1001 at 10.1.1.4:36762] 10.1.1.4:30320->10.1.1.4:49936 codec: 34 ms: 0 [SUCCESS] 2015-01-08 20:37:22.558880 [NOTICE] switch_core_media.c:4405 sofia/internal/sip:1001 at 10.1.1.4:36762 Starting Video thread 2015-01-08 20:37:22.558880 [INFO] switch_core_media.c:5723 Activating VIDEO RTCP PORT 0 mux -1 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4329 sofia/internal/sip:1001 at 10.1.1.4:36762 Video thread started. Echo is off 2015-01-08 20:37:22.558880 [DEBUG] switch_rtp.c:3898 RTCP send rate is: 10000 and packet rate is: 90000 Remote Port: 49937 2015-01-08 20:37:22.558880 [DEBUG] switch_rtp.c:2354 Setting RTCP remote addr to 10.1.1.4:49937 2015-01-08 20:37:22.558880 [DEBUG] switch_channel.c:3635 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:22.558880 [NOTICE] sofia.c:7475 Channel [sofia/internal/sip:1001 at 10.1.1.4:36762] has been answered 2015-01-08 20:37:22.558880 [DEBUG] switch_channel.c:3689 (sofia/internal/sip:1001 at 10.1.1.4:36762) Callstate Change RINGING -> ACTIVE send 850 bytes to udp/[10.1.1.4]:36762 at 20:37:22.566368: ------------------------------------------------------------------------ INFO sip:1001 at 10.1.1.4:36762 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bK0eZZNZetKjDKa Max-Forwards: 70 From: "Extension 1004" ;tag=vBcXc0t99F3FH To: ;tag=7d1efd53 Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028757 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 405 bytes from udp/[10.1.1.4]:36762 at 20:37:22.573895: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bK0eZZNZetKjDKa Contact: To: ;tag=7d1efd53 From: "Extension 1004";tag=vBcXc0t99F3FH Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028757 INFO User-Agent: X-Lite release 4.7.1 stamp 74247 Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.578882 [DEBUG] switch_core_codec.c:246 sofia/internal/1004 at 10.1.1.1 Restore previous codec opus:116. 2015-01-08 20:37:22.578882 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1004 at 10.1.1.1: v=0 o=FreeSWITCH 1420710602 1420710604 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:22.580379: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:22.578882 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/1004 at 10.1.1.1] has been answered 2015-01-08 20:37:22.578882 [DEBUG] switch_channel.c:3689 (sofia/internal/1004 at 10.1.1.1) Callstate Change EARLY -> ACTIVE 2015-01-08 20:37:22.578882 [DEBUG] sofia.c:6614 Channel sofia/internal/1004 at 10.1.1.1 entering state [completed][200] 2015-01-08 20:37:22.578882 [DEBUG] switch_ivr_originate.c:3580 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.1.1.4:36762] 2015-01-08 20:37:22.578882 [DEBUG] switch_ivr_originate.c:3580 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.1.1.4:36762] 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:22.578882 [DEBUG] switch_ivr_bridge.c:1465 (sofia/internal/sip:1001 at 10.1.1.4:36762) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.578882 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1001 at 10.1.1.4:36762) Running State Change CS_EXCHANGE_MEDIA 2015-01-08 20:37:22.578882 [DEBUG] switch_core_state_machine.c:538 (sofia/internal/sip:1001 at 10.1.1.4:36762) State EXCHANGE_MEDIA 2015-01-08 20:37:22.578882 [DEBUG] mod_sofia.c:594 SOFIA EXCHANGE_MEDIA 2015-01-08 20:37:22.638892 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:22.658894 [DEBUG] switch_core_media.c:4336 sofia/internal/1004 at 10.1.1.1 Video thread paused. Echo is on 2015-01-08 20:37:22.658894 [DEBUG] switch_core_media.c:4336 sofia/internal/sip:1001 at 10.1.1.4:36762 Video thread paused. Echo is off send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:23.080509: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:23.559502: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:24.081505: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ recv 280 bytes from udp/[10.1.1.3]:21194 at 20:37:24.094926: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKyvcej9cKS00DK To: ;tag=fc96497b From: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Content-Length: 0 ------------------------------------------------------------------------ send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:26.082946: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:27.559939: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 280 bytes from udp/[10.1.1.3]:21194 at 20:37:27.563906: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKyvcej9cKS00DK To: ;tag=fc96497b From: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Content-Length: 0 ------------------------------------------------------------------------ send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:30.082953: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ 2015-01-08 20:37:30.578891 [DEBUG] switch_rtp.c:1942 rtcp_stats_init: ssrc[1815112177] base_seq[0] 2015-01-08 20:37:30.578891 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-08-00 15:07:30.589357] SSRC[1815112177]RTT[39699.102036] A[464688864] - DLSR[1297566794] - LSR[860369015] send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:31.559946: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 280 bytes from udp/[10.1.1.3]:21194 at 20:37:31.564058: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKyvcej9cKS00DK To: ;tag=fc96497b From: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:32.538884 [DEBUG] switch_rtp.c:1942 rtcp_stats_init: ssrc[2055342500] base_seq[0] 2015-01-08 20:37:32.538884 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-08-00 15:07:32.545585] SSRC[2055342500]RTT[35804.765488] A[464817067] - DLSR[1097159535] - LSR[1316123717] send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:34.082954: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:35.559952: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 280 bytes from udp/[10.1.1.3]:21194 at 20:37:35.563829: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKyvcej9cKS00DK To: ;tag=fc96497b From: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Content-Length: 0 ------------------------------------------------------------------------ send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:38.082944: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 freeswitch at internal> From shisheer at tifr.res.in Thu Jan 8 18:17:36 2015 From: shisheer at tifr.res.in (Shisheer Teli) Date: Thu, 8 Jan 2015 20:47:36 +0530 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: Kindly find the attached "sofia global siptrace on" logs .. where : 10.1.1.1 -> freeswitch server and 10.1.1.4 and 10.1.1.3 x-lite clients Regards, Shisheer T On Thu, Jan 8, 2015 at 8:25 PM, Brian West wrote: > do 'sofia global siptrace on', so we can see the sip trace. > > On Thu, Jan 8, 2015 at 8:52 AM, Shisheer Teli > wrote: > >> Kindly find the attached log details .. >> >> Regards, >> Shisheer T >> >> On Thu, Jan 8, 2015 at 8:12 PM, Brian West wrote: >> >>> What does the sip traffic say? >>> >>> On Thu, Jan 8, 2015 at 4:52 AM, Shisheer Teli >>> wrote: >>> >>>> i am using FreeSWITCH Version >>>> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >>>> 03:40:22Z 64bit) >>>> >>>> and still video call disconnect after 30 seconds >>>> >>>> Regards, >>>> Shisheer T >>>> >>>> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> Try latest master or 1.4.15 >>>>> >>>>> >>>>> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >>>>> wrote: >>>>> >>>>>> Hi Team, >>>>>> >>>>>> I don't know what happen , but when I start video call it >>>>>> disconnected after every 30 seconds. >>>>>> >>>>>> e.g. >>>>>> x-lite to x-lite call : video call disconnect after 30 seconds >>>>>> >>>>>> X-lite to Zoiper : video call continue, but no video sending. >>>>>> >>>>>> >>>>>> Regards >>>>>> Shisheer T >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>>> >>>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>>> http://twitter.com/FreeSWITCH >>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>>> * >>>>> >>>>> ClueCon Weekly Development Call >>>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> Shisheer Teli >>>> Phone: +91-022 2278 2519 / 2121 >>>> shisheer at tifr.res.in >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> >> >> -- >> Regards, >> Shisheer Teli >> Phone: +91-022 2278 2519 / 2121 >> shisheer at tifr.res.in >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/d274a077/attachment-0001.html -------------- next part -------------- freeswitch at internal> sofia global siptrace on +OK Global siptrace on freeswitch at internal> recv 1050 bytes from udp/[10.1.1.3]:21194 at 20:37:10.418033: ------------------------------------------------------------------------ INVITE sip:1001 at 10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-0e061d6b0960fa75-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4.7.0 73589-abcb99a9-W6.1 Content-Length: 508 v=0 o=- 13065203331951171 1 IN IP4 10.1.1.3 s=X-Lite release 4.7.0 stamp 73589 c=IN IP4 10.1.1.3 t=0 0 m=audio 56818 RTP/AVP 125 100 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 51488 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli a=sendrecv ------------------------------------------------------------------------ send 367 bytes to udp/[10.1.1.3]:21194 at 20:37:10.418426: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-0e061d6b0960fa75-1---d8754z-;rport=21194 From: ;tag=fc96497b To: Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:10.398899 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1004 at 10.1.1.1 [04ef4178-9748-11e4-981f-3136adddb849] 2015-01-08 20:37:10.398899 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.398899 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.398899 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004 at 10.1.1.1) Running State Change CS_NEW 2015-01-08 20:37:10.398899 [DEBUG] sofia.c:8834 sofia/internal/1004 at 10.1.1.1 receiving invite from 10.1.1.3:21194 version: 1.5.15b git 1ed290e 2015-01-08 03:40:22Z 64bit 2015-01-08 20:37:10.418893 [DEBUG] sofia.c:9001 IP 10.1.1.3 Rejected by acl "domains". Falling back to Digest auth. 2015-01-08 20:37:10.418893 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1004 at 10.1.1.1) State NEW send 871 bytes to udp/[10.1.1.3]:21194 at 20:37:10.419509: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-0e061d6b0960fa75-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=tSSB99r2Fypaa Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="10.1.1.1", nonce="04ef5302-9748-11e4-9820-3136adddb849", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:10.418893 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.418893 [DEBUG] sofia.c:2067 detaching session 04ef4178-9748-11e4-981f-3136adddb849 recv 330 bytes from udp/[10.1.1.3]:21194 at 20:37:10.423840: ------------------------------------------------------------------------ ACK sip:1001 at 10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-0e061d6b0960fa75-1---d8754z-;rport Max-Forwards: 70 To: ;tag=tSSB99r2Fypaa From: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 1312 bytes from udp/[10.1.1.3]:21194 at 20:37:10.429397: ------------------------------------------------------------------------ INVITE sip:1001 at 10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1004",realm="10.1.1.1",nonce="04ef5302-9748-11e4-9820-3136adddb849",uri="sip:1001 at 10.1.1.1",response="481a9aca4f69b1b4fb3ea4426111efa4",cnonce="5dbdd76fed771a60c49b5a60917f09a4",nc=00000001,qop=auth,algorithm=MD5 Supported: replaces User-Agent: X-Lite 4.7.0 73589-abcb99a9-W6.1 Content-Length: 508 v=0 o=- 13065203331951171 1 IN IP4 10.1.1.3 s=X-Lite release 4.7.0 stamp 73589 c=IN IP4 10.1.1.3 t=0 0 m=audio 56818 RTP/AVP 125 100 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 51488 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli a=sendrecv ------------------------------------------------------------------------ send 367 bytes to udp/[10.1.1.3]:21194 at 20:37:10.429689: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:10.418893 [DEBUG] sofia.c:2175 Re-attaching to session 04ef4178-9748-11e4-981f-3136adddb849 2015-01-08 20:37:10.418893 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.418893 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:8834 sofia/internal/1004 at 10.1.1.1 receiving invite from 10.1.1.3:21194 version: 1.5.15b git 1ed290e 2015-01-08 03:40:22Z 64bit 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:9001 IP 10.1.1.3 Rejected by acl "domains". Falling back to Digest auth. 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:6614 Channel sofia/internal/1004 at 10.1.1.1 entering state [received][100] 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=- 13065203331951171 1 IN IP4 10.1.1.3 s=X-Lite release 4.7.0 stamp 73589 c=IN IP4 10.1.1.3 t=0 0 m=audio 56818 RTP/AVP 125 100 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 51488 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli 2015-01-08 20:37:10.438894 [DEBUG] sofia.c:6890 (sofia/internal/1004 at 10.1.1.1) State Change CS_NEW -> CS_INIT 2015-01-08 20:37:10.438894 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004 at 10.1.1.1) Running State Change CS_INIT 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1004 at 10.1.1.1) State INIT 2015-01-08 20:37:10.438894 [DEBUG] mod_sofia.c:87 sofia/internal/1004 at 10.1.1.1 SOFIA INIT 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1004 at 10.1.1.1 Standard INIT 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1004 at 10.1.1.1) State Change CS_INIT -> CS_ROUTING 2015-01-08 20:37:10.438894 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1004 at 10.1.1.1) State INIT going to sleep 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004 at 10.1.1.1) Running State Change CS_ROUTING 2015-01-08 20:37:10.438894 [DEBUG] switch_channel.c:2184 (sofia/internal/1004 at 10.1.1.1) Callstate Change DOWN -> RINGING 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1004 at 10.1.1.1) State ROUTING 2015-01-08 20:37:10.438894 [DEBUG] mod_sofia.c:123 sofia/internal/1004 at 10.1.1.1 SOFIA ROUTING 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1004 at 10.1.1.1 Standard ROUTING 2015-01-08 20:37:10.438894 [INFO] mod_dialplan_xml.c:635 Processing 1004 <1004>->1001 in context default Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->unloop] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->tod_example] continue=true Dialplan: sofia/internal/1004 at 10.1.1.1 Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1004 at 10.1.1.1 Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->redial] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->global] continue=true Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (PASS) [global] ${default_password}(1234) =~ /^1234$/ break=never Dialplan: sofia/internal/1004 at 10.1.1.1 Action log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) Dialplan: sofia/internal/1004 at 10.1.1.1 Action log(CRIT Open /usr/local/freeswitch/conf/vars.xml and change the default_password.) Dialplan: sofia/internal/1004 at 10.1.1.1 Action log(CRIT Once changed type 'reloadxml' at the console.) Dialplan: sofia/internal/1004 at 10.1.1.1 Action log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) Dialplan: sofia/internal/1004 at 10.1.1.1 Action sleep(10000) Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [global] ${rtp_has_crypto}() =~ /^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/ break=never Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [global] ${switch_r_sdp}(v=0 o=- 13065203331951171 1 IN IP4 10.1.1.3 s=X-Lite release 4.7.0 stamp 73589 c=IN IP4 10.1.1.3 t=0 0 m=audio 56818 RTP/AVP 125 100 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 51488 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never Dialplan: sofia/internal/1004 at 10.1.1.1 Absolute Condition [global] Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->call_return] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->del-group] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->add-group] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1004 at 10.1.1.1 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1004 at 10.1.1.1 Action export(dialed_extension=1001) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(ringback=${us-ring}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(call_timeout=30) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(continue_on_fail=true) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1004 at 10.1.1.1 Action answer() Dialplan: sofia/internal/1004 at 10.1.1.1 Action sleep(1000) Dialplan: sofia/internal/1004 at 10.1.1.1 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1004 at 10.1.1.1) State Change CS_ROUTING -> CS_EXECUTE 2015-01-08 20:37:10.438894 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1004 at 10.1.1.1) State ROUTING going to sleep 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004 at 10.1.1.1) Running State Change CS_EXECUTE 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1004 at 10.1.1.1) State EXECUTE 2015-01-08 20:37:10.438894 [DEBUG] mod_sofia.c:178 sofia/internal/1004 at 10.1.1.1 SOFIA EXECUTE 2015-01-08 20:37:10.438894 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1004 at 10.1.1.1 Standard EXECUTE EXECUTE sofia/internal/1004 at 10.1.1.1 log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) 2015-01-08 20:37:10.438894 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING EXECUTE sofia/internal/1004 at 10.1.1.1 log(CRIT Open /usr/local/freeswitch/conf/vars.xml and change the default_password.) 2015-01-08 20:37:10.438894 [CRIT] mod_dptools.c:1628 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. EXECUTE sofia/internal/1004 at 10.1.1.1 log(CRIT Once changed type 'reloadxml' at the console.) 2015-01-08 20:37:10.438894 [CRIT] mod_dptools.c:1628 Once changed type 'reloadxml' at the console. EXECUTE sofia/internal/1004 at 10.1.1.1 log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) 2015-01-08 20:37:10.438894 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING EXECUTE sofia/internal/1004 at 10.1.1.1 sleep(10000) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-spymap/1004/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial/1004/1001) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial/global/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 export(RFC2822_DATE=Thu, 08 Jan 2015 20:37:20 +0530) 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 08 Jan 2015 20:37:20 +0530] EXECUTE sofia/internal/1004 at 10.1.1.1 export(dialed_extension=1001) 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at 10.1.1.1 bind_meta_app(1 b s execute_extension::dx XML features) 2015-01-08 20:37:20.458892 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/1004 at 10.1.1.1 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1004.2015-01-08-20-37-20.wav) 2015-01-08 20:37:20.458892 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1004.2015-01-08-20-37-20.wav EXECUTE sofia/internal/1004 at 10.1.1.1 bind_meta_app(3 b s execute_extension::cf XML features) 2015-01-08 20:37:20.458892 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/1004 at 10.1.1.1 bind_meta_app(4 b s execute_extension::att_xfer XML features) 2015-01-08 20:37:20.458892 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/1004 at 10.1.1.1 set(ringback=%(2000,4000,440,480)) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/1004 at 10.1.1.1 set(transfer_ringback=local_stream://moh) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1004 at 10.1.1.1 set(call_timeout=30) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [call_timeout]=[30] EXECUTE sofia/internal/1004 at 10.1.1.1 set(hangup_after_bridge=true) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1004 at 10.1.1.1 set(continue_on_fail=true) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [continue_on_fail]=[true] EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-call_return/1001/1004) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial_ext/1001/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 set(called_party_callgroup=techsupport) 2015-01-08 20:37:20.458892 [DEBUG] mod_dptools.c:1435 sofia/internal/1004 at 10.1.1.1 SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial_ext/techsupport/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial_ext/global/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 hash(insert/10.1.1.1-last_dial/techsupport/04ef4178-9748-11e4-981f-3136adddb849) EXECUTE sofia/internal/1004 at 10.1.1.1 bridge(user/1001 at 10.1.1.1) 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1201 sofia/internal/1004 at 10.1.1.1 EXPORTING[export_vars] [RFC2822_DATE]=[Thu, 08 Jan 2015 20:37:20 +0530] to event 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1201 sofia/internal/1004 at 10.1.1.1 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2015-01-08 20:37:20.458892 [DEBUG] switch_ivr_originate.c:2103 Parsing global variables 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1201 sofia/internal/1004 at 10.1.1.1 EXPORTING[export_vars] [RFC2822_DATE]=[Thu, 08 Jan 2015 20:37:20 +0530] to event 2015-01-08 20:37:20.458892 [DEBUG] switch_channel.c:1201 sofia/internal/1004 at 10.1.1.1 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2015-01-08 20:37:20.458892 [DEBUG] switch_ivr_originate.c:2103 Parsing global variables 2015-01-08 20:37:20.458892 [DEBUG] switch_event.c:1688 Parsing variable [sip_invite_domain]=[10.1.1.1] 2015-01-08 20:37:20.458892 [DEBUG] switch_event.c:1688 Parsing variable [presence_id]=[1001 at 10.1.1.1] 2015-01-08 20:37:20.458892 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:1001 at 10.1.1.4:36762 [0aece904-9748-11e4-9842-3136adddb849] 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:4636 (sofia/internal/sip:1001 at 10.1.1.4:36762) State Change CS_NEW -> CS_INIT 2015-01-08 20:37:20.458892 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:4706 [zrtp_passthru] Setting a-leg inherit_codec=true 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:4709 [zrtp_passthru] Setting b-leg absolute_codec_string='opus at 48000h@20i at 2c,PCMU at 8000h@20i at 64000b,G722 at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b,H263 at 90000h@1c,H263-1998 at 90000h@1c' 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1001 at 10.1.1.4:36762) Running State Change CS_INIT 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/sip:1001 at 10.1.1.4:36762) State INIT 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:87 sofia/internal/sip:1001 at 10.1.1.4:36762 SOFIA INIT 2015-01-08 20:37:20.458892 [DEBUG] sofia_glue.c:1232 sofia/internal/sip:1001 at 10.1.1.4:36762 sending invite version: 1.5.15b git 1ed290e 2015-01-08 03:40:22Z 64bit Local SDP: v=0 o=FreeSWITCH 1420703660 1420703661 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 25980 RTP/AVP 125 0 9 8 101 13 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 30320 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:40 sofia/internal/sip:1001 at 10.1.1.4:36762 Standard INIT 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/sip:1001 at 10.1.1.4:36762) State Change CS_INIT -> CS_ROUTING 2015-01-08 20:37:20.458892 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/sip:1001 at 10.1.1.4:36762) State INIT going to sleep 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1001 at 10.1.1.4:36762) Running State Change CS_ROUTING send 1482 bytes to udp/[10.1.1.4]:36762 at 20:37:20.471070: ------------------------------------------------------------------------ INVITE sip:1001 at 10.1.1.4:36762;rinstance=18b07042dd52441e SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKXKKNgevFvQaUQ Max-Forwards: 69 From: "Extension 1004" ;tag=vBcXc0t99F3FH To: Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028756 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 465 X-FS-Support: update_display,send_info Remote-Party-ID: "Extension 1004" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420703660 1420703661 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 25980 RTP/AVP 125 0 9 8 101 13 a=rtpmap:125 opus/48000/2 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/sip:1001 at 10.1.1.4:36762) State ROUTING a=fmtp:125 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 2015-01-08 20:37:20.458892 [DEBUG] mod_sofia.c:123 sofia/internal/sip:1001 at 10.1.1.4:36762 SOFIA ROUTING a=fmtp:101 0-16 a=ptime:20 m=video 30320 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 2015-01-08 20:37:20.458892 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:1001 at 10.1.1.4:36762) State Change CS_ROUTING -> CS_CONSUME_MEDIA a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 ------------------------------------------------------------------------ 2015-01-08 20:37:20.458892 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/sip:1001 at 10.1.1.4:36762) State ROUTING going to sleep 2015-01-08 20:37:20.458892 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1001 at 10.1.1.4:36762) Running State Change CS_CONSUME_MEDIA 2015-01-08 20:37:20.458892 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1001 at 10.1.1.4:36762 entering state [calling][0] 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/sip:1001 at 10.1.1.4:36762) State CONSUME_MEDIA 2015-01-08 20:37:20.458892 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/sip:1001 at 10.1.1.4:36762) State CONSUME_MEDIA going to sleep recv 412 bytes from udp/[10.1.1.4]:36762 at 20:37:20.550219: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKXKKNgevFvQaUQ Contact: To: ;tag=7d1efd53 From: "Extension 1004";tag=vBcXc0t99F3FH Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028756 INVITE User-Agent: X-Lite release 4.7.1 stamp 74247 Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:20.538891 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:20.538891 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1001 at 10.1.1.4:36762 entering state [proceeding][180] 2015-01-08 20:37:20.538891 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:1001 at 10.1.1.4:36762! 2015-01-08 20:37:20.538891 [DEBUG] switch_channel.c:3277 (sofia/internal/sip:1001 at 10.1.1.4:36762) Callstate Change DOWN -> RINGING 2015-01-08 20:37:20.538891 [INFO] switch_ivr_originate.c:1192 Sending early media 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3543 Set telephone-event payload to 101 2015-01-08 20:37:20.538891 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:20.538891 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/1004 at 10.1.1.1 opus/48000 20 ms 960 samples 0 bits 1 channels 2015-01-08 20:37:20.538891 [DEBUG] switch_core_codec.c:111 sofia/internal/1004 at 10.1.1.1 Original read codec set to opus:116 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:3890 Set 2833 dtmf send/recv payload to 101 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H261:31] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263:34] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263-1998:115] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4020 Video Codec Compare [H263-1998:115] +++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263-2000:121] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H264:97] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[VP8:99] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H261:31] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263:34] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4020 Video Codec Compare [H263:34] +++ is saved as a match 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263-1998:115] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263-2000:121] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H264:97] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[VP8:99] 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:2293 Set VIDEO Codec sofia/internal/1004 at 10.1.1.1 H263-1998/90000 0 ms 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/internal/1004 at 10.1.1.1] 10.1.1.1 port 19038 -> 10.1.1.3 port 56818 codec: 125 ms: 20 2015-01-08 20:37:20.538891 [DEBUG] switch_rtp.c:3548 Starting timer [soft] 960 bytes per 20ms 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:5439 Set 2833 dtmf send payload to 101 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:5445 Set 2833 dtmf receive payload to 101 2015-01-08 20:37:20.538891 [DEBUG] switch_rtp.c:3557 Not using a timer 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:5643 VIDEO RTP [sofia/internal/1004 at 10.1.1.1] 10.1.1.3:29642->10.1.1.3:51488 codec: 115 ms: 0 [SUCCESS] 2015-01-08 20:37:20.538891 [NOTICE] switch_core_media.c:4405 sofia/internal/1004 at 10.1.1.1 Starting Video thread 2015-01-08 20:37:20.538891 [DEBUG] switch_core_media.c:4329 sofia/internal/1004 at 10.1.1.1 Video thread started. Echo is on 2015-01-08 20:37:20.538891 [INFO] switch_core_media.c:5723 Activating VIDEO RTCP PORT 0 mux -1 2015-01-08 20:37:20.538891 [DEBUG] switch_rtp.c:3898 RTCP send rate is: 10000 and packet rate is: 90000 Remote Port: 51489 2015-01-08 20:37:20.538891 [DEBUG] switch_rtp.c:2354 Setting RTCP remote addr to 10.1.1.3:51489 2015-01-08 20:37:20.538891 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1004 at 10.1.1.1! 2015-01-08 20:37:20.538891 [DEBUG] switch_channel.c:3399 (sofia/internal/1004 at 10.1.1.1) Callstate Change RINGING -> EARLY 2015-01-08 20:37:20.538891 [DEBUG] mod_sofia.c:2268 Ring SDP: v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 2015-01-08 20:37:20.538891 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:20.538891 [DEBUG] switch_ivr_originate.c:1249 Raw Codec Activation Success L16 at 48000hz 1 channel 20ms 2015-01-08 20:37:20.538891 [DEBUG] switch_core_codec.c:221 sofia/internal/1004 at 10.1.1.1 Push codec L16:70 2015-01-08 20:37:20.538891 [DEBUG] switch_ivr_originate.c:1317 Play Ringback Tone [%(2000,4000,440,480)] send 1336 bytes to udp/[10.1.1.3]:21194 at 20:37:20.558402: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "1001" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ 2015-01-08 20:37:20.538891 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:20.558650: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ 2015-01-08 20:37:20.578891 [DEBUG] sofia.c:6614 Channel sofia/internal/1004 at 10.1.1.1 entering state [early][183] 2015-01-08 20:37:20.618894 [DEBUG] switch_rtp.c:5853 Correct ip/port confirmed. send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:21.559269: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 1033 bytes from udp/[10.1.1.4]:36762 at 20:37:22.558754: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKXKKNgevFvQaUQ Contact: To: ;tag=7d1efd53 From: "Extension 1004";tag=vBcXc0t99F3FH Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028756 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.7.1 stamp 74247 Content-Length: 478 v=0 o=- 13065203368350614 3 IN IP4 10.1.1.4 s=X-Lite release 4.7.1 stamp 74247 c=IN IP4 10.1.1.4 t=0 0 m=audio 62698 RTP/AVP 125 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 49936 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=sendrecv a=rtcp-fb:* nack pli ------------------------------------------------------------------------ 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.558880 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1001 at 10.1.1.4:36762 entering state [completing][200] 2015-01-08 20:37:22.558880 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=- 13065203368350614 3 IN IP4 10.1.1.4 s=X-Lite release 4.7.1 stamp 74247 c=IN IP4 10.1.1.4 t=0 0 m=audio 62698 RTP/AVP 125 0 9 8 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 49936 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp-fb:* nack pli send 402 bytes to udp/[10.1.1.4]:36762 at 20:37:22.560661: ------------------------------------------------------------------------ ACK sip:1001 at 10.1.1.4:36762 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKZ556K4Xpp9p0e Max-Forwards: 70 From: "Extension 1004" ;tag=vBcXc0t99F3FH To: ;tag=7d1efd53 Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028756 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.558880 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1001 at 10.1.1.4:36762 entering state [ready][200] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[opus:116:48000:20:0:2] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [opus:116:48000:20:0:2] ++++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:2] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:2] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:2] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3543 Set telephone-event payload to 101 2015-01-08 20:37:22.558880 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:22.558880 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/sip:1001 at 10.1.1.4:36762 opus/48000 20 ms 960 samples 0 bits 1 channels 2015-01-08 20:37:22.558880 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1001 at 10.1.1.4:36762 Original read codec set to opus:116 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:3881 Set 2833 dtmf send payload to 101 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263:34] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4020 Video Codec Compare [H263:34] +++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263:34]/[H263-1998:115] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263:34] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4007 Video Codec Compare [H263-1998:115]/[H263-1998:115] 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4020 Video Codec Compare [H263-1998:115] +++ is saved as a match 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:2293 Set VIDEO Codec sofia/internal/sip:1001 at 10.1.1.4:36762 H263/90000 0 ms 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/internal/sip:1001 at 10.1.1.4:36762] 10.1.1.1 port 25980 -> 10.1.1.4 port 62698 codec: 125 ms: 20 2015-01-08 20:37:22.558880 [DEBUG] switch_rtp.c:3548 Starting timer [soft] 960 bytes per 20ms 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:5439 Set 2833 dtmf send payload to 101 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:5445 Set 2833 dtmf receive payload to 101 2015-01-08 20:37:22.558880 [DEBUG] switch_rtp.c:3557 Not using a timer 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:5643 VIDEO RTP [sofia/internal/sip:1001 at 10.1.1.4:36762] 10.1.1.4:30320->10.1.1.4:49936 codec: 34 ms: 0 [SUCCESS] 2015-01-08 20:37:22.558880 [NOTICE] switch_core_media.c:4405 sofia/internal/sip:1001 at 10.1.1.4:36762 Starting Video thread 2015-01-08 20:37:22.558880 [INFO] switch_core_media.c:5723 Activating VIDEO RTCP PORT 0 mux -1 2015-01-08 20:37:22.558880 [DEBUG] switch_core_media.c:4329 sofia/internal/sip:1001 at 10.1.1.4:36762 Video thread started. Echo is off 2015-01-08 20:37:22.558880 [DEBUG] switch_rtp.c:3898 RTCP send rate is: 10000 and packet rate is: 90000 Remote Port: 49937 2015-01-08 20:37:22.558880 [DEBUG] switch_rtp.c:2354 Setting RTCP remote addr to 10.1.1.4:49937 2015-01-08 20:37:22.558880 [DEBUG] switch_channel.c:3635 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:22.558880 [NOTICE] sofia.c:7475 Channel [sofia/internal/sip:1001 at 10.1.1.4:36762] has been answered 2015-01-08 20:37:22.558880 [DEBUG] switch_channel.c:3689 (sofia/internal/sip:1001 at 10.1.1.4:36762) Callstate Change RINGING -> ACTIVE send 850 bytes to udp/[10.1.1.4]:36762 at 20:37:22.566368: ------------------------------------------------------------------------ INFO sip:1001 at 10.1.1.4:36762 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bK0eZZNZetKjDKa Max-Forwards: 70 From: "Extension 1004" ;tag=vBcXc0t99F3FH To: ;tag=7d1efd53 Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028757 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 405 bytes from udp/[10.1.1.4]:36762 at 20:37:22.573895: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bK0eZZNZetKjDKa Contact: To: ;tag=7d1efd53 From: "Extension 1004";tag=vBcXc0t99F3FH Call-ID: e24b5e56-11ea-1233-39bf-005056aa6ca1 CSeq: 70028757 INFO User-Agent: X-Lite release 4.7.1 stamp 74247 Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:22.558880 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.578882 [DEBUG] switch_core_codec.c:246 sofia/internal/1004 at 10.1.1.1 Restore previous codec opus:116. 2015-01-08 20:37:22.578882 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1004 at 10.1.1.1: v=0 o=FreeSWITCH 1420710602 1420710604 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:22.580379: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:22.578882 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/1004 at 10.1.1.1] has been answered 2015-01-08 20:37:22.578882 [DEBUG] switch_channel.c:3689 (sofia/internal/1004 at 10.1.1.1) Callstate Change EARLY -> ACTIVE 2015-01-08 20:37:22.578882 [DEBUG] sofia.c:6614 Channel sofia/internal/1004 at 10.1.1.1 entering state [completed][200] 2015-01-08 20:37:22.578882 [DEBUG] switch_ivr_originate.c:3580 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.1.1.4:36762] 2015-01-08 20:37:22.578882 [DEBUG] switch_ivr_originate.c:3580 Originate Resulted in Success: [sofia/internal/sip:1001 at 10.1.1.4:36762] 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/1004 at 10.1.1.1 [BREAK] 2015-01-08 20:37:22.578882 [DEBUG] switch_ivr_bridge.c:1465 (sofia/internal/sip:1001 at 10.1.1.4:36762) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2015-01-08 20:37:22.578882 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1001 at 10.1.1.4:36762 [BREAK] 2015-01-08 20:37:22.578882 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1001 at 10.1.1.4:36762) Running State Change CS_EXCHANGE_MEDIA 2015-01-08 20:37:22.578882 [DEBUG] switch_core_state_machine.c:538 (sofia/internal/sip:1001 at 10.1.1.4:36762) State EXCHANGE_MEDIA 2015-01-08 20:37:22.578882 [DEBUG] mod_sofia.c:594 SOFIA EXCHANGE_MEDIA 2015-01-08 20:37:22.638892 [DEBUG] mod_opus.c:287 Opus encoder set bitrate to local settings [-1000bps] 2015-01-08 20:37:22.658894 [DEBUG] switch_core_media.c:4336 sofia/internal/1004 at 10.1.1.1 Video thread paused. Echo is on 2015-01-08 20:37:22.658894 [DEBUG] switch_core_media.c:4336 sofia/internal/sip:1001 at 10.1.1.4:36762 Video thread paused. Echo is off send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:23.080509: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:23.559502: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:24.081505: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ recv 280 bytes from udp/[10.1.1.3]:21194 at 20:37:24.094926: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKyvcej9cKS00DK To: ;tag=fc96497b From: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Content-Length: 0 ------------------------------------------------------------------------ send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:26.082946: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:27.559939: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 280 bytes from udp/[10.1.1.3]:21194 at 20:37:27.563906: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKyvcej9cKS00DK To: ;tag=fc96497b From: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Content-Length: 0 ------------------------------------------------------------------------ send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:30.082953: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ 2015-01-08 20:37:30.578891 [DEBUG] switch_rtp.c:1942 rtcp_stats_init: ssrc[1815112177] base_seq[0] 2015-01-08 20:37:30.578891 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-08-00 15:07:30.589357] SSRC[1815112177]RTT[39699.102036] A[464688864] - DLSR[1297566794] - LSR[860369015] send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:31.559946: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 280 bytes from udp/[10.1.1.3]:21194 at 20:37:31.564058: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKyvcej9cKS00DK To: ;tag=fc96497b From: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Content-Length: 0 ------------------------------------------------------------------------ 2015-01-08 20:37:32.538884 [DEBUG] switch_rtp.c:1942 rtcp_stats_init: ssrc[2055342500] base_seq[0] 2015-01-08 20:37:32.538884 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-08-00 15:07:32.545585] SSRC[2055342500]RTT[35804.765488] A[464817067] - DLSR[1097159535] - LSR[1316123717] send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:34.082954: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtcp:29643 IN IP4 10.1.1.1 ------------------------------------------------------------------------ send 816 bytes to udp/[10.1.1.3]:21194 at 20:37:35.559952: ------------------------------------------------------------------------ INFO sip:1004 at 10.1.1.3:21194 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1;rport;branch=z9hG4bKyvcej9cKS00DK Max-Forwards: 70 From: ;tag=U2j4a595c7cXN To: ;tag=fc96497b Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: application/media_control+xml Content-Length: 175 ------------------------------------------------------------------------ recv 280 bytes from udp/[10.1.1.3]:21194 at 20:37:35.563829: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1;rport=5060;branch=z9hG4bKyvcej9cKS00DK To: ;tag=fc96497b From: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 70028756 INFO Content-Length: 0 ------------------------------------------------------------------------ send 1306 bytes to udp/[10.1.1.3]:21194 at 20:37:38.082944: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:21194;branch=z9hG4bK-d8754z-7013cd70d2e6694e-1---d8754z-;rport=21194 From: ;tag=fc96497b To: ;tag=U2j4a595c7cXN Call-ID: OWViNzk2ZmFlMzQ1YWFkZTY3ZDM1ODMyNmQ5ZGIwMjI CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150108T034022Z~1ed290e930~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 398 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1420710602 1420710603 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 19038 RTP/AVP 125 101 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29642 RTP/AVP 115 a=rtpmap:115 H263-1998/90000 freeswitch at internal> From aqsyounas at gmail.com Thu Jan 8 18:22:36 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 8 Jan 2015 20:22:36 +0500 Subject: [Freeswitch-users] how can i do stream piping in freeswitch In-Reply-To: References: <54ADDBF2.50106@mst.edu> Message-ID: Actually we are running a radio setup in which users listen to different radio's streams by pressing the desired extension for that radio. We have more than 400 radio streams and concurrent peak of 350 calls. For example, 100 users are listening to same station and after couple of seconds they will be listening to different station. Currently all these connections are being made with stream provider, i.e, 100 connections will be open with stream provider. We want is that if a user is listening to a specific station and another user also wants to listen that station he(second user) must share the stream or connection created by the first user instead of creating a separate connection with stream provider. Thanks for you time. On 8 January 2015 at 19:56, Brian West wrote: > You're going to need 400 local streams, or to re-think you approach. > > On Thu, Jan 8, 2015 at 8:47 AM, Aqs Younas wrote: > >> Thanks for your suggestions. It really helped. >> >> But i have few questions. >> >> What if i have multiple streams actually about 400 streams and a lot of >> users who can switch between different streams, if i put all stream in a >> file then users will not be able to switch between desired streams. >> >> And if i create separate folder each containing a file with single stream >> then there will be a lots of directories and for every directory and also i >> have to manually enter its name in local_stream.conf.xml >> >> Because i see this setup suitable for only single stream or in a scenario >> where users don't want to switch between desired streams by pressing >> extension. >> >> Is there any other way to do so.? Or how can i improve this? >> >> Really thankful for your help. >> >> Regards. >> >> On 8 January 2015 at 06:22, Nathan Neulinger wrote: >> >>> Look at the Shout stream as MOH section of: >>> >>> https://wiki.freeswitch.org/wiki/Mod_shout >>> >>> You have to define the stream as a local_stream and then reference it >>> via local_stream://moh/whatever >>> >>> I use this to stream a local campus radio station via mp3/icecast as MOH >>> stream. >>> >>> -- Nathan >>> >>> On 01/07/2015 03:21 PM, Aqs Younas wrote: >>> > Currently i am playing a stream with mod_shout and this is my default >>> xml. >>> > >>> > >>> > >>> > >> expression="^14049002000$"> >>> > >>> > >>> > >>> > >>> > >>> > >>> > Every time a user makes a call for stream it opens a separate >>> connection with the stream provider. If 100 users dials >>> > this number, there would be 100 connections with stream provider >>> listening to same stream, means more rtp packets >>> > containing same data for different users. >>> > >>> > What i want is, if a user is listening to stream then other users must >>> share the same listening connection that the >>> > first user is opened, instead of creating a separate connection with >>> stream provider for same stream. >>> > >>> > Someone told me this is possible in asterisk, so there must be a way >>> in freeswitch. >>> > >>> > How can i do this.? >>> > Any help would be much appreciated. >>> > >>> > Thanks >>> > >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > 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 >>> > >>> >>> -- >>> ------------------------------------------------------------ >>> Nathan Neulinger nneul at mst.edu >>> Missouri S&T Information Technology (573) 612-1412 >>> System Administrator - Architect >>> >>> _________________________________________________________________________ >>> 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/89b238af/attachment.html From aqsyounas at gmail.com Thu Jan 8 18:44:44 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 8 Jan 2015 20:44:44 +0500 Subject: [Freeswitch-users] how can i do stream piping in freeswitch In-Reply-To: References: <54ADDBF2.50106@mst.edu> Message-ID: Thank you guys for your valuable suggestions. It really helped me. On 8 January 2015 at 20:22, Aqs Younas wrote: > Actually we are running a radio setup in which users listen to different > radio's streams by pressing the desired extension for that radio. > > We have more than 400 radio streams and concurrent peak of 350 calls. For > example, 100 users are listening to same station and after couple of > seconds they will be listening to different station. Currently all these > connections are being made with stream provider, i.e, 100 connections will > be open with stream provider. > > We want is that if a user is listening to a specific station and another > user also wants to listen that station he(second user) must share the > stream or connection created by the first user instead of creating a > separate connection with stream provider. > > Thanks for you time. > > On 8 January 2015 at 19:56, Brian West wrote: > >> You're going to need 400 local streams, or to re-think you approach. >> >> On Thu, Jan 8, 2015 at 8:47 AM, Aqs Younas wrote: >> >>> Thanks for your suggestions. It really helped. >>> >>> But i have few questions. >>> >>> What if i have multiple streams actually about 400 streams and a lot of >>> users who can switch between different streams, if i put all stream in a >>> file then users will not be able to switch between desired streams. >>> >>> And if i create separate folder each containing a file with single >>> stream then there will be a lots of directories and for every directory and >>> also i have to manually enter its name in local_stream.conf.xml >>> >>> Because i see this setup suitable for only single stream or in a >>> scenario where users don't want to switch between desired streams by >>> pressing extension. >>> >>> Is there any other way to do so.? Or how can i improve this? >>> >>> Really thankful for your help. >>> >>> Regards. >>> >>> On 8 January 2015 at 06:22, Nathan Neulinger wrote: >>> >>>> Look at the Shout stream as MOH section of: >>>> >>>> https://wiki.freeswitch.org/wiki/Mod_shout >>>> >>>> You have to define the stream as a local_stream and then reference it >>>> via local_stream://moh/whatever >>>> >>>> I use this to stream a local campus radio station via mp3/icecast as >>>> MOH stream. >>>> >>>> -- Nathan >>>> >>>> On 01/07/2015 03:21 PM, Aqs Younas wrote: >>>> > Currently i am playing a stream with mod_shout and this is my default >>>> xml. >>>> > >>>> > >>>> > >>>> > >>> expression="^14049002000$"> >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > Every time a user makes a call for stream it opens a separate >>>> connection with the stream provider. If 100 users dials >>>> > this number, there would be 100 connections with stream provider >>>> listening to same stream, means more rtp packets >>>> > containing same data for different users. >>>> > >>>> > What i want is, if a user is listening to stream then other users >>>> must share the same listening connection that the >>>> > first user is opened, instead of creating a separate connection with >>>> stream provider for same stream. >>>> > >>>> > Someone told me this is possible in asterisk, so there must be a way >>>> in freeswitch. >>>> > >>>> > How can i do this.? >>>> > Any help would be much appreciated. >>>> > >>>> > Thanks >>>> > >>>> > >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > 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 >>>> > >>>> >>>> -- >>>> ------------------------------------------------------------ >>>> Nathan Neulinger nneul at mst.edu >>>> Missouri S&T Information Technology (573) 612-1412 >>>> System Administrator - Architect >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >> >> _________________________________________________________________________ >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/84299aae/attachment-0001.html From brian at freeswitch.org Thu Jan 8 20:26:14 2015 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2015 11:26:14 -0600 Subject: [Freeswitch-users] Wrong ice candidate (internal ip) when using mod_verto In-Reply-To: References: Message-ID: I noticed you quit IRC without your question being answered, I would highly recommend you email consulting at freeswitch.org for more information about this. On Wed, Jan 7, 2015 at 4:21 PM, Adam Ben-Ayoun wrote: > Hi, > > I am experimenting with mod_verto and using the demo to call to a > conference on my server (logs on FreeSwitch seems ok generally), the issue > is that I am not receiving/sending any audio (I would expect to hear moh at > this point), I suspect this is because FreeSwitch sends an ice host > candidate set to the internal ip instead of the external, here is the SDP: > > o=FreeSWITCH 1420641299 1420641300 IN IP4 10.245.43.195 > s=FreeSWITCH > c=IN IP4 10.245.43.195 > t=0 0 > a=msid-semantic: WMS Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br > m=audio 26060 RTP/SAVPF 111 126 > a=rtpmap:111 opus/48000/2 > a=fmtp:111 maxplaybackrate=48000; minptime=10 > a=rtpmap:126 telephone-event/8000 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 > 0D:BB:87:E0:B6:8A:AE:0D:32:3F:54:9F:36:FF:9B:89:28:AB:B4:17:BC:95:0D:2E:48:0C:33:30:78:52:5F:24 > a=rtcp-mux > a=rtcp:26060 IN IP4 10.245.43.195 > a=ssrc:1435389439 cname:TfPkXSQ755IUEU1s > a=ssrc:1435389439 msid:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br a0 > a=ssrc:1435389439 mslabel:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Br > a=ssrc:1435389439 label:Z2NxBScFjmpLZIlgYwYbxA7GyEbpC0Bra0 > a=ice-ufrag:hQuEgraFWcUo2lGL > a=ice-pwd:qmWe5Cb0xJL2tDkAjF4pW5UJ > a=candidate:0460306999 1 udp 659136 10.245.43.195 26060 typ host > generation 0 > > Using Wireshark I see that the client is trying to send packets to > 10.245.43.195 which obviously fails.. > > How can I fix this? > > Thanks, > Adam > > _________________________________________________________________________ > 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/20150108/f9f78f46/attachment.html From mike at jerris.com Thu Jan 8 20:54:40 2015 From: mike at jerris.com (Michael Jerris) Date: Thu, 8 Jan 2015 12:54:40 -0500 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: This is some sort of nat issue.. we are sending the 200 OK and not getting an ack back. Double check all your nat related configuration. > On Jan 8, 2015, at 10:17 AM, Shisheer Teli wrote: > > Kindly find the attached "sofia global siptrace on" logs .. > > where : > 10.1.1.1 -> freeswitch server > and > 10.1.1.4 and 10.1.1.3 x-lite clients > > Regards, > Shisheer T > > On Thu, Jan 8, 2015 at 8:25 PM, Brian West > wrote: > do 'sofia global siptrace on', so we can see the sip trace. > > On Thu, Jan 8, 2015 at 8:52 AM, Shisheer Teli > wrote: > Kindly find the attached log details .. > > Regards, > Shisheer T > > On Thu, Jan 8, 2015 at 8:12 PM, Brian West > wrote: > What does the sip traffic say? > > On Thu, Jan 8, 2015 at 4:52 AM, Shisheer Teli > wrote: > i am using FreeSWITCH Version 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 03:40:22Z 64bit) > > and still video call disconnect after 30 seconds > > Regards, > Shisheer T > > On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale > wrote: > Try latest master or 1.4.15 > > > On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli > wrote: > Hi Team, > > I don't know what happen , but when I start video call it disconnected after every 30 seconds. > > e.g. > x-lite to x-lite call : video call disconnect after 30 seconds > > X-lite to Zoiper : video call continue, but no video sending. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/1ecae21a/attachment.html From brian at freeswitch.org Thu Jan 8 22:38:02 2015 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2015 13:38:02 -0600 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: Sounds like xlite is giving its public ip, disable that On Thursday, January 8, 2015, Michael Jerris wrote: > This is some sort of nat issue.. we are sending the 200 OK and not getting > an ack back. Double check all your nat related configuration. > > On Jan 8, 2015, at 10:17 AM, Shisheer Teli > wrote: > > Kindly find the attached "sofia global siptrace on" logs .. > > where : > 10.1.1.1 -> freeswitch server > and > 10.1.1.4 and 10.1.1.3 x-lite clients > > Regards, > Shisheer T > > On Thu, Jan 8, 2015 at 8:25 PM, Brian West > wrote: > >> do 'sofia global siptrace on', so we can see the sip trace. >> >> On Thu, Jan 8, 2015 at 8:52 AM, Shisheer Teli > > wrote: >> >>> Kindly find the attached log details .. >>> >>> Regards, >>> Shisheer T >>> >>> On Thu, Jan 8, 2015 at 8:12 PM, Brian West >> > wrote: >>> >>>> What does the sip traffic say? >>>> >>>> On Thu, Jan 8, 2015 at 4:52 AM, Shisheer Teli >>> > wrote: >>>> >>>>> i am using FreeSWITCH Version >>>>> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >>>>> 03:40:22Z 64bit) >>>>> >>>>> and still video call disconnect after 30 seconds >>>>> >>>>> Regards, >>>>> Shisheer T >>>>> >>>>> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >>>>> anthony.minessale at gmail.com >>>>> > wrote: >>>>> >>>>>> Try latest master or 1.4.15 >>>>>> >>>>>> >>>>>> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >>>>> > wrote: >>>>>> >>>>>>> Hi Team, >>>>>>> >>>>>>> I don't know what happen , but when I start video call it >>>>>>> disconnected after every 30 seconds. >>>>>>> >>>>>>> e.g. >>>>>>> x-lite to x-lite call : video call disconnect after 30 seconds >>>>>>> >>>>>>> X-lite to Zoiper : video call continue, but no video sending. >>>>>>> >>>>>>> > -- *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/20150108/3e830b52/attachment-0001.html From aqsyounas at gmail.com Thu Jan 8 22:50:09 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 9 Jan 2015 00:50:09 +0500 Subject: [Freeswitch-users] how can i do stream piping in freeswitch In-Reply-To: References: <54ADDBF2.50106@mst.edu> Message-ID: Guys, i have another question from you. Is there anyway to find out how many users are listening on a particular stream.Actually this will help me to stop any stream from running locally if no user is listening to it. On 8 January 2015 at 20:44, Aqs Younas wrote: > Thank you guys for your valuable suggestions. > > It really helped me. > > > > On 8 January 2015 at 20:22, Aqs Younas wrote: > >> Actually we are running a radio setup in which users listen to different >> radio's streams by pressing the desired extension for that radio. >> >> We have more than 400 radio streams and concurrent peak of 350 calls. >> For example, 100 users are listening to same station and after couple of >> seconds they will be listening to different station. Currently all these >> connections are being made with stream provider, i.e, 100 connections will >> be open with stream provider. >> >> We want is that if a user is listening to a specific station and another >> user also wants to listen that station he(second user) must share the >> stream or connection created by the first user instead of creating a >> separate connection with stream provider. >> >> Thanks for you time. >> >> On 8 January 2015 at 19:56, Brian West wrote: >> >>> You're going to need 400 local streams, or to re-think you approach. >>> >>> On Thu, Jan 8, 2015 at 8:47 AM, Aqs Younas wrote: >>> >>>> Thanks for your suggestions. It really helped. >>>> >>>> But i have few questions. >>>> >>>> What if i have multiple streams actually about 400 streams and a lot >>>> of users who can switch between different streams, if i put all stream in a >>>> file then users will not be able to switch between desired streams. >>>> >>>> And if i create separate folder each containing a file with single >>>> stream then there will be a lots of directories and for every directory and >>>> also i have to manually enter its name in local_stream.conf.xml >>>> >>>> Because i see this setup suitable for only single stream or in a >>>> scenario where users don't want to switch between desired streams by >>>> pressing extension. >>>> >>>> Is there any other way to do so.? Or how can i improve this? >>>> >>>> Really thankful for your help. >>>> >>>> Regards. >>>> >>>> On 8 January 2015 at 06:22, Nathan Neulinger wrote: >>>> >>>>> Look at the Shout stream as MOH section of: >>>>> >>>>> https://wiki.freeswitch.org/wiki/Mod_shout >>>>> >>>>> You have to define the stream as a local_stream and then reference it >>>>> via local_stream://moh/whatever >>>>> >>>>> I use this to stream a local campus radio station via mp3/icecast as >>>>> MOH stream. >>>>> >>>>> -- Nathan >>>>> >>>>> On 01/07/2015 03:21 PM, Aqs Younas wrote: >>>>> > Currently i am playing a stream with mod_shout and this is my >>>>> default xml. >>>>> > >>>>> > >>>>> > >>>>> > >>>> expression="^14049002000$"> >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Every time a user makes a call for stream it opens a separate >>>>> connection with the stream provider. If 100 users dials >>>>> > this number, there would be 100 connections with stream provider >>>>> listening to same stream, means more rtp packets >>>>> > containing same data for different users. >>>>> > >>>>> > What i want is, if a user is listening to stream then other users >>>>> must share the same listening connection that the >>>>> > first user is opened, instead of creating a separate connection with >>>>> stream provider for same stream. >>>>> > >>>>> > Someone told me this is possible in asterisk, so there must be a way >>>>> in freeswitch. >>>>> > >>>>> > How can i do this.? >>>>> > Any help would be much appreciated. >>>>> > >>>>> > Thanks >>>>> > >>>>> > >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > 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 >>>>> > >>>>> >>>>> -- >>>>> ------------------------------------------------------------ >>>>> Nathan Neulinger nneul at mst.edu >>>>> Missouri S&T Information Technology (573) 612-1412 >>>>> System Administrator - Architect >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/fbcf3993/attachment.html From sertys at gmail.com Thu Jan 8 22:53:39 2015 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 8 Jan 2015 20:53:39 +0100 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: The call disconnects are almost every time a NAT issue and a timer expiring on call. Look up on that. Maybe ICE or stun issues. 8 ???. 2015 ?. 21:39 ???????????? "Brian West" ???????: > Sounds like xlite is giving its public ip, disable that > > On Thursday, January 8, 2015, Michael Jerris wrote: > >> This is some sort of nat issue.. we are sending the 200 OK and not >> getting an ack back. Double check all your nat related configuration. >> >> On Jan 8, 2015, at 10:17 AM, Shisheer Teli wrote: >> >> Kindly find the attached "sofia global siptrace on" logs .. >> >> where : >> 10.1.1.1 -> freeswitch server >> and >> 10.1.1.4 and 10.1.1.3 x-lite clients >> >> Regards, >> Shisheer T >> >> On Thu, Jan 8, 2015 at 8:25 PM, Brian West wrote: >> >>> do 'sofia global siptrace on', so we can see the sip trace. >>> >>> On Thu, Jan 8, 2015 at 8:52 AM, Shisheer Teli >>> wrote: >>> >>>> Kindly find the attached log details .. >>>> >>>> Regards, >>>> Shisheer T >>>> >>>> On Thu, Jan 8, 2015 at 8:12 PM, Brian West >>>> wrote: >>>> >>>>> What does the sip traffic say? >>>>> >>>>> On Thu, Jan 8, 2015 at 4:52 AM, Shisheer Teli >>>>> wrote: >>>>> >>>>>> i am using FreeSWITCH Version >>>>>> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >>>>>> 03:40:22Z 64bit) >>>>>> >>>>>> and still video call disconnect after 30 seconds >>>>>> >>>>>> Regards, >>>>>> Shisheer T >>>>>> >>>>>> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >>>>>> anthony.minessale at gmail.com> wrote: >>>>>> >>>>>>> Try latest master or 1.4.15 >>>>>>> >>>>>>> >>>>>>> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >>>>>>> wrote: >>>>>>> >>>>>>>> Hi Team, >>>>>>>> >>>>>>>> I don't know what happen , but when I start video call it >>>>>>>> disconnected after every 30 seconds. >>>>>>>> >>>>>>>> e.g. >>>>>>>> x-lite to x-lite call : video call disconnect after 30 seconds >>>>>>>> >>>>>>>> X-lite to Zoiper : video call continue, but no video sending. >>>>>>>> >>>>>>>> >> > > -- > > *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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/06a5a89d/attachment-0001.html From GeorgePhelps at gfphelps.com Fri Jan 9 02:10:43 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Thu, 8 Jan 2015 18:10:43 -0500 Subject: [Freeswitch-users] P-Asserted-Identity: Message-ID: <0ed801d02b98$546e4300$fd4ac900$@gfphelps.com> Inside my dialplan, I have: In the INVITE SIP trace, I see: P-Asserted-Identity: I.e., the data portion of my original character string - "tel:+1404XXXXXXX" - has been encapsulated by less-than ("<") and greater-than (">") characters. How do I code my dialplan to get. P-Asserted-Identity: tel:+1404XXXXXXX .as opposed to the current: P-Asserted-Identity: Thanks, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150108/351146ef/attachment.html From dujinfang at gmail.com Fri Jan 9 02:24:09 2015 From: dujinfang at gmail.com (Seven Du) Date: Fri, 9 Jan 2015 07:24:09 +0800 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: Message-ID: <24EA0E57C97E4003A10704499F21E931@gmail.com> perhaps report a jira with debug level log and sofia global siptrace on -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, January 8, 2015 at 6:52 PM, Shisheer Teli wrote: > i am using FreeSWITCH Version 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 03:40:22Z 64bit) > > and still video call disconnect after 30 seconds > > Regards, > Shisheer T > > > On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale wrote: > > Try latest master or 1.4.15 > > > > > > On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli wrote: > > > Hi Team, > > > > > > I don't know what happen , but when I start video call it disconnected after every 30 seconds. > > > > > > e.g. > > > x-lite to x-lite call : video call disconnect after 30 seconds > > > > > > X-lite to Zoiper : video call continue, but no video sending. > > > > > > > > > Regards > > > Shisheer T > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto: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 (mailto: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 > > > > > > > > -- > > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > > > ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > > ? irc.freenode.net (http://irc.freenode.net) #freeswitch ? http://freeswitch.org/g+ > > > > ClueCon Weekly Development Call > > ? sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org) ? +19193869900 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto: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 (mailto: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 > > > > -- > Regards,Shisheer Teli > Phone: +91-022 2278 2519 / 2121 > shisheer at tifr.res.in (http://tifr.res.in/) > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto: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 (mailto: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 > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/ac5b7c0b/attachment.html From telishisheer at gmail.com Fri Jan 9 09:12:44 2015 From: telishisheer at gmail.com (Shisheer Teli) Date: Fri, 9 Jan 2015 11:42:44 +0530 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: <24EA0E57C97E4003A10704499F21E931@gmail.com> References: <24EA0E57C97E4003A10704499F21E931@gmail.com> Message-ID: I did following changes but still video call disconnect after 30 seconds, audio call working properly. Attched log file vars.xml host:domain.example.com is another possible value; however this will not toggle the autonat flags. If you are behind NAT, with dynamic DNS (and stun doesn't work) you should write a script that determines your public IP address, makes the change and calls reloadxml. This also holds true for the external profile. No special processing happens to determine the IP address before the variable gets passed to the external profile. host:domain.example.com may be used in places "where you have two interfaces in a box and one is public facing and one isn't, so one never has to tell the lies." - source bwk on irc. internal.xml external.xml On Fri, Jan 9, 2015 at 4:54 AM, Seven Du wrote: > perhaps report a jira with debug level log and > > sofia global siptrace on > > > -- > Seven Du > http://about.me/dujinfang > http://www.dujinfang.com > http://www.freeswitch.org.cn > > Sent with Sparrow > > On Thursday, January 8, 2015 at 6:52 PM, Shisheer Teli wrote: > > i am using FreeSWITCH Version > 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 > 03:40:22Z 64bit) > > and still video call disconnect after 30 seconds > > Regards, > Shisheer T > > On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > Try latest master or 1.4.15 > > > On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli > wrote: > > Hi Team, > > I don't know what happen , but when I start video call it disconnected > after every 30 seconds. > > e.g. > x-lite to x-lite call : video call disconnect after 30 seconds > > X-lite to Zoiper : video call continue, but no video sending. > > > Regards > Shisheer T > > _________________________________________________________________________ > 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 > > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > 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 > > > > > -- > Regards, > Shisheer Teli > Phone: +91-022 2278 2519 / 2121 > shisheer at tifr.res.in > _________________________________________________________________________ > 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 > -- Regards, Shisheer Teli Phone: +91-022 2278 2519 / 2121 shisheer at tifr.res.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/0ee45457/attachment-0001.html -------------- next part -------------- freeswitch at RHEL62> 2015-01-09 11:35:30.418894 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1001 at 10.1.1.1 [83d838fa-97c5-11e4-8ea3-cbf2cf7962b1] 2015-01-09 11:35:30.438892 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->1003 in context default 2015-01-09 11:35:30.438892 [CONSOLE] sofia_presence.c:1618 Event Thread Started 2015-01-09 11:35:30.438892 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2015-01-09 11:35:30.438892 [CRIT] mod_dptools.c:1628 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2015-01-09 11:35:30.438892 [CRIT] mod_dptools.c:1628 Once changed type 'reloadxml' at the console. 2015-01-09 11:35:30.438892 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2015-01-09 11:35:40.478894 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *1 execute_extension::dx XML features 2015-01-09 11:35:40.478894 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1001.2015-01-09-11-35-40.wav 2015-01-09 11:35:40.478894 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *3 execute_extension::cf XML features 2015-01-09 11:35:40.478894 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *4 execute_extension::att_xfer XML features 2015-01-09 11:35:40.478894 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:1003 at 10.1.1.3:21756 [89d767a8-97c5-11e4-8ec7-cbf2cf7962b1] 2015-01-09 11:35:40.978940 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:1003 at 10.1.1.3:21756! 2015-01-09 11:35:40.998899 [INFO] switch_ivr_originate.c:1192 Sending early media 2015-01-09 11:35:41.298884 [NOTICE] switch_core_media.c:4405 sofia/internal/1001 at 10.1.1.1 Starting Video thread 2015-01-09 11:35:41.298884 [INFO] switch_core_media.c:5723 Activating VIDEO RTCP PORT 0 mux -1 2015-01-09 11:35:41.298884 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1001 at 10.1.1.1! 2015-01-09 11:35:42.818902 [NOTICE] switch_core_media.c:4405 sofia/internal/sip:1003 at 10.1.1.3:21756 Starting Video thread 2015-01-09 11:35:42.818902 [INFO] switch_core_media.c:5723 Activating VIDEO RTCP PORT 0 mux -1 2015-01-09 11:35:42.818902 [NOTICE] sofia.c:7475 Channel [sofia/internal/sip:1003 at 10.1.1.3:21756] has been answered 2015-01-09 11:35:42.838894 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/1001 at 10.1.1.1] has been answered 2015-01-09 11:35:42.918913 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:42.937179] SSRC[437272288]RTT[5988.191940] A[3996577770] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:44.238898 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:44.248887] SSRC[437272288]RTT[5989.503662] A[3996663735] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:44.918951 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:44.937213] SSRC[437272288]RTT[5990.191986] A[3996708845] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:46.038901 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:46.40917] SSRC[437272288]RTT[5991.295685] A[3996781177] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:46.918951 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:46.938125] SSRC[437272288]RTT[5992.192886] A[3996839976] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:48.438898 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:48.442348] SSRC[437272288]RTT[5993.697113] A[3996938557] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:49.438897 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:49.456462] SSRC[437272288]RTT[5994.711227] A[3997005018] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:49.938905 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:49.945723] SSRC[437272288]RTT[5995.200485] A[3997037082] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:50.838903 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:50.840488] SSRC[437272288]RTT[5996.095261] A[3997095722] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:51.318892 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:51.330626] SSRC[789999412]RTT[17543.483673] A[3997127843] - DLSR[1732800362] - LSR[1114597735] 2015-01-09 11:35:51.538903 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:51.545053] SSRC[437272288]RTT[5996.799820] A[3997141896] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:52.938898 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:52.954654] SSRC[437272288]RTT[5998.209427] A[3997234276] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:54.058900 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:54.63639] SSRC[437272288]RTT[5999.318405] A[3997306954] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:55.338888 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:55.352605] SSRC[437272288]RTT[6000.607376] A[3997391428] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:56.438889 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:56.456837] SSRC[437272288]RTT[6001.711609] A[3997463795] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:57.338947 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:57.352705] SSRC[437272288]RTT[6002.607468] A[3997522506] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:58.238900 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:58.247475] SSRC[437272288]RTT[6003.502243] A[3997581146] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:35:59.338898 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:05:59.352233] SSRC[437272288]RTT[6004.606995] A[3997653547] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:00.438899 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:00.454995] SSRC[437272288]RTT[6005.709763] A[3997725818] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:01.318905 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:01.330992] SSRC[789999412]RTT[17553.484039] A[3997783227] - DLSR[1732800362] - LSR[1114597735] 2015-01-09 11:36:01.458900 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:01.463167] SSRC[437272288]RTT[6006.717941] A[3997791890] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:02.458899 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:02.459957] SSRC[437272288]RTT[6007.714722] A[3997857215] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:03.558931 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:03.560177] SSRC[437272288]RTT[6008.814941] A[3997929319] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:04.038897 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:04.55052] SSRC[437272288]RTT[6009.309814] A[3997961751] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:05.158939 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:05.161244] SSRC[437272288]RTT[6010.416016] A[3998034247] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:06.038900 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:06.56440] SSRC[437272288]RTT[6011.311203] A[3998092914] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:06.858901 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:06.869711] SSRC[437272288]RTT[6012.124481] A[3998146213] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:08.178902 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:08.196115] SSRC[437272288]RTT[6013.450882] A[3998233140] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:09.138898 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:09.145584] SSRC[437272288]RTT[6014.400345] A[3998295364] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:09.618898 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:09.629768] SSRC[437272288]RTT[6014.884537] A[3998327096] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:10.738904 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:10.757128] SSRC[437272288]RTT[6016.011902] A[3998400979] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:11.218901 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:11.226911] SSRC[437272288]RTT[6016.481674] A[3998431766] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:11.318906 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:11.336547] SSRC[789999412]RTT[17563.489594] A[3998438951] - DLSR[1732800362] - LSR[1114597735] 2015-01-09 11:36:12.018920 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:12.26310] SSRC[437272288]RTT[6017.281082] A[3998484156] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:12.818942 [NOTICE] switch_rtp.c:5168 Receiving an RTCP packet[2015-09-00 06:06:12.825843] SSRC[437272288]RTT[6018.080612] A[3998536554] - DLSR[1970041170] - LSR[1634094453] 2015-01-09 11:36:12.898901 [NOTICE] sofia.c:952 Hangup sofia/internal/sip:1003 at 10.1.1.3:21756 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-09 11:36:12.918915 [NOTICE] switch_ivr_bridge.c:1608 Hangup sofia/internal/1001 at 10.1.1.1 [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-09 11:36:12.918915 [NOTICE] switch_core_session.c:1633 Session 4 (sofia/internal/sip:1003 at 10.1.1.3:21756) Ended 2015-01-09 11:36:12.918915 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:1003 at 10.1.1.3:21756 [CS_DESTROY] 2015-01-09 11:36:12.938895 [NOTICE] switch_core_session.c:1633 Session 3 (sofia/internal/1001 at 10.1.1.1) Ended 2015-01-09 11:36:12.938895 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/1001 at 10.1.1.1 [CS_DESTROY] From myforums.indra at gmail.com Fri Jan 9 13:54:30 2015 From: myforums.indra at gmail.com (indra sena) Date: Fri, 9 Jan 2015 16:24:30 +0530 Subject: [Freeswitch-users] REg IVR and codec negotiation In-Reply-To: References: Message-ID: Hi All, Do any body have any suggestion on this ? I have observed in freeswitch that, In case of IVR scenario prior to the bridge FreeSwitch plays IVR with the 1st priority codec then invite to the bridge endpoint (B-leg) with SDP containing only the codec negotiated for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only with IVR negotiated codec. I am having one issue here for example. Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated with G729 and starts playing IVR with G729 and answered(183) to originator with SDP containing only G729. And Invited to termination endpoint with SDP having only G729 and termination gateway having support of only G711, and it is rejecting call after receiving invite with only G729. It should be work like after IVR , invite can be having sdp with all the originator supported codecs and if termination (b-leg) having different priority codecs then it can send re-Invite sanding the same. I have observed one issue in freeswitch Jeera ( https://freeswitch.org/jira/browse/FS-880) , but there is no solution in that page. Do we have any solution for this right now ? Your answers will be very much helpful for me. I appreciate if you can quick response or give some solution for this. Thanks & Reagrds, GISR.. On Thu, Jan 8, 2015 at 12:39 PM, indra sena wrote: > Hi , > > I have observed in freeswitch that, In case of IVR scenario prior to the > bridge FreeSwitch plays IVR with the 1st priority codec then invite to the > bridge endpoint (B-leg) with SDP containing only the codec negotiated for > IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only > with IVR negotiated codec. > > I am having one issue here for example. > > Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated > with G729 and starts playing IVR with G729 and answered(183) to originator > with SDP containing only G729. And Invited to termination endpoint with SDP > having only G729 and termination gateway having support of only G711, and > it is rejecting call after receiving invite with only G729. > > It should be work like after IVR , invite can be having sdp with all the > originator supported codecs and if termination (b-leg) having different > priority codecs then it can send re-Invite sanding the same. > > I have observed one issue in freeswitch Jeera ( > https://freeswitch.org/jira/browse/FS-880) , but there is no solution in > that page. > > Do we have any solution for this right now ? > > Your answers will be very much helpful for me. I appreciate if you can > quick response or give some solution for this. > > Thanks in advance. > > Thanks & Regards, > GISR.. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/0cdcfab2/attachment.html From steveayre at gmail.com Fri Jan 9 14:02:11 2015 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 9 Jan 2015 11:02:11 +0000 Subject: [Freeswitch-users] REg IVR and codec negotiation In-Reply-To: References: Message-ID: Default behaviour should be that it offers all codecs to the gateway and transcodes G729-G711 if required. Check outbound-codec-prefs on the profile you're sending to the gateway from? Check it includes both G729 and G711. Check disable-transcoding is not set to true If you're using absolute_codec_string check it's including G711, this would override any other settings On 9 January 2015 at 10:54, indra sena wrote: > Hi All, > > Do any body have any suggestion on this ? > > > I have observed in freeswitch that, In case of IVR scenario prior to the > bridge FreeSwitch plays IVR with the 1st priority codec then invite to the > bridge endpoint (B-leg) with SDP containing only the codec negotiated for > IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only > with IVR negotiated codec. > > I am having one issue here for example. > > Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated > with G729 and starts playing IVR with G729 and answered(183) to originator > with SDP containing only G729. And Invited to termination endpoint with SDP > having only G729 and termination gateway having support of only G711, and > it is rejecting call after receiving invite with only G729. > > It should be work like after IVR , invite can be having sdp with all the > originator supported codecs and if termination (b-leg) having different > priority codecs then it can send re-Invite sanding the same. > > I have observed one issue in freeswitch Jeera ( > https://freeswitch.org/jira/browse/FS-880) , but there is no solution in > that page. > > Do we have any solution for this right now ? > > Your answers will be very much helpful for me. I appreciate if you can > quick response or give some solution for this. > > Thanks & Reagrds, > GISR.. > > On Thu, Jan 8, 2015 at 12:39 PM, indra sena > wrote: > >> Hi , >> >> I have observed in freeswitch that, In case of IVR scenario prior to the >> bridge FreeSwitch plays IVR with the 1st priority codec then invite to the >> bridge endpoint (B-leg) with SDP containing only the codec negotiated for >> IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only >> with IVR negotiated codec. >> >> I am having one issue here for example. >> >> Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated >> with G729 and starts playing IVR with G729 and answered(183) to originator >> with SDP containing only G729. And Invited to termination endpoint with SDP >> having only G729 and termination gateway having support of only G711, and >> it is rejecting call after receiving invite with only G729. >> >> It should be work like after IVR , invite can be having sdp with all the >> originator supported codecs and if termination (b-leg) having different >> priority codecs then it can send re-Invite sanding the same. >> >> I have observed one issue in freeswitch Jeera ( >> https://freeswitch.org/jira/browse/FS-880) , but there is no solution in >> that page. >> >> Do we have any solution for this right now ? >> >> Your answers will be very much helpful for me. I appreciate if you can >> quick response or give some solution for this. >> >> Thanks in advance. >> >> Thanks & Regards, >> GISR.. >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/8d5add66/attachment.html From giggsey at gmail.com Fri Jan 9 13:33:51 2015 From: giggsey at gmail.com (Joshua Gigg) Date: Fri, 9 Jan 2015 10:33:51 +0000 Subject: [Freeswitch-users] Event Socket: CHANNEL_DESTROY for a instantly Failing Call not containing custom channel variables Message-ID: Hi, I'm using the Inbound Event Socket to listen to events. When I try to originate a call that fails instantly, the CHANNEL_DESTROY event I receive (which is the first event for that Call) does not seem to contain the custom variables I passed in the origination command. E.g. - Subscribe to all events - originate {myFirstVar=bla}sofia/external/ &park; This errors straight away in the CLI, but event wise: {"Event-Name":"CHANNEL_DESTROY","Core-UUID":"0bdb8ae0-95be-11e4-a629-7deec23cf31f","FreeSWITCH-Hostname":" test-fs1.example.com","FreeSWITCH-Switchname":"test-fs1.example.com","FreeSWITCH-IPv4":"203.0.113.1","FreeSWITCH-IPv6":"::1","Event-Date-Local":"2015-01-09 10:07:47","Event-Date-GMT":"Fri, 09 Jan 2015 10:07:47 GMT","Event-Date-Timestamp":"1420798067526747","Event-Calling-File":"switch_core_session.c","Event-Calling-Function":"switch_core_session_perform_destroy","Event-Calling-Line-Number":"1465","Event-Sequence":"43492","Channel-State":"CS_NONE","Channel-Call-State":"DOWN","Channel-State-Number":"0","Channel-Name":"N/A","Unique-ID":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","Call-Direction":"outbound","Presence-Call-Direction":"outbound","Channel-HIT-Dialplan":"false","Channel-Call-UUID":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","Answer-State":"ringing","variable_direction":"outbound","variable_is_outbound":"true","variable_uuid":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","variable_call_uuid":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","variable_session_id":"420"}Content-Length: 880 Content-Type: text/event-json {"Event-Name":"CHANNEL_STATE","Core-UUID":"0bdb8ae0-95be-11e4-a629-7deec23cf31f","FreeSWITCH-Hostname":" test-fs1.example.com","FreeSWITCH-Switchname":"test-fs1.example.com","FreeSWITCH-IPv4":"203.0.113.1","FreeSWITCH-IPv6":"::1","Event-Date-Local":"2015-01-09 10:07:47","Event-Date-GMT":"Fri, 09 Jan 2015 10:07:47 GMT","Event-Date-Timestamp":"1420798067526747","Event-Calling-File":"switch_channel.c","Event-Calling-Function":"switch_channel_perform_set_running_state","Event-Calling-Line-Number":"2193","Event-Sequence":"43493","Channel-State":"CS_DESTROY","Channel-Call-State":"DOWN","Channel-State-Number":"0","Channel-Name":"N/A","Unique-ID":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","Call-Direction":"outbound","Presence-Call-Direction":"outbound","Channel-HIT-Dialplan":"false","Channel-Call-UUID":"5ca1cbc6-97e7-11e4-b033-7deec23cf31f","Answer-State":"ringing"}Content-Length: 608 Content-Type: text/event-json Ideally, I shouldn't be making calls to numbers that would fail like this, but shouldn't I still be getting the variables I send across? -- Joshua Gigg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/6c3ffacd/attachment.html From questions.looking.for.answers at gmail.com Fri Jan 9 13:52:17 2015 From: questions.looking.for.answers at gmail.com (Question Answer) Date: Fri, 9 Jan 2015 11:52:17 +0100 Subject: [Freeswitch-users] Dialplan sip chat with SIPml5: client receives message, but freeswitch reports 'Error! Message Not Sent' Message-ID: Hi, I have successfully configured a SIPml5 client in Chrome to communicate with a local FreeSWITCH server. It is possible to successfully make calls, and use most applications in the freeswitch dial plan. Problem: I cannot figure out how to successfully send a custom SIP message from FreeSWITCH [via either the CLI or the dialplan] to the web client. The client actually *receives* the message, but FreeSWITCH then hangs for about 10s, times out and reports: 'Error! Message Not Sent Sent'. I run the following command from the CLI: freeswitch> chat sip|1001 at 192.168.59.103|1000 at 172.17.0.19|test test test send 764 bytes to ws/[192.168.59.3]:51933 at 09:48:10.190253: ------------------------------------------------------------------------ MESSAGE sip:1000 at 192.168.59.3:51933 SIP/2.0 Via: SIP/2.0/WS 172.17.0.19:5066;branch=z9hG4bKFpeyg98ZZ240r Route: ;rtcweb-breaker=yes;transport=ws Max-Forwards: 70 From: "1001" ;tag=U9prcNXypBBUr To: ;rtcweb-breaker=yes;transport=ws Call-ID: 5b3c7d7c-8149-422c-9f0f-f1b3dd3661f2 CSeq: 70061165 MESSAGE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150106T091916Z~f48ec61d54~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Type: text/html Content-Length: 14 X-FS-Sending-Message: 4a972b6a-3c72-4ba1-b7a4-de2d07760380 test test test ------------------------------------------------------------------------ [~10 seconds pass...] Error! Message Not Sent ... freeswitch> -------------------------------------------------------------------------------- But on the client side, in the Chrome developer console, I can see the following log messages: SIPml-api.js?svn=222:1 __tsip_transport_ws_onmessage 2015-01-09 11:37:09.647SIPml-api.js?svn=222:1 recv=MESSAGE sip:1000 at 192.168.59.3:52190 SIP/2.0 Via: SIP/2.0/WS 172.17.0.19:5066;branch=z9hG4bKKtK1QNcem6XBQ From: "1001";tag=29gZHme856Fmc To: ;rtcweb-breaker=yes;transport=ws Contact: Call-ID: 210f6e1f-0415-450b-bbb3-7725a6a37434 CSeq: 70061168 MESSAGE Content-Type: text/html Content-Length: 14 Route: Max-Forwards: 70 User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150106T091916Z~f48ec61d54~64bit Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE Supported: timer,path,replaces X-FS-Sending-Message: 4a972b6a-3c72-4ba1-b7a4-de2d07760380 test test test 2015-01-09 11:37:09.647SIPml-api.js?svn=222:1 State machine: tsip_dialog_generic_Started_2_Incoming_X_iMessage 2015-01-09 11:37:09.649SIPml-api.js?svn=222:1 ==stack event = i_new_message XXX test test test 2015-01-09 11:37:09.649call.htm:964 test test test ---------------------------------------------------------------------------------- That is, the message is correctly received by the client, but FreeSWITCH doesn't seem to 'realise' this. Perhaps I need to be sending some kind explicit confirmation back from the simpml5 client application? Or perhaps there is some kind of mismatch with the SIP profile? I have set the freeswitch log level to Debug, but the above is the only message I see, which is not particularly helpful. Any insights would be appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/59c7d7b1/attachment-0001.html From myforums.indra at gmail.com Fri Jan 9 14:22:07 2015 From: myforums.indra at gmail.com (indra sena) Date: Fri, 9 Jan 2015 16:52:07 +0530 Subject: [Freeswitch-users] REg IVR and codec negotiation In-Reply-To: References: Message-ID: Hi Steven, Thanks for your quick response and valuable suggestions. With out transcoding is there any way to achieve this ? Like after receiving b-leg codec as G711 can we switch from G729 codec to G711 in A-leg by sending reinvite/update to A-leg with SDP with G711 ? Is this procedure works ? currently is freeswitch supports this ? I have one below query. After IVR , while giving answer to the A-leg why it is putting only one codec (with which it has playing IVR), not putting all the allowables codecs by A-leg ? Thanks in advance. Thanks, GISR. On Fri, Jan 9, 2015 at 4:32 PM, Steven Ayre wrote: > Default behaviour should be that it offers all codecs to the gateway and > transcodes G729-G711 if required. > > Check outbound-codec-prefs on the profile you're sending to the gateway > from? Check it includes both G729 and G711. > > Check disable-transcoding is not set to true > > If you're using absolute_codec_string check it's including G711, this > would override any other settings > > > > On 9 January 2015 at 10:54, indra sena wrote: > >> Hi All, >> >> Do any body have any suggestion on this ? >> >> >> I have observed in freeswitch that, In case of IVR scenario prior to the >> bridge FreeSwitch plays IVR with the 1st priority codec then invite to the >> bridge endpoint (B-leg) with SDP containing only the codec negotiated for >> IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only >> with IVR negotiated codec. >> >> I am having one issue here for example. >> >> Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated >> with G729 and starts playing IVR with G729 and answered(183) to originator >> with SDP containing only G729. And Invited to termination endpoint with SDP >> having only G729 and termination gateway having support of only G711, and >> it is rejecting call after receiving invite with only G729. >> >> It should be work like after IVR , invite can be having sdp with all the >> originator supported codecs and if termination (b-leg) having different >> priority codecs then it can send re-Invite sanding the same. >> >> I have observed one issue in freeswitch Jeera ( >> https://freeswitch.org/jira/browse/FS-880) , but there is no solution in >> that page. >> >> Do we have any solution for this right now ? >> >> Your answers will be very much helpful for me. I appreciate if you can >> quick response or give some solution for this. >> >> Thanks & Reagrds, >> GISR.. >> >> On Thu, Jan 8, 2015 at 12:39 PM, indra sena >> wrote: >> >>> Hi , >>> >>> I have observed in freeswitch that, In case of IVR scenario prior to >>> the bridge FreeSwitch plays IVR with the 1st priority codec then invite to >>> the bridge endpoint (B-leg) with SDP containing only the codec negotiated >>> for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing >>> only with IVR negotiated codec. >>> >>> I am having one issue here for example. >>> >>> Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated >>> with G729 and starts playing IVR with G729 and answered(183) to originator >>> with SDP containing only G729. And Invited to termination endpoint with SDP >>> having only G729 and termination gateway having support of only G711, and >>> it is rejecting call after receiving invite with only G729. >>> >>> It should be work like after IVR , invite can be having sdp with all the >>> originator supported codecs and if termination (b-leg) having different >>> priority codecs then it can send re-Invite sanding the same. >>> >>> I have observed one issue in freeswitch Jeera ( >>> https://freeswitch.org/jira/browse/FS-880) , but there is no solution >>> in that page. >>> >>> Do we have any solution for this right now ? >>> >>> Your answers will be very much helpful for me. I appreciate if you can >>> quick response or give some solution for this. >>> >>> Thanks in advance. >>> >>> Thanks & Regards, >>> GISR.. >>> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/2cf315b8/attachment.html From questions.looking.for.answers at gmail.com Fri Jan 9 15:38:53 2015 From: questions.looking.for.answers at gmail.com (Question Answer) Date: Fri, 9 Jan 2015 13:38:53 +0100 Subject: [Freeswitch-users] Dialplan sip chat with SIPml5: client receives message, but freeswitch reports 'Error! Message Not Sent' In-Reply-To: References: Message-ID: Hi, I solved this. The issue was on the client side. I needed to confirm acceptance of the message on the client side and respond to FreeSWITCH: http://www.sipml5.org/docgen/index.html#acceptMessageFunc After doing that FreeSWITCH no longer hangs, and confirms that the messages was sent successfully in the log. On Fri, Jan 9, 2015 at 11:52 AM, Question Answer < questions.looking.for.answers at gmail.com> wrote: > Hi, > I have successfully configured a SIPml5 client in Chrome to communicate > with a local FreeSWITCH server. > It is possible to successfully make calls, and use most applications in > the freeswitch dial plan. > > Problem: I cannot figure out how to successfully send a custom SIP > message from FreeSWITCH [via either the CLI or the dialplan] to the web > client. > The client actually *receives* the message, but FreeSWITCH then hangs > for about 10s, times out and reports: 'Error! Message Not Sent Sent'. > > I run the following command from the CLI: > > freeswitch> chat sip|1001 at 192.168.59.103|1000 at 172.17.0.19|test test test > send 764 bytes to ws/[192.168.59.3]:51933 at 09:48:10.190253: > ------------------------------------------------------------------------ > MESSAGE sip:1000 at 192.168.59.3:51933 SIP/2.0 > Via: SIP/2.0/WS 172.17.0.19:5066;branch=z9hG4bKFpeyg98ZZ240r > Route: ;rtcweb-breaker=yes;transport=ws > Max-Forwards: 70 > From: "1001" ;tag=U9prcNXypBBUr > To: ;rtcweb-breaker=yes;transport=ws > Call-ID: 5b3c7d7c-8149-422c-9f0f-f1b3dd3661f2 > CSeq: 70061165 MESSAGE > Contact: > User-Agent: > FreeSWITCH-mod_sofia/1.5.15b+git~20150106T091916Z~f48ec61d54~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Content-Type: text/html > Content-Length: 14 > X-FS-Sending-Message: 4a972b6a-3c72-4ba1-b7a4-de2d07760380 > > test test test > ------------------------------------------------------------------------ > [~10 seconds pass...] > Error! Message Not Sent > ... > freeswitch> > > > -------------------------------------------------------------------------------- > But on the client side, in the Chrome developer console, I can see the > following log messages: > > SIPml-api.js?svn=222:1 __tsip_transport_ws_onmessage > 2015-01-09 11:37:09.647SIPml-api.js?svn=222:1 recv=MESSAGE > sip:1000 at 192.168.59.3:52190 SIP/2.0 > Via: SIP/2.0/WS 172.17.0.19:5066;branch=z9hG4bKKtK1QNcem6XBQ > From: "1001";tag=29gZHme856Fmc > To: ;rtcweb-breaker=yes;transport=ws > Contact: > Call-ID: 210f6e1f-0415-450b-bbb3-7725a6a37434 > CSeq: 70061168 MESSAGE > Content-Type: text/html > Content-Length: 14 > Route: > Max-Forwards: 70 > User-Agent: > FreeSWITCH-mod_sofia/1.5.15b+git~20150106T091916Z~f48ec61d54~64bit > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE > Supported: timer,path,replaces > X-FS-Sending-Message: 4a972b6a-3c72-4ba1-b7a4-de2d07760380 > > test test test > 2015-01-09 11:37:09.647SIPml-api.js?svn=222:1 State machine: > tsip_dialog_generic_Started_2_Incoming_X_iMessage > 2015-01-09 11:37:09.649SIPml-api.js?svn=222:1 ==stack event = > i_new_message XXX test test test > 2015-01-09 11:37:09.649call.htm:964 test test test > > ---------------------------------------------------------------------------------- > > That is, the message is correctly received by the client, but FreeSWITCH > doesn't seem to 'realise' this. > Perhaps I need to be sending some kind explicit confirmation back from the > simpml5 client application? > > Or perhaps there is some kind of mismatch with the SIP profile? I have > set the freeswitch log level to Debug, but the above is the only message I > see, which is not particularly helpful. > > Any insights would be appreciated! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/b5542217/attachment-0001.html From steveayre at gmail.com Fri Jan 9 16:27:51 2015 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 9 Jan 2015 13:27:51 +0000 Subject: [Freeswitch-users] REg IVR and codec negotiation In-Reply-To: References: Message-ID: > Like after receiving b-leg codec as G711 can we switch from G729 codec to G711 in A-leg by sending reinvite/update to A-leg with SDP with G711 ? Is this procedure works ? currently is freeswitch supports this ? Before the bridge to that gateway you could try renegotiating the aleg codec to G711, but it might not work with every client. https://wiki.freeswitch.org/wiki/Mod_commands#uuid_media_reneg But I'd first look at why the B-leg isn't being offered more codecs. Avoiding transcoding would be more of an optimisation, you still need to look at why the gateway wasn't offered G711. > After IVR , while giving answer to the A-leg why it is putting only one codec (with which it has playing IVR), not putting all the allowables codecs by A-leg ? Because at that point it's negotiated which codec to use for the call. The a-leg sends the list of codecs it supports, freeswitch compares that to its own list and picks the single best match. On 9 January 2015 at 11:22, indra sena wrote: > Hi Steven, > > Thanks for your quick response and valuable suggestions. > > With out transcoding is there any way to achieve this ? > > Like after receiving b-leg codec as G711 can we switch from G729 codec to > G711 in A-leg by sending reinvite/update to A-leg with SDP with G711 ? Is > this procedure works ? currently is freeswitch supports this ? > > I have one below query. > After IVR , while giving answer to the A-leg why it is putting only one > codec (with which it has playing IVR), not putting all the allowables > codecs by A-leg ? > > Thanks in advance. > > Thanks, > GISR. > > > > On Fri, Jan 9, 2015 at 4:32 PM, Steven Ayre wrote: > >> Default behaviour should be that it offers all codecs to the gateway and >> transcodes G729-G711 if required. >> >> Check outbound-codec-prefs on the profile you're sending to the gateway >> from? Check it includes both G729 and G711. >> >> Check disable-transcoding is not set to true >> >> If you're using absolute_codec_string check it's including G711, this >> would override any other settings >> >> >> >> On 9 January 2015 at 10:54, indra sena wrote: >> >>> Hi All, >>> >>> Do any body have any suggestion on this ? >>> >>> >>> I have observed in freeswitch that, In case of IVR scenario prior to >>> the bridge FreeSwitch plays IVR with the 1st priority codec then invite to >>> the bridge endpoint (B-leg) with SDP containing only the codec negotiated >>> for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing >>> only with IVR negotiated codec. >>> >>> I am having one issue here for example. >>> >>> Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated >>> with G729 and starts playing IVR with G729 and answered(183) to originator >>> with SDP containing only G729. And Invited to termination endpoint with SDP >>> having only G729 and termination gateway having support of only G711, and >>> it is rejecting call after receiving invite with only G729. >>> >>> It should be work like after IVR , invite can be having sdp with all the >>> originator supported codecs and if termination (b-leg) having different >>> priority codecs then it can send re-Invite sanding the same. >>> >>> I have observed one issue in freeswitch Jeera ( >>> https://freeswitch.org/jira/browse/FS-880) , but there is no solution >>> in that page. >>> >>> Do we have any solution for this right now ? >>> >>> Your answers will be very much helpful for me. I appreciate if you can >>> quick response or give some solution for this. >>> >>> Thanks & Reagrds, >>> GISR.. >>> >>> On Thu, Jan 8, 2015 at 12:39 PM, indra sena >>> wrote: >>> >>>> Hi , >>>> >>>> I have observed in freeswitch that, In case of IVR scenario prior to >>>> the bridge FreeSwitch plays IVR with the 1st priority codec then invite to >>>> the bridge endpoint (B-leg) with SDP containing only the codec negotiated >>>> for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing >>>> only with IVR negotiated codec. >>>> >>>> I am having one issue here for example. >>>> >>>> Originator(A-leg) sends invite with G729, G711 and Freeswitch >>>> negotiated with G729 and starts playing IVR with G729 and answered(183) to >>>> originator with SDP containing only G729. And Invited to termination >>>> endpoint with SDP having only G729 and termination gateway having support >>>> of only G711, and it is rejecting call after receiving invite with only >>>> G729. >>>> >>>> It should be work like after IVR , invite can be having sdp with all >>>> the originator supported codecs and if termination (b-leg) having different >>>> priority codecs then it can send re-Invite sanding the same. >>>> >>>> I have observed one issue in freeswitch Jeera ( >>>> https://freeswitch.org/jira/browse/FS-880) , but there is no solution >>>> in that page. >>>> >>>> Do we have any solution for this right now ? >>>> >>>> Your answers will be very much helpful for me. I appreciate if you can >>>> quick response or give some solution for this. >>>> >>>> Thanks in advance. >>>> >>>> Thanks & Regards, >>>> GISR.. >>>> >>> >>> >>> _________________________________________________________________________ >>> 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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/9b5305e7/attachment.html From anthony.minessale at gmail.com Fri Jan 9 17:00:10 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Jan 2015 08:00:10 -0600 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: <24EA0E57C97E4003A10704499F21E931@gmail.com> Message-ID: iirc, were asked to make a jira with attached logs. On Friday, January 9, 2015, Shisheer Teli wrote: > I did following changes but still video call disconnect after 30 seconds, > audio call working properly. > > Attched log file > vars.xml > > > > > host:domain.example.com is another possible value; however this will not > toggle the autonat flags. If you are behind NAT, with dynamic DNS (and stun > doesn't work) you should write a script that determines your public IP > address, makes the change and calls reloadxml. This also holds true for the > external profile. No special processing happens to determine the IP address > before the variable gets passed to the external profile. > > host:domain.example.com may be used in places "where you have two > interfaces in a box and one is public facing and one isn't, so one never > has to tell the lies." > > - source bwk on irc. > > internal.xml > > > > external.xml > > > > > > On Fri, Jan 9, 2015 at 4:54 AM, Seven Du > wrote: > >> perhaps report a jira with debug level log and >> >> sofia global siptrace on >> >> >> -- >> Seven Du >> http://about.me/dujinfang >> http://www.dujinfang.com >> http://www.freeswitch.org.cn >> >> Sent with Sparrow >> >> On Thursday, January 8, 2015 at 6:52 PM, Shisheer Teli wrote: >> >> i am using FreeSWITCH Version >> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >> 03:40:22Z 64bit) >> >> and still video call disconnect after 30 seconds >> >> Regards, >> Shisheer T >> >> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >> anthony.minessale at gmail.com >> > wrote: >> >> Try latest master or 1.4.15 >> >> >> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli > > wrote: >> >> Hi Team, >> >> I don't know what happen , but when I start video call it disconnected >> after every 30 seconds. >> >> e.g. >> x-lite to x-lite call : video call disconnect after 30 seconds >> >> X-lite to Zoiper : video call continue, but no video sending. >> >> >> Regards >> Shisheer T >> >> _________________________________________________________________________ >> 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 >> >> >> >> >> -- >> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org >> ? >> +19193869900 >> >> >> _________________________________________________________________________ >> 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 >> >> >> >> >> -- >> Regards, >> Shisheer Teli >> Phone: +91-022 2278 2519 / 2121 >> shisheer at tifr.res.in >> _________________________________________________________________________ >> 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 >> > > > > -- > Regards, > Shisheer Teli > Phone: +91-022 2278 2519 / 2121 > shisheer at tifr.res.in > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/1e0088cd/attachment-0001.html From bee at netempire.de Fri Jan 9 17:01:22 2015 From: bee at netempire.de (bee) Date: Fri, 9 Jan 2015 07:01:22 -0700 (MST) Subject: [Freeswitch-users] Freeswitch only records empty files Message-ID: <1420812082840-7596139.post@n2.nabble.com> Hey, i got an script which makes a call to an outbound number (so only 1 channel is open) and then i want to record the input. But the record function only create empty 44kb files. When i join the channel over a softphone the recorded files have input. Any ideas? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-only-records-empty-files-tp7596139.html Sent from the freeswitch-users mailing list archive at Nabble.com. From wstephen80 at gmail.com Fri Jan 9 17:49:30 2015 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 9 Jan 2015 15:49:30 +0100 Subject: [Freeswitch-users] Freeswitch only records empty files In-Reply-To: <1420812082840-7596139.post@n2.nabble.com> References: <1420812082840-7596139.post@n2.nabble.com> Message-ID: This can happens if you bypass media On Fri, Jan 9, 2015 at 3:01 PM, bee wrote: > Hey, i got an script which makes a call to an outbound number (so only 1 > channel is open) and then i want to record the input. But the record > function only create empty 44kb files. When i join the channel over a > softphone the recorded files have input. > > Any ideas? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-only-records-empty-files-tp7596139.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/88ef4c41/attachment.html From aqsyounas at gmail.com Fri Jan 9 17:55:23 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 9 Jan 2015 19:55:23 +0500 Subject: [Freeswitch-users] Launch a script on errors in freeswitch log file Message-ID: Hi, How can i launch a script when there are error logs in freeswitch log file. Actually i am using mod_shout as MOH with mod_local_stream, when someone enters a url of stream that is not responding i see logs all these. 2015-01-09 19:38:39.338003 [ERR] mod_local_stream.c:214 Can't open shout:// 174.37.194.139:8316 2015-01-09 19:38:41.337988 [ERR] switch_core_file.c:149 Invalid file format [shout] for [174.37.194.139:8316]! 2015-01-09 19:38:42.337988 [ERR] mod_local_stream.c:214 Can't open shout:// s9.voscast.com:7584 2015-01-09 19:38:42.337988 [ERR] switch_core_file.c:149 Invalid file format [shout] for [s9.voscast.com:7584]! I want to run an external script to remove that file or urls when there are errors logs in freeswitch. But i don't know how to trigger that script on errors in freeswitch log. Any help would be much appreciated. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/ad922a6d/attachment.html From myforums.indra at gmail.com Fri Jan 9 18:05:32 2015 From: myforums.indra at gmail.com (indra sena) Date: Fri, 9 Jan 2015 20:35:32 +0530 Subject: [Freeswitch-users] REg IVR and codec negotiation In-Reply-To: References: Message-ID: Hi Steven, Thanks for all your deep insights and bearing with interminable questions. 1) But I'd first look at why the B-leg isn't being offered more codecs. Avoiding transcoding would be more of an optimisation, you still need to look at why the gateway wasn't offered G711. I have found the reason for why gateway wasn't offered G711, because form my dialplan is sending absolute_codec_string as 'ep_codec_string', i.e. "absolute_codec_string=\${ep_codec_string}". It is received by fs as "absolute_codec_string=G729 at 8000h@20i at 8000b,PCMU at 8000h@20i at 64000b" but while parsing with switch_separate_string_string() it only assigning to first one upto first "," (there is a stmt Make sure you have single quotes ( 'PCMA,PCMU' ) around comma ( ',' ) separated list of codecs to protect it from parsing list of variables inside of {var1=val1,var2=val2,absolute_codec_string='GSM,PCMU'}). next cdecs are considering as separate vars and it is considering like this. Q1: So do we need to send all codecs by substituting instead 'ep_codec_string' like absolute_codec_string='G729,PCMU' from dialplan or can we send 'ep_codec_string' any other way from dialplan so that it can parse properly? After hardcoding absolute_codec_string='G729,PCMU' in dialplan , B-leg is getting offer with both G729, G711 and it is responding but due to transcoder is not there there no voice between these two. 2) Before the bridge to that gateway you could try renegotiating the aleg codec to G711, but it might not work with every client. https://wiki.freeswitch.org/wiki/Mod_commands#uuid_media_reneg Q1) As I need to avoid transcoder and IVR should be present. So after playing IVR can we need to renegotiate with A-leb by uuid_media_reneg , can we achieve this functionality through dialplan configuration ? 3Q) FS-880 jeera https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-880/, they mentioned that playing your ringback then using refer "deflect" app to blind transfer the call to the same box in proxy mode.How this will work. Thanks for your suggestions. Regards, GISR. On Fri, Jan 9, 2015 at 6:57 PM, Steven Ayre wrote: > > Like after receiving b-leg codec as G711 can we switch from G729 codec to > G711 in A-leg by sending reinvite/update to A-leg with SDP with G711 ? Is > this procedure works ? currently is freeswitch supports this ? > > Before the bridge to that gateway you could try renegotiating the aleg > codec to G711, but it might not work with every client. > https://wiki.freeswitch.org/wiki/Mod_commands#uuid_media_reneg > But I'd first look at why the B-leg isn't being offered more codecs. > Avoiding transcoding would be more of an optimisation, you still need to > look at why the gateway wasn't offered G711. > > > After IVR , while giving answer to the A-leg why it is putting only one > codec (with which it has playing IVR), not putting all the allowables > codecs by A-leg ? > > Because at that point it's negotiated which codec to use for the call. The > a-leg sends the list of codecs it supports, freeswitch compares that to its > own list and picks the single best match. > > > > > > On 9 January 2015 at 11:22, indra sena wrote: > >> Hi Steven, >> >> Thanks for your quick response and valuable suggestions. >> >> With out transcoding is there any way to achieve this ? >> >> Like after receiving b-leg codec as G711 can we switch from G729 codec >> to G711 in A-leg by sending reinvite/update to A-leg with SDP with G711 ? >> Is this procedure works ? currently is freeswitch supports this ? >> >> I have one below query. >> After IVR , while giving answer to the A-leg why it is putting only one >> codec (with which it has playing IVR), not putting all the allowables >> codecs by A-leg ? >> >> Thanks in advance. >> >> Thanks, >> GISR. >> >> >> >> On Fri, Jan 9, 2015 at 4:32 PM, Steven Ayre wrote: >> >>> Default behaviour should be that it offers all codecs to the gateway and >>> transcodes G729-G711 if required. >>> >>> Check outbound-codec-prefs on the profile you're sending to the gateway >>> from? Check it includes both G729 and G711. >>> >>> Check disable-transcoding is not set to true >>> >>> If you're using absolute_codec_string check it's including G711, this >>> would override any other settings >>> >>> >>> >>> On 9 January 2015 at 10:54, indra sena wrote: >>> >>>> Hi All, >>>> >>>> Do any body have any suggestion on this ? >>>> >>>> >>>> I have observed in freeswitch that, In case of IVR scenario prior to >>>> the bridge FreeSwitch plays IVR with the 1st priority codec then invite to >>>> the bridge endpoint (B-leg) with SDP containing only the codec negotiated >>>> for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing >>>> only with IVR negotiated codec. >>>> >>>> I am having one issue here for example. >>>> >>>> Originator(A-leg) sends invite with G729, G711 and Freeswitch >>>> negotiated with G729 and starts playing IVR with G729 and answered(183) to >>>> originator with SDP containing only G729. And Invited to termination >>>> endpoint with SDP having only G729 and termination gateway having support >>>> of only G711, and it is rejecting call after receiving invite with only >>>> G729. >>>> >>>> It should be work like after IVR , invite can be having sdp with all >>>> the originator supported codecs and if termination (b-leg) having different >>>> priority codecs then it can send re-Invite sanding the same. >>>> >>>> I have observed one issue in freeswitch Jeera ( >>>> https://freeswitch.org/jira/browse/FS-880) , but there is no solution >>>> in that page. >>>> >>>> Do we have any solution for this right now ? >>>> >>>> Your answers will be very much helpful for me. I appreciate if you can >>>> quick response or give some solution for this. >>>> >>>> Thanks & Reagrds, >>>> GISR.. >>>> >>>> On Thu, Jan 8, 2015 at 12:39 PM, indra sena >>>> wrote: >>>> >>>>> Hi , >>>>> >>>>> I have observed in freeswitch that, In case of IVR scenario prior to >>>>> the bridge FreeSwitch plays IVR with the 1st priority codec then invite to >>>>> the bridge endpoint (B-leg) with SDP containing only the codec negotiated >>>>> for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing >>>>> only with IVR negotiated codec. >>>>> >>>>> I am having one issue here for example. >>>>> >>>>> Originator(A-leg) sends invite with G729, G711 and Freeswitch >>>>> negotiated with G729 and starts playing IVR with G729 and answered(183) to >>>>> originator with SDP containing only G729. And Invited to termination >>>>> endpoint with SDP having only G729 and termination gateway having support >>>>> of only G711, and it is rejecting call after receiving invite with only >>>>> G729. >>>>> >>>>> It should be work like after IVR , invite can be having sdp with all >>>>> the originator supported codecs and if termination (b-leg) having different >>>>> priority codecs then it can send re-Invite sanding the same. >>>>> >>>>> I have observed one issue in freeswitch Jeera ( >>>>> https://freeswitch.org/jira/browse/FS-880) , but there is no solution >>>>> in that page. >>>>> >>>>> Do we have any solution for this right now ? >>>>> >>>>> Your answers will be very much helpful for me. I appreciate if you can >>>>> quick response or give some solution for this. >>>>> >>>>> Thanks in advance. >>>>> >>>>> Thanks & Regards, >>>>> GISR.. >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/f0ac33eb/attachment-0001.html From krice at freeswitch.org Fri Jan 9 18:05:35 2015 From: krice at freeswitch.org (Ken Rice) Date: Fri, 09 Jan 2015 15:05:35 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <54afee3f4a26b_694f93f328596e7@ip-10-33-129-37.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/233e29cb/attachment.html From david.villasmil at gmail.com Fri Jan 9 18:23:04 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Fri, 9 Jan 2015 16:23:04 +0100 Subject: [Freeswitch-users] Launch a script on errors in freeswitch log file In-Reply-To: References: Message-ID: You could just tail the log file and grab what you need... On Jan 9, 2015 3:56 PM, "Aqs Younas" wrote: > Hi, > > How can i launch a script when there are error logs in freeswitch log > file. Actually i am using mod_shout as MOH with mod_local_stream, when > someone enters a url of stream that is not responding i see logs all these. > > 2015-01-09 19:38:39.338003 [ERR] mod_local_stream.c:214 Can't open shout:// > 174.37.194.139:8316 > 2015-01-09 19:38:41.337988 [ERR] switch_core_file.c:149 Invalid file > format [shout] for [174.37.194.139:8316]! > 2015-01-09 19:38:42.337988 [ERR] mod_local_stream.c:214 Can't open shout:// > s9.voscast.com:7584 > 2015-01-09 19:38:42.337988 [ERR] switch_core_file.c:149 Invalid file > format [shout] for [s9.voscast.com:7584]! > > > I want to run an external script to remove that file or urls when there > are errors logs in freeswitch. > > But i don't know how to trigger that script on errors in freeswitch log. > > Any help would be much appreciated. > > Thanks. > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/8c654312/attachment.html From steveayre at gmail.com Fri Jan 9 22:17:46 2015 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 9 Jan 2015 19:17:46 +0000 Subject: [Freeswitch-users] Launch a script on errors in freeswitch log file In-Reply-To: References: Message-ID: For what purpose? Monitoring can be done with a number of tools such as logwatch. If you're wanting to remove the lines to save space look at using logrotate to archive/remove old logs. Steve On 9 January 2015 at 14:55, Aqs Younas wrote: > Hi, > > How can i launch a script when there are error logs in freeswitch log > file. Actually i am using mod_shout as MOH with mod_local_stream, when > someone enters a url of stream that is not responding i see logs all these. > > 2015-01-09 19:38:39.338003 [ERR] mod_local_stream.c:214 Can't open shout:// > 174.37.194.139:8316 > 2015-01-09 19:38:41.337988 [ERR] switch_core_file.c:149 Invalid file > format [shout] for [174.37.194.139:8316]! > 2015-01-09 19:38:42.337988 [ERR] mod_local_stream.c:214 Can't open shout:// > s9.voscast.com:7584 > 2015-01-09 19:38:42.337988 [ERR] switch_core_file.c:149 Invalid file > format [shout] for [s9.voscast.com:7584]! > > > I want to run an external script to remove that file or urls when there > are errors logs in freeswitch. > > But i don't know how to trigger that script on errors in freeswitch log. > > Any help would be much appreciated. > > Thanks. > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/cbcf6848/attachment.html From brian at freeswitch.org Fri Jan 9 22:22:04 2015 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jan 2015 13:22:04 -0600 Subject: [Freeswitch-users] Launch a script on errors in freeswitch log file In-Reply-To: References: Message-ID: Looks like you don't have mod_shout loaded. On Fri, Jan 9, 2015 at 8:55 AM, Aqs Younas wrote: > Hi, > > How can i launch a script when there are error logs in freeswitch log > file. Actually i am using mod_shout as MOH with mod_local_stream, when > someone enters a url of stream that is not responding i see logs all these. > > 2015-01-09 19:38:39.338003 [ERR] mod_local_stream.c:214 Can't open shout:// > 174.37.194.139:8316 > 2015-01-09 19:38:41.337988 [ERR] switch_core_file.c:149 Invalid file > format [shout] for [174.37.194.139:8316]! > 2015-01-09 19:38:42.337988 [ERR] mod_local_stream.c:214 Can't open shout:// > s9.voscast.com:7584 > 2015-01-09 19:38:42.337988 [ERR] switch_core_file.c:149 Invalid file > format [shout] for [s9.voscast.com:7584]! > > > I want to run an external script to remove that file or urls when there > are errors logs in freeswitch. > > But i don't know how to trigger that script on errors in freeswitch log. > > Any help would be much appreciated. > > Thanks. > > _________________________________________________________________________ > 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/20150109/20348846/attachment.html From aqsyounas at gmail.com Fri Jan 9 23:19:11 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 10 Jan 2015 01:19:11 +0500 Subject: [Freeswitch-users] Launch a script on errors in freeswitch log file In-Reply-To: References: Message-ID: Yup i deliberately have not loaded it. Actually when we add url's which are not mp3(mod_shout supported format) they give us error. As i am using mod_local_stream with it to run it as local stream, any invalid entry in my playlist.loc file will flood my freeswitch with logs. I want to remove that entry by running an external script upon error. Streams url are added by client through some website. And sometime they add invalid stream this gives us error. I manually want to remove that entry. Thanks for your reply. On 10 January 2015 at 00:22, Brian West wrote: > Looks like you don't have mod_shout loaded. > > On Fri, Jan 9, 2015 at 8:55 AM, Aqs Younas wrote: > >> Hi, >> >> How can i launch a script when there are error logs in freeswitch log >> file. Actually i am using mod_shout as MOH with mod_local_stream, when >> someone enters a url of stream that is not responding i see logs all these. >> >> 2015-01-09 19:38:39.338003 [ERR] mod_local_stream.c:214 Can't open >> shout://174.37.194.139:8316 >> 2015-01-09 19:38:41.337988 [ERR] switch_core_file.c:149 Invalid file >> format [shout] for [174.37.194.139:8316]! >> 2015-01-09 19:38:42.337988 [ERR] mod_local_stream.c:214 Can't open >> shout://s9.voscast.com:7584 >> 2015-01-09 19:38:42.337988 [ERR] switch_core_file.c:149 Invalid file >> format [shout] for [s9.voscast.com:7584]! >> >> >> I want to run an external script to remove that file or urls when there >> are errors logs in freeswitch. >> >> But i don't know how to trigger that script on errors in freeswitch log. >> >> Any help would be much appreciated. >> >> Thanks. >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150110/b1848177/attachment-0001.html From blefko5361 at gmail.com Sat Jan 10 00:02:30 2015 From: blefko5361 at gmail.com (Bruce Lefko) Date: Fri, 9 Jan 2015 15:02:30 -0600 Subject: [Freeswitch-users] g711 fallback on calls where t38 negotiation fails Message-ID: Is there a way in mod_spandsp to allow for g711 fallback in calls where t38 negotiation fails? I would like to not have to call a number back with fax_enable_t38=false and have the call continue with audio fallback. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/2c31f10a/attachment.html From adam.ben.ayoun1 at gmail.com Sat Jan 10 02:26:30 2015 From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun) Date: Sat, 10 Jan 2015 01:26:30 +0200 Subject: [Freeswitch-users] SIP over Websocket VS SIP over TCP Message-ID: Hi, We are developing a mobile client that will use the WebRTC media stack and Freeswitch as an MCU (only for conference calls). My question is, since we build a native app, can we use SIP over TCP for signalling? In other words, if Freeswitch receives the WebRTC kind of SDP, will it be able to communicate in the same way as if we were using the SIP over Websocket (the other Freeswitch option)? Any corner cases/considerations with this? Our goal is to avoid implementing SIP over Websocket on the client as much as possible. Thanks, Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150110/caf7dd85/attachment.html From sdame at 207me.com Sat Jan 10 06:32:45 2015 From: sdame at 207me.com (Stephen Dame) Date: Fri, 9 Jan 2015 22:32:45 -0500 Subject: [Freeswitch-users] Auto Changing stun/rtp/dtls port every second? Message-ID: <013e01d02c86$1d8f8850$58ae98f0$@207me.com> Auto Changing stun/rtp/dtls port every second? Just built from master FreeSWITCH Version 1.5.15b+git~20150109T000119Z~d199060249~64bit (git d199060 2015-01-09 00:01:19Z 64bit) And when connecting with chrome web-rtc thru sip.js I noticed switch appears to be changing or flipping ports every second bouncing between them. Happens with Chrome, But not Firefox. 2015-01-10 01:57:22.095932 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 2015-01-10 01:57:22.535937 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42751 to xx.xx.xx.xx:42748 2015-01-10 01:57:23.095959 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 2015-01-10 01:57:23.495890 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42751 to xx.xx.xx.xx:42748 2015-01-10 01:57:23.995945 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 ..continues on every second for duration of call. Audio connection seems to be working fine, but have never seen this in prior builds. Not clear to me if this is a real issue, or just annoyance in fs_cli console. Thanks in advance for any suggestions. I have included the SDPs below for both Chrome and FireFox. Regards, Stephen CHROME REMOTE 2015-01-10 02:40:03.535934 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=- 944480236656755000 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 m=audio 44646 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 xx.xx.xx.xx a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=rtcp:44646 IN IP4 xx.xx.xx.xx a=candidate:331141188 1 udp 2122260223 192.168.99.220 57065 typ host generation 0 a=candidate:331141188 2 udp 2122260223 192.168.99.220 57065 typ host generation 0 a=candidate:693976877 1 udp 2122194687 192.168.52.1 57066 typ host generation 0 a=candidate:693976877 2 udp 2122194687 192.168.52.1 57066 typ host generation 0 a=candidate:2448668656 1 udp 2122129151 192.168.142.1 57067 typ host generation 0 a=candidate:2448668656 2 udp 2122129151 192.168.142.1 57067 typ host generation 0 a=candidate:4031593493 1 udp 2122063615 192.168.99.193 57068 typ host generation 0 a=candidate:4031593493 2 udp 2122063615 192.168.99.193 57068 typ host generation 0 a=candidate:1564421300 1 tcp 1518280447 192.168.99.220 0 typ host tcptype active generation 0 a=candidate:1564421300 2 tcp 1518280447 192.168.99.220 0 typ host tcptype active generation 0 a=candidate:1742652381 1 tcp 1518214911 192.168.52.1 0 typ host tcptype active generation 0 a=candidate:1742652381 2 tcp 1518214911 192.168.52.1 0 typ host tcptype active generation 0 a=candidate:3748678400 1 tcp 1518149375 192.168.142.1 0 typ host tcptype active generation 0 a=candidate:3748678400 2 tcp 1518149375 192.168.142.1 0 typ host tcptype active generation 0 a=candidate:3201220837 1 tcp 1518083839 192.168.99.193 0 typ host tcptype active generation 0 a=candidate:3201220837 2 tcp 1518083839 192.168.99.193 0 typ host tcptype active generation 0 a=candidate:3744591831 1 udp 1686052607 xx.xx.xx.xx 44646 typ srflx raddr 192.168.99.220 rport 57065 generation 0 a=candidate:3744591831 2 udp 1686052607 xx.xx.xx.xx 44646 typ srflx raddr 192.168.99.220 rport 57065 generation 0 a=candidate:1019216774 1 udp 1685855999 xx.xx.xx.xx 44647 typ srflx raddr 192.168.99.193 rport 57068 generation 0 a=candidate:1019216774 2 udp 1685855999 xx.xx.xx.xx 44647 typ srflx raddr 192.168.99.193 rport 57068 generation 0 a=ice-ufrag:vWJYtjRU7/AK7HTV a=ice-pwd:tPowc1rLFI4tlG9OuSJsZmhm a=ice-options:google-ice a=fingerprint:sha-256 60:DB:2E:2E:4C:EC:C8:E4:E8:9A:0A:DD:9E:5B:F5:72:7A:AA:E2:BA:4C:B5:6E:77:1B:0 8:CA:D0:33:69:76:18 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtcp-mux a=maxptime:60 a=ssrc:207143798 cname:RQHJDeTxW6AqEXek a=ssrc:207143798 msid:YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 4c7d7f12-cdc6-4fc7-8fde-ab184c4374c6 a=ssrc:207143798 mslabel:YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 a=ssrc:207143798 label:4c7d7f12-cdc6-4fc7-8fde-ab184c4374c6 2015-01-10 02:40:03.555959 [DEBUG] sofia.c:6890 (sofia/external/ID-llll at sb.domain.com) State Change CS_NEW -> CS_INIT LOCAL CHROME 2015-01-10 02:40:03.555959 [DEBUG] mod_sofia.c:780 Local SDP sofia/external/ID-llll at sb.domain.com: v=0 o=FreeSWITCH 1420828455 1420828456 IN IP4 126.254.27.18 s=FreeSWITCH c=IN IP4 126.254.27.18 t=0 0 a=msid-semantic: WMS nURTbPNjtYX6b7AHGQff8WLdypK5IlTV m=audio 29148 UDP/TLS/RTP/SAVPF 111 126 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:126 telephone-event/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 C5:4F:B0:97:3E:08:F7:4D:28:D4:DC:66:B9:CB:40:D1:43:14:00:7F:7E:F9:0B:FE:3D:A D:AC:11:1C:09:44:76 a=rtcp-mux a=rtcp:29148 IN IP4 126.254.27.18 a=ssrc:1689569427 cname:5AF6gjgz74w81WFy a=ssrc:1689569427 msid:nURTbPNjtYX6b7AHGQff8WLdypK5IlTV a0 a=ssrc:1689569427 mslabel:nURTbPNjtYX6b7AHGQff8WLdypK5IlTV a=ssrc:1689569427 label:nURTbPNjtYX6b7AHGQff8WLdypK5IlTVa0 a=ice-ufrag:DLDCxeGVugR3rAOm a=ice-pwd:1kj7tqxqkTpbGVAaw3CU78Gs a=candidate:4398516811 1 udp 659136 126.254.27.18 29148 typ host generation 0 ---------------------------------------------------------------------------- ---------------------------------------------------------------- REMOTE FIREFOX 2015-01-10 02:54:48.315868 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=Mozilla-SIPUA-34.0.5 24651 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 a=ice-ufrag:0c83e766 a=ice-pwd:3e4fe41a976d4d295b87ef47898f1bb1 a=fingerprint:sha-256 A4:64:42:CB:31:42:04:21:4B:23:4A:44:93:73:2C:64:73:AC:CB:64:DD:0B:00:CA:ED:1 C:35:5A:F8:19:D2:2B m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 c=IN IP4 127.0.0.1 a=rtpmap:109 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=setup:actpass a=rtcp-mux a=candidate:0 1 UDP 2128609535 192.168.99.193 50443 typ host a=candidate:2 1 UDP 2128543999 192.168.99.220 50444 typ host a=candidate:4 1 UDP 2128478463 192.168.142.1 50445 typ host a=candidate:6 1 UDP 2128412927 192.168.52.1 50446 typ host a=candidate:0 2 UDP 2128609534 192.168.99.193 50447 typ host a=candidate:2 2 UDP 2128543998 192.168.99.220 50448 typ host a=candidate:4 2 UDP 2128478462 192.168.142.1 50449 typ host a=candidate:6 2 UDP 2128412926 192.168.52.1 50450 typ host a=candidate:3 1 UDP 1692401663 xx.xx.xx.xx 41316 typ srflx raddr 192.168.99.220 rport 50444 a=candidate:1 1 UDP 1692467199 xx.xx.xx.xx 41315 typ srflx raddr 192.168.99.193 rport 50443 a=candidate:3 2 UDP 1692401662 xx.xx.xx.xx 41318 typ srflx raddr 192.168.99.220 rport 50448 a=candidate:1 2 UDP 1692467198 xx.xx.xx.xx 41317 typ srflx raddr 192.168.99.193 rport 50447 2015-01-10 02:54:48.315868 [DEBUG] sofia.c:6890 (sofia/external/ID-sss at sb.domain.com) State Change CS_NEW -> CS_INIT LOCAL FIREFOX 2015-01-10 02:54:48.335813 [DEBUG] mod_sofia.c:780 Local SDP sofia/external/ID-sss at sb.domain.com: v=0 o=FreeSWITCH 1420830508 1420830509 IN IP4 126.254.27.18 s=FreeSWITCH c=IN IP4 126.254.27.18 t=0 0 a=msid-semantic: WMS 8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX m=audio 27980 UDP/TLS/RTP/SAVPF 109 101 a=rtpmap:109 opus/48000/2 a=fmtp:109 useinbandfec=1; usedtx=1; maxaveragebitrate=30000; maxplaybackrate=48000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 C5:4F:B0:97:3E:08:F7:4D:28:D4:DC:66:B9:CB:40:D1:43:14:00:7F:7E:F9:0B:FE:3D:A D:AC:11:1C:09:44:76 a=rtcp-mux a=rtcp:27980 IN IP4 126.254.27.18 a=ssrc:1689570312 cname:VqmJkBFXjGg5ik9T a=ssrc:1689570312 msid:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX a0 a=ssrc:1689570312 mslabel:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX a=ssrc:1689570312 label:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPXa0 a=ice-ufrag:1otrhmztPY6EkLs6 a=ice-pwd:2FGc6bZfI6aRhaA9oUqw6QPL a=candidate:1819979940 1 udp 659136 126.254.27.18 27980 typ host generation 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150109/e3f38f7a/attachment-0001.html From telishisheer at gmail.com Sat Jan 10 09:55:00 2015 From: telishisheer at gmail.com (Shisheer Teli) Date: Sat, 10 Jan 2015 12:25:00 +0530 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: <24EA0E57C97E4003A10704499F21E931@gmail.com> Message-ID: jira has been made. FS-7143 https://freeswitch.org/jira/browse/FS-7143 Regrads, Shisheer T On Fri, Jan 9, 2015 at 7:30 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > iirc, were asked to make a jira with attached logs. > > > On Friday, January 9, 2015, Shisheer Teli wrote: > >> I did following changes but still video call disconnect after 30 seconds, >> audio call working properly. >> >> Attched log file >> vars.xml >> >> >> >> >> host:domain.example.com is another possible value; however this will not >> toggle the autonat flags. If you are behind NAT, with dynamic DNS (and stun >> doesn't work) you should write a script that determines your public IP >> address, makes the change and calls reloadxml. This also holds true for the >> external profile. No special processing happens to determine the IP address >> before the variable gets passed to the external profile. >> >> host:domain.example.com may be used in places "where you have two >> interfaces in a box and one is public facing and one isn't, so one never >> has to tell the lies." >> >> - source bwk on irc. >> >> internal.xml >> >> >> >> external.xml >> >> >> >> >> >> On Fri, Jan 9, 2015 at 4:54 AM, Seven Du wrote: >> >>> perhaps report a jira with debug level log and >>> >>> sofia global siptrace on >>> >>> >>> -- >>> Seven Du >>> http://about.me/dujinfang >>> http://www.dujinfang.com >>> http://www.freeswitch.org.cn >>> >>> Sent with Sparrow >>> >>> On Thursday, January 8, 2015 at 6:52 PM, Shisheer Teli wrote: >>> >>> i am using FreeSWITCH Version >>> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >>> 03:40:22Z 64bit) >>> >>> and still video call disconnect after 30 seconds >>> >>> Regards, >>> Shisheer T >>> >>> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>> Try latest master or 1.4.15 >>> >>> >>> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >>> wrote: >>> >>> Hi Team, >>> >>> I don't know what happen , but when I start video call it disconnected >>> after every 30 seconds. >>> >>> e.g. >>> x-lite to x-lite call : video call disconnect after 30 seconds >>> >>> X-lite to Zoiper : video call continue, but no video sending. >>> >>> >>> Regards >>> Shisheer T >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >>> >>> >>> -- >>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>> >>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>> http://twitter.com/FreeSWITCH >>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >>> >>> >>> -- >>> Regards, >>> Shisheer Teli >>> Phone: +91-022 2278 2519 / 2121 >>> shisheer at tifr.res.in >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> Regards, >> Shisheer Teli >> Phone: +91-022 2278 2519 / 2121 >> shisheer at tifr.res.in >> > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > > _________________________________________________________________________ > 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 > -- Regards, Shisheer Teli Phone: +91-022 2278 2519 / 2121 shisheer at tifr.res.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150110/dd1968a3/attachment.html From notify.sina at gmail.com Sat Jan 10 12:46:12 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Sat, 10 Jan 2015 10:46:12 +0100 Subject: [Freeswitch-users] Call back through SIP trunk In-Reply-To: References: <0000014aa16b2a0b-ee3e05bc-9993-4da0-8da1-2edff71f346b-000000@email.amazonses.com> <0000014aa59db0b0-5b7762c1-dc93-4407-acf9-a05d6d8543ea-000000@email.amazonses.com> Message-ID: Hi Michael I'm still struggling, so much that I didn't see your response til now! :-) I'm trying to connect the outbound leg through a SIP gateway, using the inbound DID as both origination caller number and name, if I understand you correctly. I hope I understand you correctly! On Tue, Jan 6, 2015 at 5:42 PM, Michael Collins wrote: > Hi Sina, > > Were you able to figure this one out? Also, I didn't see anywhere in this > thread what the outbound leg would be connected to. For example, when you > call out to the caller ID number received on the initial call, what will the > other leg be connected to? Do you have a user or an IVR or what? Just > curious. You can't make an outbound call without connecting that leg to > something. > > -MC > > > On Fri, Jan 2, 2015 at 4:22 PM, Sina Owolabi wrote: >> >> Hi Avi >> >> I'm really sorry to bother you offlist again, but I am really, really >> stumped. >> This is my first time with lua or any kind of coding and I am >> scrambling to make amends for that. >> >> Could you please help me with some kind of example script I can use for >> this? >> >> On Thu, Jan 1, 2015 at 2:09 PM, Avi Marcus wrote: >> > Once you hangup, unless you have zombie exec, the call ends and it won't >> > transfer nor execute the lua script. >> > >> > Also, if you have the lua do the hangup, it can directly access all the >> > channel variables itself. >> > >> > Alternatively, you can set a hangup hook and pass everything: >> > >> > >> > -Avi >> > >> > On Thu, Jan 1, 2015 at 2:12 PM, Sina Owolabi >> > wrote: >> >> >> >> Avi, >> >> >> >> thanks and happy new year! >> >> I'm trying to using a public extension to receive the call, hangup >> >> and transfer to the extension that is actually doing the lua'ing (so >> >> to speak): >> >> >> >> >> >> > >> expression="^0(\d{10})$"require-nested="false"> >> >> > >> data="effective_caller_id_number=+123${1}"/> >> >> > >> data="effective_caller_id_name=+123${1}"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > >> >> >> Please is this the proper way to collect data from the first leg to >> >> pass to the lua script? >> >> >> >> On Wed, Dec 31, 2014 at 6:36 PM, Avi Marcus wrote: >> >> > I've used a lua script to grab the information (and make sure it's a >> >> > valid >> >> > callback), hangup, and then run: >> >> > >> >> > freeswitch.msleep(2000); --wait 2 seconds to make sure their side >> >> > will >> >> > actually have the call over >> >> > api = freeswitch.API() >> >> > api:execute("originate", your-dialstring) >> >> > >> >> > -Avi >> >> > >> >> > On Wed, Dec 31, 2014 at 11:37 AM, Sina Owolabi >> >> > >> >> > wrote: >> >> >> >> >> >> Hi List! >> >> >> >> >> >> >> >> >> FreeSWITCHNewbie here. >> >> >> Please can I have some guidance on how to setup call back? >> >> >> >> >> >> I would like to be able to dial the DID attached to the SIP trunk >> >> >> Freeswitch is registered to, and then have freeSWITCH hang up the >> >> >> call >> >> >> and dial the caller id number back through any other SIP trunk >> >> >> FreeSWITCH Is registered with, but with the origination number set >> >> >> to >> >> >> the DID that the first call came through in the first place. >> >> >> >> >> >> Please is this possible just through the dial plan? >> >> >> >> >> >> Thanks for any help! >> >> >> >> >> >> > > > > _________________________________________________________________________ > 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 From ilya at e-2-m.mobi Sat Jan 10 14:18:01 2015 From: ilya at e-2-m.mobi (ilyakn) Date: Sat, 10 Jan 2015 04:18:01 -0700 (MST) Subject: [Freeswitch-users] 488 Not Acceptable Media during blind call transfer in bypass media mode In-Reply-To: <1420712364644-7596138.post@n2.nabble.com> References: <1420712364644-7596138.post@n2.nabble.com> Message-ID: <1420888681477-7596140.post@n2.nabble.com> As a clarification, I've uploaded two wireshark captures one taken on host running freeswitch and second one from IVR host to our public FTP server: ftp://78.47.185.45 The FreeSwitch version I'm using: 1.5.13b~64bit ( 64bit) Thanks, Ilya -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/488-Not-Acceptable-Media-during-blind-call-transfer-in-bypass-media-mode-tp7596138p7596140.html Sent from the freeswitch-users mailing list archive at Nabble.com. From luis.daniel.lucio at gmail.com Sat Jan 10 18:03:03 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sat, 10 Jan 2015 10:03:03 -0500 Subject: [Freeswitch-users] Call back through SIP trunk In-Reply-To: References: <0000014aa16b2a0b-ee3e05bc-9993-4da0-8da1-2edff71f346b-000000@email.amazonses.com> <0000014aa59db0b0-5b7762c1-dc93-4407-acf9-a05d6d8543ea-000000@email.amazonses.com> Message-ID: read my private email 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 2015-01-10 4:46 GMT-05:00 Sina Owolabi : > Hi Michael > > I'm still struggling, so much that I didn't see your response til now! :-) > I'm trying to connect the outbound leg through a SIP gateway, using > the inbound DID as both origination caller number and name, if I > understand you correctly. > > I hope I understand you correctly! > > > > On Tue, Jan 6, 2015 at 5:42 PM, Michael Collins wrote: >> Hi Sina, >> >> Were you able to figure this one out? Also, I didn't see anywhere in this >> thread what the outbound leg would be connected to. For example, when you >> call out to the caller ID number received on the initial call, what will the >> other leg be connected to? Do you have a user or an IVR or what? Just >> curious. You can't make an outbound call without connecting that leg to >> something. >> >> -MC >> >> >> On Fri, Jan 2, 2015 at 4:22 PM, Sina Owolabi wrote: >>> >>> Hi Avi >>> >>> I'm really sorry to bother you offlist again, but I am really, really >>> stumped. >>> This is my first time with lua or any kind of coding and I am >>> scrambling to make amends for that. >>> >>> Could you please help me with some kind of example script I can use for >>> this? >>> >>> On Thu, Jan 1, 2015 at 2:09 PM, Avi Marcus wrote: >>> > Once you hangup, unless you have zombie exec, the call ends and it won't >>> > transfer nor execute the lua script. >>> > >>> > Also, if you have the lua do the hangup, it can directly access all the >>> > channel variables itself. >>> > >>> > Alternatively, you can set a hangup hook and pass everything: >>> > >>> > >>> > -Avi >>> > >>> > On Thu, Jan 1, 2015 at 2:12 PM, Sina Owolabi >>> > wrote: >>> >> >>> >> Avi, >>> >> >>> >> thanks and happy new year! >>> >> I'm trying to using a public extension to receive the call, hangup >>> >> and transfer to the extension that is actually doing the lua'ing (so >>> >> to speak): >>> >> >>> >> >>> >> >> >> expression="^0(\d{10})$"require-nested="false"> >>> >> >> >> data="effective_caller_id_number=+123${1}"/> >>> >> >> >> data="effective_caller_id_name=+123${1}"/> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >> >> >>> >> Please is this the proper way to collect data from the first leg to >>> >> pass to the lua script? >>> >> >>> >> On Wed, Dec 31, 2014 at 6:36 PM, Avi Marcus wrote: >>> >> > I've used a lua script to grab the information (and make sure it's a >>> >> > valid >>> >> > callback), hangup, and then run: >>> >> > >>> >> > freeswitch.msleep(2000); --wait 2 seconds to make sure their side >>> >> > will >>> >> > actually have the call over >>> >> > api = freeswitch.API() >>> >> > api:execute("originate", your-dialstring) >>> >> > >>> >> > -Avi >>> >> > >>> >> > On Wed, Dec 31, 2014 at 11:37 AM, Sina Owolabi >>> >> > >>> >> > wrote: >>> >> >> >>> >> >> Hi List! >>> >> >> >>> >> >> >>> >> >> FreeSWITCHNewbie here. >>> >> >> Please can I have some guidance on how to setup call back? >>> >> >> >>> >> >> I would like to be able to dial the DID attached to the SIP trunk >>> >> >> Freeswitch is registered to, and then have freeSWITCH hang up the >>> >> >> call >>> >> >> and dial the caller id number back through any other SIP trunk >>> >> >> FreeSWITCH Is registered with, but with the origination number set >>> >> >> to >>> >> >> the DID that the first call came through in the first place. >>> >> >> >>> >> >> Please is this possible just through the dial plan? >>> >> >> >>> >> >> Thanks for any help! >>> >> >> >>> >> >> >> >> >> >> _________________________________________________________________________ >> 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 From anthony.minessale at gmail.com Sat Jan 10 20:49:29 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 10 Jan 2015 11:49:29 -0600 Subject: [Freeswitch-users] SIP over Websocket VS SIP over TCP In-Reply-To: References: Message-ID: The WebRTC media engine is driven completely by the SDP, the transport will not make any difference. On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun wrote: > Hi, > > We are developing a mobile client that will use the WebRTC media stack and > Freeswitch as an MCU (only for conference calls). My question is, since we > build a native app, can we use SIP over TCP for signalling? In other words, > if Freeswitch receives the WebRTC kind of SDP, will it be able to > communicate in the same way as if we were using the SIP over Websocket (the > other Freeswitch option)? Any corner cases/considerations with this? Our > goal is to avoid implementing SIP over Websocket on the client as much as > possible. > > Thanks, > Adam > > _________________________________________________________________________ > 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150110/cc3580c8/attachment.html From adam.ben.ayoun1 at gmail.com Sat Jan 10 21:07:13 2015 From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun) Date: Sat, 10 Jan 2015 20:07:13 +0200 Subject: [Freeswitch-users] SIP over Websocket VS SIP over TCP In-Reply-To: References: Message-ID: Thanks Anthony. I assume that means I can use SIP over TCP/TLS for signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work when communicating with FS (when stuff like fingerprint, etc)? On 10 January 2015 at 19:49, Anthony Minessale wrote: > The WebRTC media engine is driven completely by the SDP, the transport > will not make any difference. > > > On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun > wrote: > >> Hi, >> >> We are developing a mobile client that will use the WebRTC media stack >> and Freeswitch as an MCU (only for conference calls). My question is, since >> we build a native app, can we use SIP over TCP for signalling? In other >> words, if Freeswitch receives the WebRTC kind of SDP, will it be able to >> communicate in the same way as if we were using the SIP over Websocket (the >> other Freeswitch option)? Any corner cases/considerations with this? Our >> goal is to avoid implementing SIP over Websocket on the client as much as >> possible. >> >> Thanks, >> Adam >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150110/360c9874/attachment.html From carlos.ruizdiaz at gmail.com Sat Jan 10 21:35:55 2015 From: carlos.ruizdiaz at gmail.com (=?UTF-8?Q?Carlos_Ruiz_D=C3=ADaz?=) Date: Sat, 10 Jan 2015 12:35:55 -0600 Subject: [Freeswitch-users] SIP over Websocket VS SIP over TCP In-Reply-To: References: Message-ID: WebRTC doesn't specify a signalling protocol. This means that you can use SIP over any transport you want to carry the webRTC enabled SDP. FS will receive the SDP, detect that has a RTP/SAVPF profile and start handling it accordingly. Take for example Jitsi or IMSDroid, they both support webRTC and do SIP over UDP/TCP/TLS. Regards, Carlos On Jan 10, 2015 12:08 PM, "Adam Ben-Ayoun" wrote: > Thanks Anthony. I assume that means I can use SIP over TCP/TLS for > signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work > when communicating with FS (when stuff like fingerprint, etc)? > > On 10 January 2015 at 19:49, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> The WebRTC media engine is driven completely by the SDP, the transport >> will not make any difference. >> >> >> On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun > > wrote: >> >>> Hi, >>> >>> We are developing a mobile client that will use the WebRTC media stack >>> and Freeswitch as an MCU (only for conference calls). My question is, since >>> we build a native app, can we use SIP over TCP for signalling? In other >>> words, if Freeswitch receives the WebRTC kind of SDP, will it be able to >>> communicate in the same way as if we were using the SIP over Websocket (the >>> other Freeswitch option)? Any corner cases/considerations with this? Our >>> goal is to avoid implementing SIP over Websocket on the client as much as >>> possible. >>> >>> Thanks, >>> Adam >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150110/fd4308ad/attachment-0001.html From adam.ben.ayoun1 at gmail.com Sat Jan 10 21:38:34 2015 From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun) Date: Sat, 10 Jan 2015 20:38:34 +0200 Subject: [Freeswitch-users] SIP over Websocket VS SIP over TCP In-Reply-To: References: Message-ID: Great to hear that. Thanks again. On 10 January 2015 at 20:35, Carlos Ruiz D?az wrote: > WebRTC doesn't specify a signalling protocol. This means that you can use > SIP over any transport you want to carry the webRTC enabled SDP. > > FS will receive the SDP, detect that has a RTP/SAVPF profile and start > handling it accordingly. > > Take for example Jitsi or IMSDroid, they both support webRTC and do SIP > over UDP/TCP/TLS. > > Regards, > Carlos > On Jan 10, 2015 12:08 PM, "Adam Ben-Ayoun" > wrote: > >> Thanks Anthony. I assume that means I can use SIP over TCP/TLS for >> signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work >> when communicating with FS (when stuff like fingerprint, etc)? >> >> On 10 January 2015 at 19:49, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> The WebRTC media engine is driven completely by the SDP, the transport >>> will not make any difference. >>> >>> >>> On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun < >>> adam.ben.ayoun1 at gmail.com> wrote: >>> >>>> Hi, >>>> >>>> We are developing a mobile client that will use the WebRTC media stack >>>> and Freeswitch as an MCU (only for conference calls). My question is, since >>>> we build a native app, can we use SIP over TCP for signalling? In other >>>> words, if Freeswitch receives the WebRTC kind of SDP, will it be able to >>>> communicate in the same way as if we were using the SIP over Websocket (the >>>> other Freeswitch option)? Any corner cases/considerations with this? Our >>>> goal is to avoid implementing SIP over Websocket on the client as much as >>>> possible. >>>> >>>> Thanks, >>>> Adam >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>> >>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>> http://twitter.com/FreeSWITCH >>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>> >>> >>> _________________________________________________________________________ >>> 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 >> > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150110/56daedc0/attachment.html From nick.zaitsev at mail.ru Sat Jan 10 23:33:24 2015 From: nick.zaitsev at mail.ru (=?UTF-8?B?TmljayBaYWl0c2V2?=) Date: Sat, 10 Jan 2015 23:33:24 +0300 Subject: [Freeswitch-users] =?utf-8?q?postgresql?= Message-ID: <1420922004.999033200@f404.i.mail.ru> Good day to you, Please,advise me, how can i use postgres schema in freeswitch config (cdr_pg_csv.conf.xml)? ?I'd like to use cdr schema instead public, for example here ? -- Nick Zaitsev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150110/a4e06979/attachment.html From jeremy.ardley at gmail.com Sun Jan 11 04:08:25 2015 From: jeremy.ardley at gmail.com (jeremy ardley) Date: Sun, 11 Jan 2015 09:08:25 +0800 Subject: [Freeswitch-users] Set From User in SIP request Message-ID: <54B1CD09.7090408@gmail.com> Hi, I have a DID provider with multiple incoming numbers. I can send outgoing calls on particular DID numbers by sending a SIP request with the From header formatted as From: dddddddddd ;tag=yyy Where the From name is the DID in full national number form (10 digit) and the From user is the 8 digit account number. Each unique DID has a matching unique account number. I have internal extensions mapped to different DID numbers. I need a mechanism to feed different DID number and account number pairs to the gateway for each outgoing call. I hope to do that via the internal extension xml files e.g. The question is how to pass the new variable outbound_caller_from_user to the gateway and get it substituted in the user field of the SIP From header? I have a secondary issue in that I can set the outbound DID part of the >From Header but it has quotes around it From: "dddddddddd" ;tag=yyy Which is undesirable and it would be helpful to suppress that in Freeswitch itself. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150111/c2a9e0e4/attachment.html From dragic.dusan at gmail.com Sun Jan 11 14:45:09 2015 From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=) Date: Sun, 11 Jan 2015 12:45:09 +0100 Subject: [Freeswitch-users] Set From User in SIP request In-Reply-To: <54B1CD09.7090408@gmail.com> References: <54B1CD09.7090408@gmail.com> Message-ID: Use user_data to get the user variables https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-user_data. Example, where outbound_caller_id_name/number are your did number and account code (you can use whatever var you want): For the second part, I'm not aware of of any standard mechanism in fs to remove the quotes (is your provider really this broken?). Maybe try with sip_invite_full_from channel var. I haven't tried it, but based on the name I'm guessing that you'll have to generate a full From: field including unique tag value (;tag=xxx) manually. On 11 January 2015 at 02:08, jeremy ardley wrote: > Hi, > > I have a DID provider with multiple incoming numbers. I can send outgoing > calls on particular DID numbers by sending a SIP request with the From > header formatted as > > From: dddddddddd ;tag=yyy > > Where the From name is the DID in full national number form (10 digit) and > the From user is the 8 digit account number. Each unique DID has a matching > unique account number. > > I have internal extensions mapped to different DID numbers. > > I need a mechanism to feed different DID number and account number pairs to > the gateway for each outgoing call. I hope to do that via the internal > extension xml files e.g. > > > > > > > > > > > > > > > > > > > > > > The question is how to pass the new variable outbound_caller_from_user to > the gateway and get it substituted in the user field of the SIP From header? > > I have a secondary issue in that I can set the outbound DID part of the From > Header but it has quotes around it > > From: "dddddddddd" ;tag=yyy > > Which is undesirable and it would be helpful to suppress that in Freeswitch > itself. > > Thanks, > > > > _________________________________________________________________________ > 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 -- Du?an Dragi? From olegstolyar at gmail.com Sun Jan 11 20:00:00 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Sun, 11 Jan 2015 09:00:00 -0800 Subject: [Freeswitch-users] Strange noise when using conference_set_auto_outcall Message-ID: Hi guys, When I use conference_set_auto_outcall to dial a PSTN number via a SIP trunk (Level3 is the provider), the existing conference participants start hearing noise after a couple of seconds. It happens whether at that time the line is still ringing or the PSTN party picked up.. The noise persists through the session when the PSTN party is silent. The noise does not happen when I use the simple bridge to dial the PSTN number instead of a conference, so I don't think it's the PSTN provider's fault. On the other hand, when I use the same settings to outcall another FS server, the noise is not there. Is there some conference setting I am missing? Has anyone been able to successfully use set_auto_outcall to call PSTN numbers? I am on FreeSWITCH Version 1.5.14b+git~20140730T171747Z~5075d4af0d~64bit (git 5075d4a 2014-07-30 17:17:47Z 64bit) I am trying to reproduce this on the latest master but having trouble building on my old CentOS 5.9. I'll keep trying. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150111/f294f7aa/attachment.html From GeorgePhelps at gfphelps.com Mon Jan 12 00:07:43 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Sun, 11 Jan 2015 16:07:43 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? Message-ID: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The "transfer" statement, shown below, works (in my inbound dialplan): What is the correct syntax for using "bridge" instead of "transfer"? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: Thanks, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150111/0873d30c/attachment.html From david.villasmil at gmail.com Mon Jan 12 00:19:04 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Sun, 11 Jan 2015 22:19:04 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> Message-ID: https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150111/70248a4c/attachment-0001.html From GeorgePhelps at gfphelps.com Mon Jan 12 03:27:56 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Sun, 11 Jan 2015 19:27:56 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> Message-ID: <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150111/ed403caf/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/20150111/ed403caf/attachment-0001.bin From david.villasmil at gmail.com Mon Jan 12 03:31:03 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Mon, 12 Jan 2015 01:31:03 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> Message-ID: Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/fb9a478b/attachment.html From GeorgePhelps at gfphelps.com Mon Jan 12 04:00:35 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Sun, 11 Jan 2015 20:00:35 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> Message-ID: <120701d02e03$2c900540$85b00fc0$@gfphelps.com> David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150111/72955d88/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/20150111/72955d88/attachment-0001.bin From david.villasmil at gmail.com Mon Jan 12 04:30:40 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Mon, 12 Jan 2015 02:30:40 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <120701d02e03$2c900540$85b00fc0$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> Message-ID: Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/75f8f1a6/attachment-0001.html From GeorgePhelps at gfphelps.com Mon Jan 12 04:47:24 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Sun, 11 Jan 2015 20:47:24 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> Message-ID: <121e01d02e09$b72011e0$256035a0$@gfphelps.com> Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150111/359a0b8e/attachment-0001.html From jeremy.ardley at gmail.com Mon Jan 12 05:19:31 2015 From: jeremy.ardley at gmail.com (jeremy ardley) Date: Mon, 12 Jan 2015 10:19:31 +0800 Subject: [Freeswitch-users] Set From User in SIP request In-Reply-To: References: <54B1CD09.7090408@gmail.com> Message-ID: <54B32F33.1090002@gmail.com> I tried your suggestions and variants on them without much luck. I can set the user name on the B leg no problem butI can't set the user number. However the non-standard header Remote-Party-ID: does contain my requested user number so I know that the requested user number is being passed to the B leg. I also checked the logs and saw the variables effective_caller_id_name effective_caller_id_number origination_caller_id_number Are being set and exported. The relevant stanza from my dial-plan is (I tried with export instead of set but it made no difference) Is there any other parameters or settings I could use to force the default from-user to my variable one? On 11/01/15 19:45, Du?an Dragi? wrote: > Use user_data to get the user variables > https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-user_data. > > Example, where outbound_caller_id_name/number are your did number and > account code (you can use whatever var you want): > data="origination_caller_id_name=${user_data(${user_name}@${domain_name} > var outbound_caller_id_name)}"/> > data="origination_caller_id_number=${user_data(${user_name}@${domain_name} > var outbound_caller_id_number)}"/> > > For the second part, I'm not aware of of any standard mechanism in fs > to remove the quotes (is your provider really this broken?). > Maybe try with sip_invite_full_from channel var. I haven't tried it, > but based on the name I'm guessing that you'll have to generate a full > From: field including unique tag value (;tag=xxx) manually. > > On 11 January 2015 at 02:08, jeremy ardley wrote: >> Hi, >> >> I have a DID provider with multiple incoming numbers. I can send outgoing >> calls on particular DID numbers by sending a SIP request with the From >> header formatted as >> >> From: dddddddddd ;tag=yyy >> >> Where the From name is the DID in full national number form (10 digit) and >> the From user is the 8 digit account number. Each unique DID has a matching >> unique account number. >> >> I have internal extensions mapped to different DID numbers. >> >> I need a mechanism to feed different DID number and account number pairs to >> the gateway for each outgoing call. I hope to do that via the internal >> extension xml files e.g. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> The question is how to pass the new variable outbound_caller_from_user to >> the gateway and get it substituted in the user field of the SIP From header? >> >> I have a secondary issue in that I can set the outbound DID part of the From >> Header but it has quotes around it >> >> From: "dddddddddd" ;tag=yyy >> >> Which is undesirable and it would be helpful to suppress that in Freeswitch >> itself. >> >> Thanks, >> >> >> >> _________________________________________________________________________ >> 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 > > From david.villasmil at gmail.com Mon Jan 12 06:17:30 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Mon, 12 Jan 2015 04:17:30 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <121e01d02e09$b72011e0$256035a0$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> Message-ID: Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/d034fe21/attachment-0001.html From GeorgePhelps at gfphelps.com Mon Jan 12 06:33:02 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Sun, 11 Jan 2015 22:33:02 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> Message-ID: <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150111/59a40dcb/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/20150111/59a40dcb/attachment-0001.bin From myforums.indra at gmail.com Mon Jan 12 08:31:21 2015 From: myforums.indra at gmail.com (indra sena) Date: Mon, 12 Jan 2015 11:01:21 +0530 Subject: [Freeswitch-users] REg IVR and codec negotiation In-Reply-To: References: Message-ID: Hi Steve, Could please provide your comments/suggestions on below quetions marked as Q ? GISR. On Fri, Jan 9, 2015 at 8:35 PM, indra sena wrote: > Hi Steven, > > Thanks for all your deep insights and bearing with interminable questions. > > 1) But I'd first look at why the B-leg isn't being offered more codecs. > Avoiding transcoding would be more of an optimisation, you still need to > look at why the gateway wasn't offered G711. > I have found the reason for why gateway wasn't offered G711, because form > my dialplan is sending absolute_codec_string as 'ep_codec_string', i.e. > "absolute_codec_string=\${ep_codec_string}". It is received by fs as > "absolute_codec_string=G729 at 8000h@20i at 8000b,PCMU at 8000h@20i at 64000b" but > while parsing with switch_separate_string_string() it only assigning to > first one upto first "," (there is a stmt Make sure you have single quotes > ( 'PCMA,PCMU' ) around comma ( ',' ) separated list of codecs to protect it > from parsing list of variables inside of > {var1=val1,var2=val2,absolute_codec_string='GSM,PCMU'}). next cdecs are > considering as separate vars and it is considering like this. > > Q1: So do we need to send all codecs by substituting instead > 'ep_codec_string' like absolute_codec_string='G729,PCMU' from dialplan or > can we send 'ep_codec_string' any other way from dialplan so that it can > parse properly? > > After hardcoding absolute_codec_string='G729,PCMU' in dialplan , B-leg is > getting offer with both G729, G711 and it is responding but due to > transcoder is not there there no voice between these two. > > 2) Before the bridge to that gateway you could try renegotiating the aleg > codec to G711, but it might not work with every client. > https://wiki.freeswitch.org/wiki/Mod_commands#uuid_media_reneg > Q1) As I need to avoid transcoder and IVR should be present. So after > playing IVR can we need to renegotiate with A-leb by uuid_media_reneg , can > we achieve this functionality through dialplan configuration ? > > 3Q) FS-880 jeera > https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-880/, they > mentioned that playing your ringback then using refer "deflect" app to > blind transfer the call to the same box in proxy mode.How this will work. > > Thanks for your suggestions. > > Regards, > GISR. > > > > > On Fri, Jan 9, 2015 at 6:57 PM, Steven Ayre wrote: > >> > Like after receiving b-leg codec as G711 can we switch from G729 codec to >> G711 in A-leg by sending reinvite/update to A-leg with SDP with G711 ? Is >> this procedure works ? currently is freeswitch supports this ? >> >> Before the bridge to that gateway you could try renegotiating the aleg >> codec to G711, but it might not work with every client. >> https://wiki.freeswitch.org/wiki/Mod_commands#uuid_media_reneg >> But I'd first look at why the B-leg isn't being offered more codecs. >> Avoiding transcoding would be more of an optimisation, you still need to >> look at why the gateway wasn't offered G711. >> >> > After IVR , while giving answer to the A-leg why it is putting only one >> codec (with which it has playing IVR), not putting all the allowables >> codecs by A-leg ? >> >> Because at that point it's negotiated which codec to use for the call. >> The a-leg sends the list of codecs it supports, freeswitch compares that to >> its own list and picks the single best match. >> >> >> >> >> >> On 9 January 2015 at 11:22, indra sena wrote: >> >>> Hi Steven, >>> >>> Thanks for your quick response and valuable suggestions. >>> >>> With out transcoding is there any way to achieve this ? >>> >>> Like after receiving b-leg codec as G711 can we switch from G729 codec >>> to G711 in A-leg by sending reinvite/update to A-leg with SDP with G711 ? >>> Is this procedure works ? currently is freeswitch supports this ? >>> >>> I have one below query. >>> After IVR , while giving answer to the A-leg why it is putting only one >>> codec (with which it has playing IVR), not putting all the allowables >>> codecs by A-leg ? >>> >>> Thanks in advance. >>> >>> Thanks, >>> GISR. >>> >>> >>> >>> On Fri, Jan 9, 2015 at 4:32 PM, Steven Ayre wrote: >>> >>>> Default behaviour should be that it offers all codecs to the gateway >>>> and transcodes G729-G711 if required. >>>> >>>> Check outbound-codec-prefs on the profile you're sending to the gateway >>>> from? Check it includes both G729 and G711. >>>> >>>> Check disable-transcoding is not set to true >>>> >>>> If you're using absolute_codec_string check it's including G711, this >>>> would override any other settings >>>> >>>> >>>> >>>> On 9 January 2015 at 10:54, indra sena >>>> wrote: >>>> >>>>> Hi All, >>>>> >>>>> Do any body have any suggestion on this ? >>>>> >>>>> >>>>> I have observed in freeswitch that, In case of IVR scenario prior to >>>>> the bridge FreeSwitch plays IVR with the 1st priority codec then invite to >>>>> the bridge endpoint (B-leg) with SDP containing only the codec negotiated >>>>> for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing >>>>> only with IVR negotiated codec. >>>>> >>>>> I am having one issue here for example. >>>>> >>>>> Originator(A-leg) sends invite with G729, G711 and Freeswitch >>>>> negotiated with G729 and starts playing IVR with G729 and answered(183) to >>>>> originator with SDP containing only G729. And Invited to termination >>>>> endpoint with SDP having only G729 and termination gateway having support >>>>> of only G711, and it is rejecting call after receiving invite with only >>>>> G729. >>>>> >>>>> It should be work like after IVR , invite can be having sdp with all >>>>> the originator supported codecs and if termination (b-leg) having different >>>>> priority codecs then it can send re-Invite sanding the same. >>>>> >>>>> I have observed one issue in freeswitch Jeera ( >>>>> https://freeswitch.org/jira/browse/FS-880) , but there is no solution >>>>> in that page. >>>>> >>>>> Do we have any solution for this right now ? >>>>> >>>>> Your answers will be very much helpful for me. I appreciate if you can >>>>> quick response or give some solution for this. >>>>> >>>>> Thanks & Reagrds, >>>>> GISR.. >>>>> >>>>> On Thu, Jan 8, 2015 at 12:39 PM, indra sena >>>>> wrote: >>>>> >>>>>> Hi , >>>>>> >>>>>> I have observed in freeswitch that, In case of IVR scenario prior to >>>>>> the bridge FreeSwitch plays IVR with the 1st priority codec then invite to >>>>>> the bridge endpoint (B-leg) with SDP containing only the codec negotiated >>>>>> for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing >>>>>> only with IVR negotiated codec. >>>>>> >>>>>> I am having one issue here for example. >>>>>> >>>>>> Originator(A-leg) sends invite with G729, G711 and Freeswitch >>>>>> negotiated with G729 and starts playing IVR with G729 and answered(183) to >>>>>> originator with SDP containing only G729. And Invited to termination >>>>>> endpoint with SDP having only G729 and termination gateway having support >>>>>> of only G711, and it is rejecting call after receiving invite with only >>>>>> G729. >>>>>> >>>>>> It should be work like after IVR , invite can be having sdp with all >>>>>> the originator supported codecs and if termination (b-leg) having different >>>>>> priority codecs then it can send re-Invite sanding the same. >>>>>> >>>>>> I have observed one issue in freeswitch Jeera ( >>>>>> https://freeswitch.org/jira/browse/FS-880) , but there is no >>>>>> solution in that page. >>>>>> >>>>>> Do we have any solution for this right now ? >>>>>> >>>>>> Your answers will be very much helpful for me. I appreciate if you >>>>>> can quick response or give some solution for this. >>>>>> >>>>>> Thanks in advance. >>>>>> >>>>>> Thanks & Regards, >>>>>> GISR.. >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/4e034b8d/attachment.html From shisheer at tifr.res.in Mon Jan 12 10:34:23 2015 From: shisheer at tifr.res.in (Shisheer Teli) Date: Mon, 12 Jan 2015 13:04:23 +0530 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: <24EA0E57C97E4003A10704499F21E931@gmail.com> Message-ID: i still not found any solution for this ... On Sat, Jan 10, 2015 at 12:25 PM, Shisheer Teli wrote: > jira has been made. > > FS-7143 > > https://freeswitch.org/jira/browse/FS-7143 > > Regrads, > Shisheer T > > > > On Fri, Jan 9, 2015 at 7:30 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> iirc, were asked to make a jira with attached logs. >> >> >> On Friday, January 9, 2015, Shisheer Teli wrote: >> >>> I did following changes but still video call disconnect after 30 >>> seconds, audio call working properly. >>> >>> Attched log file >>> vars.xml >>> >>> >>> >>> >>> host:domain.example.com is another possible value; however this will >>> not toggle the autonat flags. If you are behind NAT, with dynamic DNS (and >>> stun doesn't work) you should write a script that determines your public IP >>> address, makes the change and calls reloadxml. This also holds true for the >>> external profile. No special processing happens to determine the IP address >>> before the variable gets passed to the external profile. >>> >>> host:domain.example.com may be used in places "where you have two >>> interfaces in a box and one is public facing and one isn't, so one never >>> has to tell the lies." >>> >>> - source bwk on irc. >>> >>> internal.xml >>> >>> >>> >>> external.xml >>> >>> >>> >>> >>> >>> On Fri, Jan 9, 2015 at 4:54 AM, Seven Du wrote: >>> >>>> perhaps report a jira with debug level log and >>>> >>>> sofia global siptrace on >>>> >>>> >>>> -- >>>> Seven Du >>>> http://about.me/dujinfang >>>> http://www.dujinfang.com >>>> http://www.freeswitch.org.cn >>>> >>>> Sent with Sparrow >>>> >>>> On Thursday, January 8, 2015 at 6:52 PM, Shisheer Teli wrote: >>>> >>>> i am using FreeSWITCH Version >>>> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >>>> 03:40:22Z 64bit) >>>> >>>> and still video call disconnect after 30 seconds >>>> >>>> Regards, >>>> Shisheer T >>>> >>>> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>> Try latest master or 1.4.15 >>>> >>>> >>>> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >>>> wrote: >>>> >>>> Hi Team, >>>> >>>> I don't know what happen , but when I start video call it disconnected >>>> after every 30 seconds. >>>> >>>> e.g. >>>> x-lite to x-lite call : video call disconnect after 30 seconds >>>> >>>> X-lite to Zoiper : video call continue, but no video sending. >>>> >>>> >>>> Regards >>>> Shisheer T >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>> >>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>> http://twitter.com/FreeSWITCH >>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> Shisheer Teli >>>> Phone: +91-022 2278 2519 / 2121 >>>> shisheer at tifr.res.in >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>> >>> >>> >>> -- >>> Regards, >>> Shisheer Teli >>> Phone: +91-022 2278 2519 / 2121 >>> shisheer at tifr.res.in >>> >> >> >> -- >> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> >> ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Regards, > Shisheer Teli > Phone: +91-022 2278 2519 / 2121 > shisheer at tifr.res.in > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/745121fd/attachment-0001.html From david.villasmil at gmail.com Mon Jan 12 11:21:13 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Mon, 12 Jan 2015 09:21:13 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> Message-ID: try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: > Here you go: > > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > > > 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, January 11, 2015 10:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/3d20053f/attachment-0001.html From GeorgePhelps at gfphelps.com Mon Jan 12 14:03:38 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Mon, 12 Jan 2015 06:03:38 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> Message-ID: <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/1e00cb6f/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/20150112/1e00cb6f/attachment-0001.bin From david.villasmil at gmail.com Mon Jan 12 14:11:48 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Mon, 12 Jan 2015 12:11:48 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> Message-ID: Try it with Sofia/internal/1001%yourip, Sofia/internal/1002%yourip Use the actual ip address, not the domain to see what happens... On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: > David Govea, > > > > That syntax, with more than one extension specified, causes the following > Freeswitch warning log message: > > > > [WARNING] switch_ivr_originate.c:2531 Only calling the first element in > the list in this mode. > > > > However, the call ? to only the first extension on the list ? does work. > > > > 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:* Monday, January 12, 2015 3:21 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > try this: > > > > > > > > > > On Jan 12, 2015 4:33 AM, "George F. Phelps" > wrote: > > Here you go: > > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > > > 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, January 11, 2015 10:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/9b1ed4f4/attachment-0001.html From david.villasmil at gmail.com Mon Jan 12 14:13:34 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Mon, 12 Jan 2015 12:13:34 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> Message-ID: You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: > David Govea, > > > > That syntax, with more than one extension specified, causes the following > Freeswitch warning log message: > > > > [WARNING] switch_ivr_originate.c:2531 Only calling the first element in > the list in this mode. > > > > However, the call ? to only the first extension on the list ? does work. > > > > 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:* Monday, January 12, 2015 3:21 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > try this: > > > > > > > > > > On Jan 12, 2015 4:33 AM, "George F. Phelps" > wrote: > > Here you go: > > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > > > 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, January 11, 2015 10:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/f7dbd781/attachment-0001.html From GeorgePhelps at gfphelps.com Mon Jan 12 15:06:12 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Mon, 12 Jan 2015 07:06:12 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> Message-ID: <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/653bd140/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/20150112/653bd140/attachment-0001.bin From david.villasmil at gmail.com Mon Jan 12 15:12:24 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Mon, 12 Jan 2015 13:12:24 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> Message-ID: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: > David Govea, > > > > Still fails; both extensions rang. However, before I can answer either > one, I heard the same verbal error code: ?231?. > > > > How do I track down the meaning of ?231?? > > > > 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:* Monday, January 12, 2015 6:14 AM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > You can also try: > > bridge user/1001:_:user/1002 > > On Jan 12, 2015 12:04 PM, "George F. Phelps" > wrote: > > David Govea, > > > > That syntax, with more than one extension specified, causes the following > Freeswitch warning log message: > > > > [WARNING] switch_ivr_originate.c:2531 Only calling the first element in > the list in this mode. > > > > However, the call ? to only the first extension on the list ? does work. > > > > 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:* Monday, January 12, 2015 3:21 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > try this: > > > > > > > > > > On Jan 12, 2015 4:33 AM, "George F. Phelps" > wrote: > > Here you go: > > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > > > 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, January 11, 2015 10:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/f979b954/attachment-0001.html From david.villasmil at gmail.com Mon Jan 12 15:15:03 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Mon, 12 Jan 2015 13:15:03 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> Message-ID: BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Are you using freeswitch with its default config or did you install > something like fusionpbx? > Can you please post your log now? the log for the last dial string, where > calls go out and then get hung up. > (Are you sure your codecs are correct?) > > On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > >> David Govea, >> >> >> >> Still fails; both extensions rang. However, before I can answer either >> one, I heard the same verbal error code: ?231?. >> >> >> >> How do I track down the meaning of ?231?? >> >> >> >> 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:* Monday, January 12, 2015 6:14 AM >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? >> >> >> >> You can also try: >> >> bridge user/1001:_:user/1002 >> >> On Jan 12, 2015 12:04 PM, "George F. Phelps" >> wrote: >> >> David Govea, >> >> >> >> That syntax, with more than one extension specified, causes the following >> Freeswitch warning log message: >> >> >> >> [WARNING] switch_ivr_originate.c:2531 Only calling the first element in >> the list in this mode. >> >> >> >> However, the call ? to only the first extension on the list ? does work. >> >> >> >> 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:* Monday, January 12, 2015 3:21 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? >> >> >> >> try this: >> >> >> >> >> >> >> >> >> >> On Jan 12, 2015 4:33 AM, "George F. Phelps" >> wrote: >> >> Here you go: >> >> >> >> >> >> >> >> >> >> >> >> > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> >> >> >> >> >> >> >> >> >> >> Symbol ${domain} resolves to the local LAN, IP address. >> >> >> >> 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, January 11, 2015 10:18 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? >> >> >> >> Cab you paste your dialplan? >> Also, never EVER show your ip addresses. >> >> On Jan 12, 2015 2:48 AM, "George F. Phelps" >> wrote: >> >> Yes, I tested with that dialstring. My extension was registered, and >> online. >> >> >> >> The call disconnects with verbal error code ?231?. The associated >> logfile is at: >> >> >> >> http://pastebin.com/BeWhhgSU >> >> >> >> 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, January 11, 2015 8:31 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? >> >> >> >> Did you try the dialstring with /sofia/internal/1001% your_ip ? If >> extension 1001 is registered they should get the call. What happens when >> you do that? >> >> On Jan 12, 2015 2:01 AM, "George F. Phelps" >> wrote: >> >> David Govea, >> >> >> >> I am attempting to implement simultaneous ringing ? where when one of my >> inbound DIDs is called, then two SIP extensions and one outbound DID are >> all rung at the same time. Simultaneous ringing is also referred, in the >> Freeswitch documentation, as ?forked dialing? and ?calling multiple >> destinations.? >> >> >> >> I am trying to get the first extension to work with ?bridge.? >> >> >> >> This Freeswitch example shows bridging (I thought?) to two (2) extensions: >> >> >> >> *Calling multiple destinations >> * >> >> By using commas to separate the addresses, bridge will dial them >> simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate >> multiple destinations to be dialed in a multi-threaded manner (this is >> referred to as "Enterprise Origination") - this gives more flexibility (and >> avoids the "Only calling the first element in the list in this mode" >> warning) >> >> If you need to set different channel variables for each destination, you >> may prefix the destinations with [] and the variables inside the brackets. >> Example: >> >> > data="[origination_caller_id_number=1234]sofia//, >> [origination_caller_id_number=55555]sofia//"/> >> >> >> >> 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, January 11, 2015 7:31 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? >> >> >> >> Sorry, >> >> I thought you wanted to call the user 1001, because you spoke about >> bridge. You can't "bridge" to an extension. Can you please explain in >> detail what you want to do? >> >> On Jan 12, 2015 1:29 AM, "George F. Phelps" >> wrote: >> >> David Govea, >> >> >> >> Thanks for your input. I tried that coding yesterday, and the call >> failed. I wasn?t 100 percent sure I was using the correct coding. When I >> call, I hear spoken error ?231? and then the call hangs up. >> >> >> >> I created a pastebin.com of the failed call log, at: >> >> >> >> http://pastebin.com/BeWhhgSU >> >> >> >> A reminder that this ?transfer? statement works: >> >> >> >> >> >> >> >> 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, January 11, 2015 4:19 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? >> >> >> >> >> https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user >> >> That's: >> >> >> >> Note the % sign..., not @ >> >> On Jan 11, 2015 10:09 PM, "George F. Phelps" >> wrote: >> >> Can someone help me with my question? >> >> >> >> Thanks, >> >> >> >> George >> >> >> >> *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] >> *Sent:* Saturday, January 10, 2015 12:02 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* How to Bridge To Local Extensions? >> >> >> >> The ?transfer? statement, shown below, works (in my inbound dialplan): >> >> >> >> >> >> >> >> What is the correct syntax for using ?bridge? instead of ?transfer?? The >> following statement does not work for me: >> >> >> >> >> >> >> >> My extensions are effectively default values and in the default directory >> location. For example: >> >> >> >> more /usr/local/freeswitch/conf/directory/default/1001.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> >> >> >> >> >> >> My goal is to configure simultaneous ringing for multiple extensions: >> >> >> >> > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> >> >> >> >> 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 >> >> >> _________________________________________________________________________ >> 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 >> >> >> _________________________________________________________________________ >> 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 >> >> >> _________________________________________________________________________ >> 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 >> > > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/7dcbee81/attachment-0001.html From john.nash778 at gmail.com Mon Jan 12 15:10:17 2015 From: john.nash778 at gmail.com (John Nash) Date: Mon, 12 Jan 2015 17:40:17 +0530 Subject: [Freeswitch-users] Freeswitch + serial forking at opensips Message-ID: Hello, I am facing an issue when using opensips to serial fork Invites to some SIP gateways and using freeswitch in the middle as b2bua to send calls. That means if first Invite fails opensips will pick next available gateway and append branch to send this again to freeswitch. But I have noticed when first gateway sends a failed response (and freeswitch log shows call destroyed) and second invite is sent (Same call id, cseq but different branch tag) , freeswitch seem to have old transaction alive and sends 482 Request merged response. I saw in mail archives that it was discussed few years back but there was no conclusion. Is there a reason freeswitch keeps transaction alive (for 4 seconds I think) even if it received "ACK" for failed response? Regards John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/0760d6d2/attachment.html From wowstar21 at freenet.de Mon Jan 12 15:15:54 2015 From: wowstar21 at freenet.de (bee) Date: Mon, 12 Jan 2015 12:15:54 +0000 (UTC) Subject: [Freeswitch-users] Freeswitch only records empty files References: <1420812082840-7596139.post@n2.nabble.com> Message-ID: What do you mean? Stephen Wilde writes: > > > This can happens if you bypass media > > On Fri, Jan 9, 2015 at 3:01 PM, bee wrote:Hey, i got an script which makes a call to an outbound number (so only 1 > channel is open) and then i want to record the input. But the record > function only create empty 44kb files. When i join the channel over a > softphone the recorded files have input. > Any ideas? > -- > View this message in context: http://freeswitch- users.2379917.n2.nabble.com/Freeswitch-only-records-empty-files- tp7596139.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > ___________________________________________________________________ ______ > Professional FreeSWITCH Consulting Services:consulting- YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsoluti ons.com > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.c luecon.com > FreeSWITCH-users mailing listFreeSWITCH-users lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch- users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- usershttp://www.freeswitch.org > > > > > > > ___________________________________________________________________ ______ > Professional FreeSWITCH Consulting Services: > consulting at ... > 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 ... > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users > http://www.freeswitch.org From rbarroetavena at gmail.com Mon Jan 12 16:59:26 2015 From: rbarroetavena at gmail.com (Ricardo Barroetavena) Date: Mon, 12 Jan 2015 10:59:26 -0300 Subject: [Freeswitch-users] Multitenant subscription restriction Message-ID: Hi, In a multitenant environment, is there a way to restrict one tenant UA from subscribing to another tenant events? For example, if I've got bob at domain1 and alice at domain2, is there a way to prevent bob from subscribing to let's say alice message-summary events? Thanks for the hints -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/19c7bcaf/attachment.html From vipkilla at gmail.com Mon Jan 12 17:22:57 2015 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 12 Jan 2015 09:22:57 -0500 Subject: [Freeswitch-users] Multitenant subscription restriction In-Reply-To: References: Message-ID: Hi Ricardo, I believe you can turn SUBSCRIBE authentication on which would require a user/password to complete SUBSCRIBE, but if they have a valid user/pass, they could probably still SUBSCRIBE to other users in other domains. I hope this helps. Thanks, /V On Mon, Jan 12, 2015 at 8:59 AM, Ricardo Barroetavena < rbarroetavena at gmail.com> wrote: > Hi, > In a multitenant environment, is there a way to restrict one tenant UA > from subscribing to another tenant events? > > For example, if I've got bob at domain1 and alice at domain2, is there a way to > prevent bob from subscribing to let's say alice message-summary events? > > Thanks for the hints > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/beb5c57f/attachment.html From gb at cm.nl Mon Jan 12 17:52:56 2015 From: gb at cm.nl (Grant Bagdasarian) Date: Mon, 12 Jan 2015 14:52:56 +0000 Subject: [Freeswitch-users] Freeswitch + serial forking at opensips In-Reply-To: References: Message-ID: <512d0758f50e4163ab4501540858a6f4@CM-EX-V05.cm.local> I?ve fought with the 482 Request merged response for a long time in Freeswitch, and lost. I ended up sending any subsequent INVITEs to another instance of Freeswitch. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John Nash Sent: Monday, January 12, 2015 1:10 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch + serial forking at opensips Hello, I am facing an issue when using opensips to serial fork Invites to some SIP gateways and using freeswitch in the middle as b2bua to send calls. That means if first Invite fails opensips will pick next available gateway and append branch to send this again to freeswitch. But I have noticed when first gateway sends a failed response (and freeswitch log shows call destroyed) and second invite is sent (Same call id, cseq but different branch tag) , freeswitch seem to have old transaction alive and sends 482 Request merged response. I saw in mail archives that it was discussed few years back but there was no conclusion. Is there a reason freeswitch keeps transaction alive (for 4 seconds I think) even if it received "ACK" for failed response? Regards John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/2ee276bd/attachment.html From john.nash778 at gmail.com Mon Jan 12 18:22:25 2015 From: john.nash778 at gmail.com (John Nash) Date: Mon, 12 Jan 2015 20:52:25 +0530 Subject: [Freeswitch-users] Freeswitch + serial forking at opensips In-Reply-To: <512d0758f50e4163ab4501540858a6f4@CM-EX-V05.cm.local> References: <512d0758f50e4163ab4501540858a6f4@CM-EX-V05.cm.local> Message-ID: Looks like a good work around. But if we have only one instance of freeswitch, can different profile be used (Listening on different port)? On Mon, Jan 12, 2015 at 8:22 PM, Grant Bagdasarian wrote: > I?ve fought with the 482 Request merged response for a long time in > Freeswitch, and lost. > > I ended up sending any subsequent INVITEs to another instance of > Freeswitch. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *John Nash > *Sent:* Monday, January 12, 2015 1:10 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch + serial forking at opensips > > > > Hello, > > I am facing an issue when using opensips to serial fork Invites to some > SIP gateways and using freeswitch in the middle as b2bua to send calls. > That means if first Invite fails opensips will pick next available gateway > and append branch to send this again to freeswitch. > > But I have noticed when first gateway sends a failed response (and > freeswitch log shows call destroyed) and second invite is sent (Same call > id, cseq but different branch tag) , freeswitch seem to have old > transaction alive and sends 482 Request merged response. > > I saw in mail archives that it was discussed few years back but there was > no conclusion. > > Is there a reason freeswitch keeps transaction alive (for 4 seconds I > think) even if it received "ACK" for failed response? > > Regards > > John > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/c522c92a/attachment-0001.html From areski at gmail.com Mon Jan 12 19:03:08 2015 From: areski at gmail.com (Areski) Date: Mon, 12 Jan 2015 17:03:08 +0100 Subject: [Freeswitch-users] Doc-Sprint Friday 16 January 2015 Message-ID: Hi Everyone, We are organizing an other Sprint to work on FreeSWITCH Documentation on *Friday 16 January 2015 at 10am CT*. It will be 4 hours long but you can join for less time. The Doc-sprint will focus on migrating the remaining pages from old FS Wiki (https://wiki.freeswitch.org) to Confluence ( https://freeswitch.org/confluence). We will use this IRC channel during the sprint: #freeswitch-docs and we will be tracking our work on a spreadsheet: https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333 to avoid working on the same content. So during the sprint, please change the page "Status" you are working on to "Editing" with your name next to it. Some extra information: - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration - https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation Let us know if you have any question. We hope to get a maximum number of people signed up! So, who is in? -- Kind regards, /Areski -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/b24b3b9d/attachment.html From mike at jerris.com Mon Jan 12 20:41:35 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Jan 2015 12:41:35 -0500 Subject: [Freeswitch-users] Auto Changing stun/rtp/dtls port every second? In-Reply-To: <013e01d02c86$1d8f8850$58ae98f0$@207me.com> References: <013e01d02c86$1d8f8850$58ae98f0$@207me.com> Message-ID: <6D3DC96C-69C2-4B45-91F1-6FCAB35456EC@jerris.com> Issue seems to be with the sdp from the browser containing multiple working paths. Not sure why we are changing so much however. Can you open up a jira on this so we can investigate? > On Jan 9, 2015, at 10:32 PM, Stephen Dame wrote: > > Auto Changing stun/rtp/dtls port every second? > > Just built from master FreeSWITCH Version 1.5.15b+git~20150109T000119Z~d199060249~64bit (git d199060 2015-01-09 00:01:19Z 64bit) > > And when connecting with chrome web-rtc thru sip.js I noticed switch appears to be changing or flipping ports every second bouncing between them. > > Happens with Chrome, But not Firefox. > > 2015-01-10 01:57:22.095932 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 > 2015-01-10 01:57:22.535937 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42751 to xx.xx.xx.xx:42748 > 2015-01-10 01:57:23.095959 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 > 2015-01-10 01:57:23.495890 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42751 to xx.xx.xx.xx:42748 > 2015-01-10 01:57:23.995945 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 > ?.continues on every second for duration of call. > > Audio connection seems to be working fine, but have never seen this in prior builds. > > Not clear to me if this is a real issue, or just annoyance in fs_cli console. > > Thanks in advance for any suggestions. > > I have included the SDPs below for both Chrome and FireFox. > > Regards, > Stephen > > > > CHROME REMOTE > > 2015-01-10 02:40:03.535934 [DEBUG] sofia.c:6624 Remote SDP: > v=0 > o=- 944480236656755000 2 IN IP4 127.0.0.1 > s=- > t=0 0 > a=group:BUNDLE audio > a=msid-semantic: WMS YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 > m=audio 44646 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 > c=IN IP4 xx.xx.xx.xx > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=rtcp:44646 IN IP4 xx.xx.xx.xx > a=candidate:331141188 1 udp 2122260223 192.168.99.220 57065 typ host generation 0 > a=candidate:331141188 2 udp 2122260223 192.168.99.220 57065 typ host generation 0 > a=candidate:693976877 1 udp 2122194687 192.168.52.1 57066 typ host generation 0 > a=candidate:693976877 2 udp 2122194687 192.168.52.1 57066 typ host generation 0 > a=candidate:2448668656 1 udp 2122129151 192.168.142.1 57067 typ host generation 0 > a=candidate:2448668656 2 udp 2122129151 192.168.142.1 57067 typ host generation 0 > a=candidate:4031593493 1 udp 2122063615 192.168.99.193 57068 typ host generation 0 > a=candidate:4031593493 2 udp 2122063615 192.168.99.193 57068 typ host generation 0 > a=candidate:1564421300 1 tcp 1518280447 192.168.99.220 0 typ host tcptype active generation 0 > a=candidate:1564421300 2 tcp 1518280447 192.168.99.220 0 typ host tcptype active generation 0 > a=candidate:1742652381 1 tcp 1518214911 192.168.52.1 0 typ host tcptype active generation 0 > a=candidate:1742652381 2 tcp 1518214911 192.168.52.1 0 typ host tcptype active generation 0 > a=candidate:3748678400 1 tcp 1518149375 192.168.142.1 0 typ host tcptype active generation 0 > a=candidate:3748678400 2 tcp 1518149375 192.168.142.1 0 typ host tcptype active generation 0 > a=candidate:3201220837 1 tcp 1518083839 192.168.99.193 0 typ host tcptype active generation 0 > a=candidate:3201220837 2 tcp 1518083839 192.168.99.193 0 typ host tcptype active generation 0 > a=candidate:3744591831 1 udp 1686052607 xx.xx.xx.xx 44646 typ srflx raddr 192.168.99.220 rport 57065 generation 0 > a=candidate:3744591831 2 udp 1686052607 xx.xx.xx.xx 44646 typ srflx raddr 192.168.99.220 rport 57065 generation 0 > a=candidate:1019216774 1 udp 1685855999 xx.xx.xx.xx 44647 typ srflx raddr 192.168.99.193 rport 57068 generation 0 > a=candidate:1019216774 2 udp 1685855999 xx.xx.xx.xx 44647 typ srflx raddr 192.168.99.193 rport 57068 generation 0 > a=ice-ufrag:vWJYtjRU7/AK7HTV > a=ice-pwd:tPowc1rLFI4tlG9OuSJsZmhm > a=ice-options:google-ice > a=fingerprint:sha-256 60:DB:2E:2E:4C:EC:C8:E4:E8:9A:0A:DD:9E:5B:F5:72:7A:AA:E2:BA:4C:B5:6E:77:1B:08:CA:D0:33:69:76:18 > a=setup:actpass > a=mid:audio > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time > a=rtcp-mux > a=maxptime:60 > a=ssrc:207143798 cname:RQHJDeTxW6AqEXek > a=ssrc:207143798 msid:YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 4c7d7f12-cdc6-4fc7-8fde-ab184c4374c6 > a=ssrc:207143798 mslabel:YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 > a=ssrc:207143798 label:4c7d7f12-cdc6-4fc7-8fde-ab184c4374c6 > > 2015-01-10 02:40:03.555959 [DEBUG] sofia.c:6890 (sofia/external/ID-llll at sb.domain.com ) State Change CS_NEW -> CS_INIT > > LOCAL CHROME > > 2015-01-10 02:40:03.555959 [DEBUG] mod_sofia.c:780 Local SDP sofia/external/ID-llll at sb.domain.com : > v=0 > o=FreeSWITCH 1420828455 1420828456 IN IP4 126.254.27.18 > s=FreeSWITCH > c=IN IP4 126.254.27.18 > t=0 0 > a=msid-semantic: WMS nURTbPNjtYX6b7AHGQff8WLdypK5IlTV > m=audio 29148 UDP/TLS/RTP/SAVPF 111 126 > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10 > a=rtpmap:126 telephone-event/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 C5:4F:B0:97:3E:08:F7:4D:28:D4:DC:66:B9:CB:40:D1:43:14:00:7F:7E:F9:0B:FE:3D:AD:AC:11:1C:09:44:76 > a=rtcp-mux > a=rtcp:29148 IN IP4 126.254.27.18 > a=ssrc:1689569427 cname:5AF6gjgz74w81WFy > a=ssrc:1689569427 msid:nURTbPNjtYX6b7AHGQff8WLdypK5IlTV a0 > a=ssrc:1689569427 mslabel:nURTbPNjtYX6b7AHGQff8WLdypK5IlTV > a=ssrc:1689569427 label:nURTbPNjtYX6b7AHGQff8WLdypK5IlTVa0 > a=ice-ufrag:DLDCxeGVugR3rAOm > a=ice-pwd:1kj7tqxqkTpbGVAaw3CU78Gs > a=candidate:4398516811 1 udp 659136 126.254.27.18 29148 typ host generation 0 > -------------------------------------------------------------------------------------------------------------------------------------------- > > REMOTE FIREFOX > 2015-01-10 02:54:48.315868 [DEBUG] sofia.c:6624 Remote SDP: > v=0 > o=Mozilla-SIPUA-34.0.5 24651 0 IN IP4 0.0.0.0 > s=SIP Call > t=0 0 > a=ice-ufrag:0c83e766 > a=ice-pwd:3e4fe41a976d4d295b87ef47898f1bb1 > a=fingerprint:sha-256 A4:64:42:CB:31:42:04:21:4B:23:4A:44:93:73:2C:64:73:AC:CB:64:DD:0B:00:CA:ED:1C:35:5A:F8:19:D2:2B > m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 > c=IN IP4 127.0.0.1 > a=rtpmap:109 opus/48000/2 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=setup:actpass > a=rtcp-mux > a=candidate:0 1 UDP 2128609535 192.168.99.193 50443 typ host > a=candidate:2 1 UDP 2128543999 192.168.99.220 50444 typ host > a=candidate:4 1 UDP 2128478463 192.168.142.1 50445 typ host > a=candidate:6 1 UDP 2128412927 192.168.52.1 50446 typ host > a=candidate:0 2 UDP 2128609534 192.168.99.193 50447 typ host > a=candidate:2 2 UDP 2128543998 192.168.99.220 50448 typ host > a=candidate:4 2 UDP 2128478462 192.168.142.1 50449 typ host > a=candidate:6 2 UDP 2128412926 192.168.52.1 50450 typ host > a=candidate:3 1 UDP 1692401663 xx.xx.xx.xx 41316 typ srflx raddr 192.168.99.220 rport 50444 > a=candidate:1 1 UDP 1692467199 xx.xx.xx.xx 41315 typ srflx raddr 192.168.99.193 rport 50443 > a=candidate:3 2 UDP 1692401662 xx.xx.xx.xx 41318 typ srflx raddr 192.168.99.220 rport 50448 > a=candidate:1 2 UDP 1692467198 xx.xx.xx.xx 41317 typ srflx raddr 192.168.99.193 rport 50447 > > 2015-01-10 02:54:48.315868 [DEBUG] sofia.c:6890 (sofia/external/ID-sss at sb.domain.com ) State Change CS_NEW -> CS_INIT > > > LOCAL FIREFOX > > 2015-01-10 02:54:48.335813 [DEBUG] mod_sofia.c:780 Local SDP sofia/external/ID-sss at sb.domain.com : > v=0 > o=FreeSWITCH 1420830508 1420830509 IN IP4 126.254.27.18 > s=FreeSWITCH > c=IN IP4 126.254.27.18 > t=0 0 > a=msid-semantic: WMS 8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX > m=audio 27980 UDP/TLS/RTP/SAVPF 109 101 > a=rtpmap:109 opus/48000/2 > a=fmtp:109 useinbandfec=1; usedtx=1; maxaveragebitrate=30000; maxplaybackrate=48000 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > a=sendrecv > a=fingerprint:sha-256 C5:4F:B0:97:3E:08:F7:4D:28:D4:DC:66:B9:CB:40:D1:43:14:00:7F:7E:F9:0B:FE:3D:AD:AC:11:1C:09:44:76 > a=rtcp-mux > a=rtcp:27980 IN IP4 126.254.27.18 > a=ssrc:1689570312 cname:VqmJkBFXjGg5ik9T > a=ssrc:1689570312 msid:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX a0 > a=ssrc:1689570312 mslabel:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX > a=ssrc:1689570312 label:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPXa0 > a=ice-ufrag:1otrhmztPY6EkLs6 > a=ice-pwd:2FGc6bZfI6aRhaA9oUqw6QPL > a=candidate:1819979940 1 udp 659136 126.254.27.18 27980 typ host generation 0 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/1712f71f/attachment-0001.html From mike at jerris.com Mon Jan 12 20:44:31 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Jan 2015 12:44:31 -0500 Subject: [Freeswitch-users] SIP over Websocket VS SIP over TCP In-Reply-To: References: Message-ID: The only caveat to this is, if you are registering to freeswitch and use that registration to send the browser a call, it will not know to send an sdp that will work with webrtc. You will need to tell it to do so using media_webrtc=true var on originate. > On Jan 10, 2015, at 1:38 PM, Adam Ben-Ayoun wrote: > > Great to hear that. Thanks again. > > On 10 January 2015 at 20:35, Carlos Ruiz D?az > wrote: > WebRTC doesn't specify a signalling protocol. This means that you can use SIP over any transport you want to carry the webRTC enabled SDP. > > FS will receive the SDP, detect that has a RTP/SAVPF profile and start handling it accordingly. > > Take for example Jitsi or IMSDroid, they both support webRTC and do SIP over UDP/TCP/TLS. > > Regards, > Carlos > > On Jan 10, 2015 12:08 PM, "Adam Ben-Ayoun" > wrote: > Thanks Anthony. I assume that means I can use SIP over TCP/TLS for signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work when communicating with FS (when stuff like fingerprint, etc)? > > On 10 January 2015 at 19:49, Anthony Minessale > wrote: > The WebRTC media engine is driven completely by the SDP, the transport will not make any difference. > > > On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun > wrote: > Hi, > > We are developing a mobile client that will use the WebRTC media stack and Freeswitch as an MCU (only for conference calls). My question is, since we build a native app, can we use SIP over TCP for signalling? In other words, if Freeswitch receives the WebRTC kind of SDP, will it be able to communicate in the same way as if we were using the SIP over Websocket (the other Freeswitch option)? Any corner cases/considerations with this? Our goal is to avoid implementing SIP over Websocket on the client as much as possible. > > Thanks, > Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/8608d51e/attachment.html From mike at jerris.com Mon Jan 12 20:46:54 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Jan 2015 12:46:54 -0500 Subject: [Freeswitch-users] Strange noise when using conference_set_auto_outcall In-Reply-To: References: Message-ID: I have used outcall like this. I can't think of anything that would create a noise. What does it sound like? > On Jan 11, 2015, at 12:00 PM, Oleg Stolyar wrote: > > Hi guys, > > When I use conference_set_auto_outcall to dial a PSTN number via a SIP trunk (Level3 is the provider), the existing conference participants start hearing noise after a couple of seconds. It happens whether at that time the line is still ringing or the PSTN party picked up.. The noise persists through the session when the PSTN party is silent. > > The noise does not happen when I use the simple bridge to dial the PSTN number instead of a conference, so I don't think it's the PSTN provider's fault. > > On the other hand, when I use the same settings to outcall another FS server, the noise is not there. > > Is there some conference setting I am missing? > > Has anyone been able to successfully use set_auto_outcall to call PSTN numbers? > > I am on > FreeSWITCH Version 1.5.14b+git~20140730T171747Z~5075d4af0d~64bit (git 5075d4a 2014-07-30 17:17:47Z 64bit) > > I am trying to reproduce this on the latest master but having trouble building on my old CentOS 5.9. I'll keep trying. From mike at jerris.com Mon Jan 12 20:50:48 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Jan 2015 12:50:48 -0500 Subject: [Freeswitch-users] Freeswitch + serial forking at opensips In-Reply-To: References: <512d0758f50e4163ab4501540858a6f4@CM-EX-V05.cm.local> Message-ID: I think that will work, yes. > On Jan 12, 2015, at 10:22 AM, John Nash wrote: > > Looks like a good work around. But if we have only one instance of freeswitch, can different profile be used (Listening on different port)? > > On Mon, Jan 12, 2015 at 8:22 PM, Grant Bagdasarian > wrote: > I?ve fought with the 482 Request merged response for a long time in Freeswitch, and lost. > > I ended up sending any subsequent INVITEs to another instance of Freeswitch. > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of John Nash > Sent: Monday, January 12, 2015 1:10 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Freeswitch + serial forking at opensips > > > > Hello, > > I am facing an issue when using opensips to serial fork Invites to some SIP gateways and using freeswitch in the middle as b2bua to send calls. That means if first Invite fails opensips will pick next available gateway and append branch to send this again to freeswitch. > > But I have noticed when first gateway sends a failed response (and freeswitch log shows call destroyed) and second invite is sent (Same call id, cseq but different branch tag) , freeswitch seem to have old transaction alive and sends 482 Request merged response. > > I saw in mail archives that it was discussed few years back but there was no conclusion. > > Is there a reason freeswitch keeps transaction alive (for 4 seconds I think) even if it received "ACK" for failed response? > > Regards > > John > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/757d1614/attachment.html From olegstolyar at gmail.com Mon Jan 12 21:04:10 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Mon, 12 Jan 2015 10:04:10 -0800 Subject: [Freeswitch-users] Strange noise when using conference_set_auto_outcall In-Reply-To: References: Message-ID: Thanks Michael! Sounds just like noise on the line. After a lot more testing, I now believe it is the PSTN provider's issue after all. The noise does exist on some bridge calls and does not exist on some conference outcalls. For some strange reason it just seems to happen much more frequently on the conference calls than on bridges. I'll update the thread when I figure it out. On Mon, Jan 12, 2015 at 9:46 AM, Michael Jerris wrote: > I have used outcall like this. I can't think of anything that would > create a noise. What does it sound like? > > > On Jan 11, 2015, at 12:00 PM, Oleg Stolyar > wrote: > > > > Hi guys, > > > > When I use conference_set_auto_outcall to dial a PSTN number via a SIP > trunk (Level3 is the provider), the existing conference participants start > hearing noise after a couple of seconds. It happens whether at that time > the line is still ringing or the PSTN party picked up.. The noise persists > through the session when the PSTN party is silent. > > > > The noise does not happen when I use the simple bridge to dial the PSTN > number instead of a conference, so I don't think it's the PSTN provider's > fault. > > > > On the other hand, when I use the same settings to outcall another FS > server, the noise is not there. > > > > Is there some conference setting I am missing? > > > > Has anyone been able to successfully use set_auto_outcall to call PSTN > numbers? > > > > I am on > > FreeSWITCH Version 1.5.14b+git~20140730T171747Z~5075d4af0d~64bit (git > 5075d4a 2014-07-30 17:17:47Z 64bit) > > > > I am trying to reproduce this on the latest master but having trouble > building on my old CentOS 5.9. I'll keep trying. > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/ad74172a/attachment-0001.html From sdame at 207me.com Mon Jan 12 21:17:36 2015 From: sdame at 207me.com (Stephen Dame) Date: Mon, 12 Jan 2015 13:17:36 -0500 Subject: [Freeswitch-users] Auto Changing stun/rtp/dtls port every second? In-Reply-To: <6D3DC96C-69C2-4B45-91F1-6FCAB35456EC@jerris.com> References: <013e01d02c86$1d8f8850$58ae98f0$@207me.com> <6D3DC96C-69C2-4B45-91F1-6FCAB35456EC@jerris.com> Message-ID: <009001d02e94$0b8dbc20$22a93460$@207me.com> Michael thks, I opened one yesterday, here it is. FS-7144 System seems to be working fine, i have not looked at the traffic to see if it is actually switching ports every second. Would be glad to trouble shoot more if pointed in the right direction, it is on a development box so can do whatever is needed. Regards, Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, January 12, 2015 12:42 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Auto Changing stun/rtp/dtls port every second? Issue seems to be with the sdp from the browser containing multiple working paths. Not sure why we are changing so much however. Can you open up a jira on this so we can investigate? On Jan 9, 2015, at 10:32 PM, Stephen Dame > wrote: Auto Changing stun/rtp/dtls port every second? Just built from master FreeSWITCH Version 1.5.15b+git~20150109T000119Z~d199060249~64bit (git d199060 2015-01-09 00:01:19Z 64bit) And when connecting with chrome web-rtc thru sip.js I noticed switch appears to be changing or flipping ports every second bouncing between them. Happens with Chrome, But not Firefox. 2015-01-10 01:57:22.095932 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 2015-01-10 01:57:22.535937 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42751 to xx.xx.xx.xx:42748 2015-01-10 01:57:23.095959 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 2015-01-10 01:57:23.495890 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42751 to xx.xx.xx.xx:42748 2015-01-10 01:57:23.995945 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 ?.continues on every second for duration of call. Audio connection seems to be working fine, but have never seen this in prior builds. Not clear to me if this is a real issue, or just annoyance in fs_cli console. Thanks in advance for any suggestions. I have included the SDPs below for both Chrome and FireFox. Regards, Stephen CHROME REMOTE 2015-01-10 02:40:03.535934 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=- 944480236656755000 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 m=audio 44646 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 xx.xx.xx.xx a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=rtcp:44646 IN IP4 xx.xx.xx.xx a=candidate:331141188 1 udp 2122260223 192.168.99.220 57065 typ host generation 0 a=candidate:331141188 2 udp 2122260223 192.168.99.220 57065 typ host generation 0 a=candidate:693976877 1 udp 2122194687 192.168.52.1 57066 typ host generation 0 a=candidate:693976877 2 udp 2122194687 192.168.52.1 57066 typ host generation 0 a=candidate:2448668656 1 udp 2122129151 192.168.142.1 57067 typ host generation 0 a=candidate:2448668656 2 udp 2122129151 192.168.142.1 57067 typ host generation 0 a=candidate:4031593493 1 udp 2122063615 192.168.99.193 57068 typ host generation 0 a=candidate:4031593493 2 udp 2122063615 192.168.99.193 57068 typ host generation 0 a=candidate:1564421300 1 tcp 1518280447 192.168.99.220 0 typ host tcptype active generation 0 a=candidate:1564421300 2 tcp 1518280447 192.168.99.220 0 typ host tcptype active generation 0 a=candidate:1742652381 1 tcp 1518214911 192.168.52.1 0 typ host tcptype active generation 0 a=candidate:1742652381 2 tcp 1518214911 192.168.52.1 0 typ host tcptype active generation 0 a=candidate:3748678400 1 tcp 1518149375 192.168.142.1 0 typ host tcptype active generation 0 a=candidate:3748678400 2 tcp 1518149375 192.168.142.1 0 typ host tcptype active generation 0 a=candidate:3201220837 1 tcp 1518083839 192.168.99.193 0 typ host tcptype active generation 0 a=candidate:3201220837 2 tcp 1518083839 192.168.99.193 0 typ host tcptype active generation 0 a=candidate:3744591831 1 udp 1686052607 xx.xx.xx.xx 44646 typ srflx raddr 192.168.99.220 rport 57065 generation 0 a=candidate:3744591831 2 udp 1686052607 xx.xx.xx.xx 44646 typ srflx raddr 192.168.99.220 rport 57065 generation 0 a=candidate:1019216774 1 udp 1685855999 xx.xx.xx.xx 44647 typ srflx raddr 192.168.99.193 rport 57068 generation 0 a=candidate:1019216774 2 udp 1685855999 xx.xx.xx.xx 44647 typ srflx raddr 192.168.99.193 rport 57068 generation 0 a=ice-ufrag:vWJYtjRU7/AK7HTV a=ice-pwd:tPowc1rLFI4tlG9OuSJsZmhm a=ice-options:google-ice a=fingerprint:sha-256 60:DB:2E:2E:4C:EC:C8:E4:E8:9A:0A:DD:9E:5B:F5:72:7A:AA:E2:BA:4C:B5:6E:77:1B:08:CA:D0:33:69:76:18 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtcp-mux a=maxptime:60 a=ssrc:207143798 cname:RQHJDeTxW6AqEXek a=ssrc:207143798 msid:YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 4c7d7f12-cdc6-4fc7-8fde-ab184c4374c6 a=ssrc:207143798 mslabel:YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 a=ssrc:207143798 label:4c7d7f12-cdc6-4fc7-8fde-ab184c4374c6 2015-01-10 02:40:03.555959 [DEBUG] sofia.c:6890 ( sofia/external/ID-llll at sb.domain.com) State Change CS_NEW -> CS_INIT LOCAL CHROME 2015-01-10 02:40:03.555959 [DEBUG] mod_sofia.c:780 Local SDP sofia/external/ID-llll at sb.domain.com: v=0 o=FreeSWITCH 1420828455 1420828456 IN IP4 126.254.27.18 s=FreeSWITCH c=IN IP4 126.254.27.18 t=0 0 a=msid-semantic: WMS nURTbPNjtYX6b7AHGQff8WLdypK5IlTV m=audio 29148 UDP/TLS/RTP/SAVPF 111 126 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:126 telephone-event/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 C5:4F:B0:97:3E:08:F7:4D:28:D4:DC:66:B9:CB:40:D1:43:14:00:7F:7E:F9:0B:FE:3D:AD:AC:11:1C:09:44:76 a=rtcp-mux a=rtcp:29148 IN IP4 126.254.27.18 a=ssrc:1689569427 cname:5AF6gjgz74w81WFy a=ssrc:1689569427 msid:nURTbPNjtYX6b7AHGQff8WLdypK5IlTV a0 a=ssrc:1689569427 mslabel:nURTbPNjtYX6b7AHGQff8WLdypK5IlTV a=ssrc:1689569427 label:nURTbPNjtYX6b7AHGQff8WLdypK5IlTVa0 a=ice-ufrag:DLDCxeGVugR3rAOm a=ice-pwd:1kj7tqxqkTpbGVAaw3CU78Gs a=candidate:4398516811 1 udp 659136 126.254.27.18 29148 typ host generation 0 -------------------------------------------------------------------------------------------------------------------------------------------- REMOTE FIREFOX 2015-01-10 02:54:48.315868 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=Mozilla-SIPUA-34.0.5 24651 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 a=ice-ufrag:0c83e766 a=ice-pwd:3e4fe41a976d4d295b87ef47898f1bb1 a=fingerprint:sha-256 A4:64:42:CB:31:42:04:21:4B:23:4A:44:93:73:2C:64:73:AC:CB:64:DD:0B:00:CA:ED:1C:35:5A:F8:19:D2:2B m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 c=IN IP4 127.0.0.1 a=rtpmap:109 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=setup:actpass a=rtcp-mux a=candidate:0 1 UDP 2128609535 192.168.99.193 50443 typ host a=candidate:2 1 UDP 2128543999 192.168.99.220 50444 typ host a=candidate:4 1 UDP 2128478463 192.168.142.1 50445 typ host a=candidate:6 1 UDP 2128412927 192.168.52.1 50446 typ host a=candidate:0 2 UDP 2128609534 192.168.99.193 50447 typ host a=candidate:2 2 UDP 2128543998 192.168.99.220 50448 typ host a=candidate:4 2 UDP 2128478462 192.168.142.1 50449 typ host a=candidate:6 2 UDP 2128412926 192.168.52.1 50450 typ host a=candidate:3 1 UDP 1692401663 xx.xx.xx.xx 41316 typ srflx raddr 192.168.99.220 rport 50444 a=candidate:1 1 UDP 1692467199 xx.xx.xx.xx 41315 typ srflx raddr 192.168.99.193 rport 50443 a=candidate:3 2 UDP 1692401662 xx.xx.xx.xx 41318 typ srflx raddr 192.168.99.220 rport 50448 a=candidate:1 2 UDP 1692467198 xx.xx.xx.xx 41317 typ srflx raddr 192.168.99.193 rport 50447 2015-01-10 02:54:48.315868 [DEBUG] sofia.c:6890 ( sofia/external/ID-sss at sb.domain.com) State Change CS_NEW -> CS_INIT LOCAL FIREFOX 2015-01-10 02:54:48.335813 [DEBUG] mod_sofia.c:780 Local SDP sofia/external/ID-sss at sb.domain.com: v=0 o=FreeSWITCH 1420830508 1420830509 IN IP4 126.254.27.18 s=FreeSWITCH c=IN IP4 126.254.27.18 t=0 0 a=msid-semantic: WMS 8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX m=audio 27980 UDP/TLS/RTP/SAVPF 109 101 a=rtpmap:109 opus/48000/2 a=fmtp:109 useinbandfec=1; usedtx=1; maxaveragebitrate=30000; maxplaybackrate=48000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 C5:4F:B0:97:3E:08:F7:4D:28:D4:DC:66:B9:CB:40:D1:43:14:00:7F:7E:F9:0B:FE:3D:AD:AC:11:1C:09:44:76 a=rtcp-mux a=rtcp:27980 IN IP4 126.254.27.18 a=ssrc:1689570312 cname:VqmJkBFXjGg5ik9T a=ssrc:1689570312 msid:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX a0 a=ssrc:1689570312 mslabel:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX a=ssrc:1689570312 label:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPXa0 a=ice-ufrag:1otrhmztPY6EkLs6 a=ice-pwd:2FGc6bZfI6aRhaA9oUqw6QPL a=candidate:1819979940 1 udp 659136 126.254.27.18 27980 typ host generation 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/b43040bf/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 1017 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/b43040bf/attachment-0001.png From alipey at gmail.com Mon Jan 12 21:51:01 2015 From: alipey at gmail.com (Ali Pey) Date: Mon, 12 Jan 2015 13:51:01 -0500 Subject: [Freeswitch-users] Record sound quality not good but pcap is quite good In-Reply-To: References: Message-ID: Hi Anthony, We tried the latest master last week and still have record issues. The pcap recording sounds quite clear but when the message is recorded by freeswitch, the result is garbled. I have examples to provide if it would be helpful. The calls are in Gy11 u law 20ms ptime and this is how we call record: con->execute('record', '${filename} 300 200 4' ); Any suggestions? Best regards, Ali Pey On Fri, Dec 12, 2014 at 4:43 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Try latest master. > > On Fri, Dec 12, 2014 at 8:54 AM, Ali Pey wrote: > >> Hello, >> >> I use record to record part of a call and sometimes I get some static >> noises and distortion; however when I listen to the call on pcap capture, >> it's quite clear and no noise. >> >> I record to a wav file and then use sox to convert it to ulaw. >> >> Any suggestions? Any help would be greatly appreciated. >> >> >> Thanks, >> Ali Pey >> >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? > > http://freeswitch.org/ > > ? > > http://cluecon.com/ > > ? > > http://twitter.com/FreeSWITCH > > ? > > irc.freenode.net > > #freeswitch ? * > http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/dc5e0fb0/attachment.html From anthony.minessale at gmail.com Mon Jan 12 21:58:13 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 12 Jan 2015 12:58:13 -0600 Subject: [Freeswitch-users] Record sound quality not good but pcap is quite good In-Reply-To: References: Message-ID: executing the record app is not the same as recording the sessions. The record app is used to record to a file in the foreground of the channel for voicemail etc. On Mon, Jan 12, 2015 at 12:51 PM, Ali Pey wrote: > Hi Anthony, > > We tried the latest master last week and still have record issues. > > The pcap recording sounds quite clear but when the message is recorded by > freeswitch, the result is garbled. I have examples to provide if it would > be helpful. > > The calls are in Gy11 u law 20ms ptime and this is how we call record: > > con->execute('record', '${filename} 300 200 4' ); > > > Any suggestions? > > > Best regards, > Ali Pey > > > On Fri, Dec 12, 2014 at 4:43 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Try latest master. >> >> On Fri, Dec 12, 2014 at 8:54 AM, Ali Pey wrote: >> >>> Hello, >>> >>> I use record to record part of a call and sometimes I get some static >>> noises and distortion; however when I listen to the call on pcap capture, >>> it's quite clear and no noise. >>> >>> I record to a wav file and then use sox to convert it to ulaw. >>> >>> Any suggestions? Any help would be greatly appreciated. >>> >>> >>> Thanks, >>> Ali Pey >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> >> ? >> >> http://freeswitch.org/ >> >> ? >> >> http://cluecon.com/ >> >> ? >> >> http://twitter.com/FreeSWITCH >> >> ? >> >> irc.freenode.net >> >> #freeswitch ? * >> http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/ef418c18/attachment-0001.html From alipey at gmail.com Mon Jan 12 22:01:09 2015 From: alipey at gmail.com (Ali Pey) Date: Mon, 12 Jan 2015 14:01:09 -0500 Subject: [Freeswitch-users] Record sound quality not good but pcap is quite good In-Reply-To: References: Message-ID: That's exactly what we use it for: voicemails. Would the same fix not apply here? Is this a different bug? On Mon, Jan 12, 2015 at 1:58 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > executing the record app is not the same as recording the sessions. > The record app is used to record to a file in the foreground of the > channel for voicemail etc. > > > > > On Mon, Jan 12, 2015 at 12:51 PM, Ali Pey wrote: > >> Hi Anthony, >> >> We tried the latest master last week and still have record issues. >> >> The pcap recording sounds quite clear but when the message is recorded by >> freeswitch, the result is garbled. I have examples to provide if it would >> be helpful. >> >> The calls are in Gy11 u law 20ms ptime and this is how we call record: >> >> con->execute('record', '${filename} 300 200 4' ); >> >> >> Any suggestions? >> >> >> Best regards, >> Ali Pey >> >> >> On Fri, Dec 12, 2014 at 4:43 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Try latest master. >>> >>> On Fri, Dec 12, 2014 at 8:54 AM, Ali Pey wrote: >>> >>>> Hello, >>>> >>>> I use record to record part of a call and sometimes I get some static >>>> noises and distortion; however when I listen to the call on pcap capture, >>>> it's quite clear and no noise. >>>> >>>> I record to a wav file and then use sox to convert it to ulaw. >>>> >>>> Any suggestions? Any help would be greatly appreciated. >>>> >>>> >>>> Thanks, >>>> Ali Pey >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>> >>> ? >>> >>> >>> http://freeswitch.org/ >>> >>> ? >>> >>> >>> http://cluecon.com/ >>> >>> ? >>> >>> >>> http://twitter.com/FreeSWITCH >>> >>> ? >>> >>> >>> irc.freenode.net >>> >>> #freeswitch ? * >>> >>> http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>> >>> >>> _________________________________________________________________________ >>> 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 >> >> > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? > > http://freeswitch.org/ > > ? > > http://cluecon.com/ > > ? > > http://twitter.com/FreeSWITCH > > ? > > irc.freenode.net > > #freeswitch ? * > http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/b256eabf/attachment-0001.html From brian at freeswitch.org Mon Jan 12 22:05:42 2015 From: brian at freeswitch.org (Brian West) Date: Mon, 12 Jan 2015 13:05:42 -0600 Subject: [Freeswitch-users] files.freeswitch.org Message-ID: We migrated this instance to a new container today, If you see any weirdness in downloads please report them ASAP, I've also re-enabled IPv6 on files dot. Our allocation is accessible from both HE and Cogent. If anyone has Cogent allocations please do what you can to apply pressure so that Cogent will peer with HE solving the issues with the partitioned IPv6 network. HE is willing to peer, its Cogent that is refusing. Thanks, -- *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/20150112/4b7cc88c/attachment.html From anthony.minessale at gmail.com Mon Jan 12 22:07:03 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 12 Jan 2015 13:07:03 -0600 Subject: [Freeswitch-users] Record sound quality not good but pcap is quite good In-Reply-To: References: Message-ID: Yes, its completely different. Such a bug should be very popular but we have no other reports of such an issue. You most likely have some sort of environmental problem. Have you tried listening to the wav files and not the converted ulaw files? Since you are using the wrong tool to report an issue, there are limited options. On Mon, Jan 12, 2015 at 1:01 PM, Ali Pey wrote: > That's exactly what we use it for: voicemails. > > Would the same fix not apply here? > Is this a different bug? > > > On Mon, Jan 12, 2015 at 1:58 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> executing the record app is not the same as recording the sessions. >> The record app is used to record to a file in the foreground of the >> channel for voicemail etc. >> >> >> >> >> On Mon, Jan 12, 2015 at 12:51 PM, Ali Pey wrote: >> >>> Hi Anthony, >>> >>> We tried the latest master last week and still have record issues. >>> >>> The pcap recording sounds quite clear but when the message is recorded >>> by freeswitch, the result is garbled. I have examples to provide if it >>> would be helpful. >>> >>> The calls are in Gy11 u law 20ms ptime and this is how we call record: >>> >>> con->execute('record', '${filename} 300 200 4' ); >>> >>> >>> Any suggestions? >>> >>> >>> Best regards, >>> Ali Pey >>> >>> >>> On Fri, Dec 12, 2014 at 4:43 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> Try latest master. >>>> >>>> On Fri, Dec 12, 2014 at 8:54 AM, Ali Pey wrote: >>>> >>>>> Hello, >>>>> >>>>> I use record to record part of a call and sometimes I get some static >>>>> noises and distortion; however when I listen to the call on pcap capture, >>>>> it's quite clear and no noise. >>>>> >>>>> I record to a wav file and then use sox to convert it to ulaw. >>>>> >>>>> Any suggestions? Any help would be greatly appreciated. >>>>> >>>>> >>>>> Thanks, >>>>> Ali Pey >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>> >>>> ? >>>> >>>> >>>> http://freeswitch.org/ >>>> >>>> ? >>>> >>>> >>>> http://cluecon.com/ >>>> >>>> ? >>>> >>>> >>>> http://twitter.com/FreeSWITCH >>>> >>>> ? >>>> >>>> >>>> irc.freenode.net >>>> >>>> #freeswitch ? * >>>> >>>> http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >>> >> >> >> >> -- >> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> >> ? >> >> http://freeswitch.org/ >> >> ? >> >> http://cluecon.com/ >> >> ? >> >> http://twitter.com/FreeSWITCH >> >> ? >> >> irc.freenode.net >> >> #freeswitch ? * >> http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ? sip:888 at conference.freeswitch.org ? +19193869900 >> >> >> _________________________________________________________________________ >> 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/9969724d/attachment-0001.html From aqsyounas at gmail.com Mon Jan 12 22:34:11 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 13 Jan 2015 00:34:11 +0500 Subject: [Freeswitch-users] How can i hangup call if there is no audio or dead air on channel. Message-ID: Hi, list I am playing stream and sometimes streams went down and i hear on audio on my soft phone. i want to hangup call on whenever this happens. But don't know how can i do so. Any help would be much appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/1f18bcb1/attachment.html From blefko5361 at gmail.com Mon Jan 12 23:06:16 2015 From: blefko5361 at gmail.com (Bruce Lefko) Date: Mon, 12 Jan 2015 14:06:16 -0600 Subject: [Freeswitch-users] g711 fallback on calls where t38 negotiation fails In-Reply-To: References: Message-ID: Does anyone know if this is possible on freeswitch? Thanks! On Fri, Jan 9, 2015 at 3:02 PM, Bruce Lefko wrote: > Is there a way in mod_spandsp to allow for g711 fallback in calls where > t38 negotiation fails? I would like to not have to call a number back with > fax_enable_t38=false and have the call continue with audio fallback. > > Thanks! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/8a06b668/attachment.html From john.nash778 at gmail.com Mon Jan 12 23:27:22 2015 From: john.nash778 at gmail.com (John Nash) Date: Tue, 13 Jan 2015 01:57:22 +0530 Subject: [Freeswitch-users] Database query from dialplan Message-ID: I need to run a DB query (Postgresql) from dialplan and store results in variables. I came across this link https://wiki.freeswitch.org/wiki/Mod_odbc_query and looks like this is what I need. But I was wondering if I can use direct (without odbc) query to postgresql. Is there any such application available? P.S. I read some posts about postgresql native support but have not been able to figure out what do I need to do to run a postgresql query from dialplan and store results with out odbc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/2caaf51b/attachment.html From italorossib at gmail.com Tue Jan 13 01:00:59 2015 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Mon, 12 Jan 2015 19:00:59 -0300 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: Message-ID: Hi John, Give Lua a try: https://freeswitch.org/confluence/display/FREESWITCH/Lua+with+Database On Mon, Jan 12, 2015 at 5:27 PM, John Nash wrote: > I need to run a DB query (Postgresql) from dialplan and store results in > variables. I came across this link > https://wiki.freeswitch.org/wiki/Mod_odbc_query and looks like this is > what I need. > > But I was wondering if I can use direct (without odbc) query to > postgresql. Is there any such application available? > > P.S. I read some posts about postgresql native support but have not been > able to figure out what do I need to do to run a postgresql query from > dialplan and store results with out odbc > > _________________________________________________________________________ > 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 > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/945e035c/attachment.html From Jeffrey.McKnight at brevardcounty.us Tue Jan 13 01:07:56 2015 From: Jeffrey.McKnight at brevardcounty.us (McKnight, Jeffrey) Date: Mon, 12 Jan 2015 22:07:56 +0000 Subject: [Freeswitch-users] FreeSWITCH employment opportunity in Brevard County, FL Message-ID: <377CF6749E4F0049902AD4EAA2B1CBA33FED8548@EOCWEXCH10V01.brevardco.int> Brevard County, (in Florida) with about 3000 employees, has been evaluating FreeSWITCH as a replacement for county wide legacy telephones for almost a year now. The system is almost completely feature functional in comparison to our Nortel system, however our FreeSWITCH expert has found more profitable employment elsewhere. We are testing the waters to see if there is anyone in the local area that would be interested in completing the perfection of a couple of features, assisting in the rollout of this system county-wide, and maintaining this system. This would be a full time employment position, good benefits, and great longevity. Please email me directly at Jeffrey.mcknight at brevardcounty.us if there is any interest. Regards, Jeff McKnight ----------------------------------------- Under Florida Law, email addresses are Public Records. If you do not want your e-mail address released in response to public record requests, do not send electronic mail to this entity. Instead, contact this office by phone or in writing. From kanem1 at gmail.com Tue Jan 13 01:50:56 2015 From: kanem1 at gmail.com (Mike Kane) Date: Mon, 12 Jan 2015 17:50:56 -0500 Subject: [Freeswitch-users] FreeSWITCH employment opportunity in Brevard County, FL In-Reply-To: <377CF6749E4F0049902AD4EAA2B1CBA33FED8548@EOCWEXCH10V01.brevardco.int> References: <377CF6749E4F0049902AD4EAA2B1CBA33FED8548@EOCWEXCH10V01.brevardco.int> Message-ID: Hello Jeffery. I would be interested in discussing your opportunity. Please contact me at you convenience to discuss. 321-215-5845 Thanks Mike On Jan 12, 2015 5:08 PM, "McKnight, Jeffrey" < Jeffrey.McKnight at brevardcounty.us> wrote: > > Brevard County, (in Florida) with about 3000 employees, has been > evaluating FreeSWITCH as a replacement for county wide legacy telephones > for almost a year now. > > The system is almost completely feature functional in comparison to our > Nortel system, however our FreeSWITCH expert has found more profitable > employment elsewhere. > > We are testing the waters to see if there is anyone in the local area that > would be interested in completing the perfection of a couple of features, > assisting in the rollout of this system county-wide, and maintaining this > system. > > This would be a full time employment position, good benefits, and great > longevity. > > Please email me directly at Jeffrey.mcknight at brevardcounty.us if there is > any interest. > > Regards, > > Jeff McKnight > > ----------------------------------------- > > Under Florida Law, email addresses are Public Records. If you do not want > your e-mail address released in response to public record requests, do not > send electronic mail to this entity. Instead, contact this office by phone > or in writing. > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/2c95cfd5/attachment.html From fs at voice2net.ca Tue Jan 13 01:59:29 2015 From: fs at voice2net.ca (Darcy Primrose) Date: Mon, 12 Jan 2015 17:59:29 -0500 Subject: [Freeswitch-users] fax relay question References: Message-ID: <755031BF67624993BD39A76A79793F69@DARCY> We use a freeswitch as a relay between our carriers and other freeswitches, it has worked great until we upgraded to the latest freeswitch version recently We we receive a fax, inbound and relay it, the subsequent re invite we get back looks like the frist block below, then when the freeswitch relays it, it adds in " Media Description, name and address (m): audio 0 RTP/AVP 19". The originating carrier sees this and responds with audio instead of T38, the call drops. Any hints would be greatly appreaciated, I do not really want to go back to a previous release if I can avoid it. Connection Information (c): IN IP4 98.158.137.200 Time Description, active time (t): 0 0 Media Description, name and address (m): image 18392 udptl t38 Media Attribute (a): T38FaxVersion:0 Media Attribute (a): T38MaxBitRate:14400 Media Attribute (a): T38FaxFillBitRemoval Media Attribute (a): T38FaxRateManagement:transferredTCF Media Attribute (a): T38FaxMaxBuffer:2000 Media Attribute (a): T38FaxMaxDatagram:400 Media Attribute (a): T38FaxUdpEC:t38UDPRedundancy Connection Information (c): IN IP4 98.158.137.200 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 0 RTP/AVP 19 Media Description, name and address (m): image 18392 udptl t38 Media Attribute (a): T38FaxVersion:0 Media Attribute (a): T38MaxBitRate:14400 Media Attribute (a): T38FaxFillBitRemoval Media Attribute (a): T38FaxRateManagement:transferredTCF Media Attribute (a): T38FaxMaxBuffer:2000 Media Attribute (a): T38FaxMaxDatagram:400 Media Attribute (a): T38FaxUdpEC:t38UDPRedundancy Darcy Voice2Net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/e3e80ff2/attachment-0001.html From GeorgePhelps at gfphelps.com Tue Jan 13 02:11:03 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Mon, 12 Jan 2015 18:11:03 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> Message-ID: <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/7e6d47a2/attachment-0001.html From david.villasmil at gmail.com Tue Jan 13 02:15:31 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 13 Jan 2015 00:15:31 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> Message-ID: Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps < GeorgePhelps at gfphelps.com> wrote: > David Govea, > > > > I uploaded a new Freeswitch debug logfile at: > > > > *http://pastebin.com/v17SyXhh * > > > > *Notes* > > > > Only extension 1001 was registered for this test. > > > > Dialstring segment: data="{ignore_early_media=true}user/1000:_:user/1001"/> > > > > I?m guessing that ?*verbal error code 231*? is being generated by my VoIP > service provider. > > > > I am running Freeswitch with (mostly) the default configuration. Changed > passphrases, added my gateway, etc. > > > > I downloaded the source code from git and built it unmodified, from > scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit > (git 1965b3b 2014-12-30 15:06:32Z 64bit)? > > > > My effective codec is G711U ? fully supported throughout the call chain. > > > > 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:* Monday, January 12, 2015 7:15 AM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > BTW, I've never heard of verbal error code 231, that's why I ask whether > you downloaded and freeswitch from the git... > > > > On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Are you using freeswitch with its default config or did you install > something like fusionpbx? > > Can you please post your log now? the log for the last dial string, where > calls go out and then get hung up. > > (Are you sure your codecs are correct?) > > > > On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > David Govea, > > > > Still fails; both extensions rang. However, before I can answer either > one, I heard the same verbal error code: ?231?. > > > > How do I track down the meaning of ?231?? > > > > 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:* Monday, January 12, 2015 6:14 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > You can also try: > > bridge user/1001:_:user/1002 > > On Jan 12, 2015 12:04 PM, "George F. Phelps" > wrote: > > David Govea, > > > > That syntax, with more than one extension specified, causes the following > Freeswitch warning log message: > > > > [WARNING] switch_ivr_originate.c:2531 Only calling the first element in > the list in this mode. > > > > However, the call ? to only the first extension on the list ? does work. > > > > 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:* Monday, January 12, 2015 3:21 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > try this: > > > > > > > > > > On Jan 12, 2015 4:33 AM, "George F. Phelps" > wrote: > > Here you go: > > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > > > 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, January 11, 2015 10:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > _________________________________________________________________________ > 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 > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/c00ed375/attachment-0001.html From chun at reachme.com Tue Jan 13 03:10:56 2015 From: chun at reachme.com (chun) Date: Mon, 12 Jan 2015 16:10:56 -0800 Subject: [Freeswitch-users] Error: cannot find profile Message-ID: Hello, We have a freeswitch installed behind opensip which handles registration. We'd like to turn on and off MWI and cisco phones. When we use the following lua code, it will fail. We also tried using fs_path but it did not work. We appreciate any help. freeswitch.console_log("info", "Lua in da house!!!\n"); local event = freeswitch.Event("message_waiting"); event:addHeader("MWI-Messages-Waiting", "no"); event:addHeader("MWI-Message-Account", "sip:sofia/users/801.24118 at 10.0.151.110"); event:fire(); freeswitch at internal> luarun /etc/freeswitch/scripts/lua/mwi_event.lua +OK 2015-01-12 16:03:19.991140 [INFO] switch_cpp.cpp:1275 Lua in da house!!! 2015-01-12 16:03:20.011145 [ERR] sofia_presence.c:542 Cannot find profile 10.0.151.110 Thank you, Chun-Ju From david.villasmil at gmail.com Tue Jan 13 03:28:53 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 13 Jan 2015 01:28:53 +0100 Subject: [Freeswitch-users] Error: cannot find profile In-Reply-To: References: Message-ID: Try "internal" instead of "users"? On Jan 13, 2015 1:22 AM, "chun" wrote: > Hello, > > We have a freeswitch installed behind opensip which handles registration. > We'd like to turn on and off MWI and cisco phones. When we use the > following > lua code, it will fail. We also tried using fs_path but it did not work. We > appreciate any help. > > freeswitch.console_log("info", "Lua in da house!!!\n"); > local event = freeswitch.Event("message_waiting"); > event:addHeader("MWI-Messages-Waiting", "no"); > event:addHeader("MWI-Message-Account", > "sip:sofia/users/801.24118 at 10.0.151.110"); > event:fire(); > > freeswitch at internal> luarun /etc/freeswitch/scripts/lua/mwi_event.lua > +OK > 2015-01-12 16:03:19.991140 [INFO] switch_cpp.cpp:1275 Lua in da house!!! > 2015-01-12 16:03:20.011145 [ERR] sofia_presence.c:542 Cannot find profile > 10.0.151.110 > > Thank you, > Chun-Ju > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/6edf355c/attachment.html From fs at voice2net.ca Tue Jan 13 04:53:08 2015 From: fs at voice2net.ca (Darcy Primrose) Date: Mon, 12 Jan 2015 20:53:08 -0500 Subject: [Freeswitch-users] fax relay question References: <755031BF67624993BD39A76A79793F69@DARCY> Message-ID: <407BCC22BAAE4CABACDFD19A510D6A62@DARCY> This can be closed, I did a little more research and found this ws aknown problem in 1.4.13, I backed up to 1.4.12 and it is ok now. I shall track this before moving ahead again. It only affects certain devices and carriers, we just happen to have on of the carriers that did not deal with correctly. Incidentally the use yate as their tandem. Darcy ----- Original Message ----- From: Darcy Primrose To: FreeSWITCH Users Help Sent: Monday, January 12, 2015 5:59 PM Subject: [Freeswitch-users] fax relay question We use a freeswitch as a relay between our carriers and other freeswitches, it has worked great until we upgraded to the latest freeswitch version recently We we receive a fax, inbound and relay it, the subsequent re invite we get back looks like the frist block below, then when the freeswitch relays it, it adds in " Media Description, name and address (m): audio 0 RTP/AVP 19". The originating carrier sees this and responds with audio instead of T38, the call drops. Any hints would be greatly appreaciated, I do not really want to go back to a previous release if I can avoid it. Connection Information (c): IN IP4 98.158.137.200 Time Description, active time (t): 0 0 Media Description, name and address (m): image 18392 udptl t38 Media Attribute (a): T38FaxVersion:0 Media Attribute (a): T38MaxBitRate:14400 Media Attribute (a): T38FaxFillBitRemoval Media Attribute (a): T38FaxRateManagement:transferredTCF Media Attribute (a): T38FaxMaxBuffer:2000 Media Attribute (a): T38FaxMaxDatagram:400 Media Attribute (a): T38FaxUdpEC:t38UDPRedundancy Connection Information (c): IN IP4 98.158.137.200 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 0 RTP/AVP 19 Media Description, name and address (m): image 18392 udptl t38 Media Attribute (a): T38FaxVersion:0 Media Attribute (a): T38MaxBitRate:14400 Media Attribute (a): T38FaxFillBitRemoval Media Attribute (a): T38FaxRateManagement:transferredTCF Media Attribute (a): T38FaxMaxBuffer:2000 Media Attribute (a): T38FaxMaxDatagram:400 Media Attribute (a): T38FaxUdpEC:t38UDPRedundancy Darcy Voice2Net ------------------------------------------------------------------------------ _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/4728db6e/attachment.html From john.nash778 at gmail.com Tue Jan 13 09:59:21 2015 From: john.nash778 at gmail.com (John Nash) Date: Tue, 13 Jan 2015 12:29:21 +0530 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: Message-ID: OK. LuaSQL provides direct access to postgresql but does not support connection pooling. ""FreeSWITCH Database Handler" looks like a way to go but I have to go through using ODBC to postgresql right? Can there be any downside of using ODBC? On Tue, Jan 13, 2015 at 3:30 AM, ?talo Rossi wrote: > Hi John, > > Give Lua a try: > > https://freeswitch.org/confluence/display/FREESWITCH/Lua+with+Database > > On Mon, Jan 12, 2015 at 5:27 PM, John Nash wrote: > >> I need to run a DB query (Postgresql) from dialplan and store results in >> variables. I came across this link >> https://wiki.freeswitch.org/wiki/Mod_odbc_query and looks like this is >> what I need. >> >> But I was wondering if I can use direct (without odbc) query to >> postgresql. Is there any such application available? >> >> P.S. I read some posts about postgresql native support but have not been >> able to figure out what do I need to do to run a postgresql query from >> dialplan and store results with out odbc >> >> _________________________________________________________________________ >> 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 >> > > > > -- > ?talo Rossi > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/54c52999/attachment-0001.html From bordmi at rarus.ru Tue Jan 13 10:11:02 2015 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Tue, 13 Jan 2015 11:11:02 +0400 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: Message-ID: Hi! https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core helps you 2015-01-13 9:59 GMT+03:00 John Nash : > OK. LuaSQL provides direct access to postgresql but does not support > connection pooling. ""FreeSWITCH Database Handler" looks like a way to go > but I have to go through using ODBC to postgresql right? Can there be any > downside of using ODBC? > > On Tue, Jan 13, 2015 at 3:30 AM, ?talo Rossi > wrote: > >> Hi John, >> >> Give Lua a try: >> >> https://freeswitch.org/confluence/display/FREESWITCH/Lua+with+Database >> >> On Mon, Jan 12, 2015 at 5:27 PM, John Nash >> wrote: >> >>> I need to run a DB query (Postgresql) from dialplan and store results >>> in variables. I came across this link >>> https://wiki.freeswitch.org/wiki/Mod_odbc_query and looks like this is >>> what I need. >>> >>> But I was wondering if I can use direct (without odbc) query to >>> postgresql. Is there any such application available? >>> >>> P.S. I read some posts about postgresql native support but have not been >>> able to figure out what do I need to do to run a postgresql query from >>> dialplan and store results with out odbc >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> ?talo Rossi >> >> _________________________________________________________________________ >> 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 > -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/a5774402/attachment.html From gkuri at ieee.org Tue Jan 13 10:13:43 2015 From: gkuri at ieee.org (Gabriel Kuri) Date: Mon, 12 Jan 2015 23:13:43 -0800 Subject: [Freeswitch-users] Cisco SPA + BLF behavior Message-ID: I noticed that on a Cisco SPA500 series phone, when configured to do BLF+SD+CP for an extension, the LED for that line on the SPA does not turn red when the line goes off-hook on another phone. It turns RED when the call is actually in progress (ie after the number has been dialed and it's ringing). This behavior is different than when the same line is configured as a shared line appearance on the SPA, the LED turns red immedialtely when the line goes off hook on another phone. It appears FreeSWITCH is sending the same SIP NOTIFY event in both cases with line showing seized (appearance-index=1;appearance-state=seized), so I'm think this might be a bug in the phone's firmware, which is running the latest version, 7.5.6a. I did try a couple older version and had the same results. Am I missing something or is this the behavior of this phone? Should I bother to open a TAC case with Cisco, the behavior seems buggy to me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/c8f77ab8/attachment.html From david.villasmil at gmail.com Tue Jan 13 10:27:20 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 13 Jan 2015 08:27:20 +0100 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: Message-ID: You should look into XML curl. On Jan 13, 2015 8:00 AM, "John Nash" wrote: > OK. LuaSQL provides direct access to postgresql but does not support > connection pooling. ""FreeSWITCH Database Handler" looks like a way to go > but I have to go through using ODBC to postgresql right? Can there be any > downside of using ODBC? > > On Tue, Jan 13, 2015 at 3:30 AM, ?talo Rossi > wrote: > >> Hi John, >> >> Give Lua a try: >> >> https://freeswitch.org/confluence/display/FREESWITCH/Lua+with+Database >> >> On Mon, Jan 12, 2015 at 5:27 PM, John Nash >> wrote: >> >>> I need to run a DB query (Postgresql) from dialplan and store results >>> in variables. I came across this link >>> https://wiki.freeswitch.org/wiki/Mod_odbc_query and looks like this is >>> what I need. >>> >>> But I was wondering if I can use direct (without odbc) query to >>> postgresql. Is there any such application available? >>> >>> P.S. I read some posts about postgresql native support but have not been >>> able to figure out what do I need to do to run a postgresql query from >>> dialplan and store results with out odbc >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> ?talo Rossi >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/5c22207e/attachment.html From regis.freeswitch.org at tornad.net Tue Jan 13 11:12:52 2015 From: regis.freeswitch.org at tornad.net (Regis M) Date: Tue, 13 Jan 2015 09:12:52 +0100 Subject: [Freeswitch-users] FreeSWITCH employment opportunity in Brevard County, FL In-Reply-To: <377CF6749E4F0049902AD4EAA2B1CBA33FED8548@EOCWEXCH10V01.brevardco.int> References: <377CF6749E4F0049902AD4EAA2B1CBA33FED8548@EOCWEXCH10V01.brevardco.int> Message-ID: Hi, I'm not interrested for full time, but if you look for a night support team, we are a french team of Freeswitch's expert which could do N2 support and system monitoring outside your normal office hour. Feel free to contact me offlist if you're interessted. Regards, 2015-01-12 23:07 GMT+01:00 McKnight, Jeffrey < Jeffrey.McKnight at brevardcounty.us>: > > Brevard County, (in Florida) with about 3000 employees, has been > evaluating FreeSWITCH as a replacement for county wide legacy telephones > for almost a year now. > > The system is almost completely feature functional in comparison to our > Nortel system, however our FreeSWITCH expert has found more profitable > employment elsewhere. > > We are testing the waters to see if there is anyone in the local area that > would be interested in completing the perfection of a couple of features, > assisting in the rollout of this system county-wide, and maintaining this > system. > > This would be a full time employment position, good benefits, and great > longevity. > > Please email me directly at Jeffrey.mcknight at brevardcounty.us if there is > any interest. > > Regards, > > Jeff McKnight > > ----------------------------------------- > > Under Florida Law, email addresses are Public Records. If you do not want > your e-mail address released in response to public record requests, do not > send electronic mail to this entity. Instead, contact this office by phone > or in writing. > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/f1d4cc24/attachment-0001.html From italorossib at gmail.com Tue Jan 13 14:25:33 2015 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 13 Jan 2015 08:25:33 -0300 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: Message-ID: I don't think so. I have several boxes running in production with unixodbc (2.3) and it's pretty stable. Em 13/01/2015 04:03, "John Nash" escreveu: > OK. LuaSQL provides direct access to postgresql but does not support > connection pooling. ""FreeSWITCH Database Handler" looks like a way to go > but I have to go through using ODBC to postgresql right? Can there be any > downside of using ODBC? > > On Tue, Jan 13, 2015 at 3:30 AM, ?talo Rossi > wrote: > >> Hi John, >> >> Give Lua a try: >> >> https://freeswitch.org/confluence/display/FREESWITCH/Lua+with+Database >> >> On Mon, Jan 12, 2015 at 5:27 PM, John Nash >> wrote: >> >>> I need to run a DB query (Postgresql) from dialplan and store results >>> in variables. I came across this link >>> https://wiki.freeswitch.org/wiki/Mod_odbc_query and looks like this is >>> what I need. >>> >>> But I was wondering if I can use direct (without odbc) query to >>> postgresql. Is there any such application available? >>> >>> P.S. I read some posts about postgresql native support but have not been >>> able to figure out what do I need to do to run a postgresql query from >>> dialplan and store results with out odbc >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> ?talo Rossi >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/0bfdba31/attachment.html From GeorgePhelps at gfphelps.com Tue Jan 13 14:29:28 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 13 Jan 2015 06:29:28 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> Message-ID: <143801d02f24$31ccae10$95660a30$@gfphelps.com> For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/984b4249/attachment-0001.html From david.villasmil at gmail.com Tue Jan 13 14:36:02 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 13 Jan 2015 12:36:02 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <143801d02f24$31ccae10$95660a30$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <143801d02f24$31ccae10$95660a30$@gfphelps.com> Message-ID: You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: > For the most recent test/logfile, only extension 1001 was registered ? to > reduce the number of debug messages in the logfile. > > > > 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:* Monday, January 12, 2015 6:16 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Is 1000 registered? The log says it's not registered... > > > > On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > David Govea, > > > > I uploaded a new Freeswitch debug logfile at: > > > > *http://pastebin.com/v17SyXhh * > > > > *Notes* > > > > Only extension 1001 was registered for this test. > > > > Dialstring segment: data="{ignore_early_media=true}user/1000:_:user/1001"/> > > > > I?m guessing that ?*verbal error code 231*? is being generated by my VoIP > service provider. > > > > I am running Freeswitch with (mostly) the default configuration. Changed > passphrases, added my gateway, etc. > > > > I downloaded the source code from git and built it unmodified, from > scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit > (git 1965b3b 2014-12-30 15:06:32Z 64bit)? > > > > My effective codec is G711U ? fully supported throughout the call chain. > > > > 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:* Monday, January 12, 2015 7:15 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > BTW, I've never heard of verbal error code 231, that's why I ask whether > you downloaded and freeswitch from the git... > > > > On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Are you using freeswitch with its default config or did you install > something like fusionpbx? > > Can you please post your log now? the log for the last dial string, where > calls go out and then get hung up. > > (Are you sure your codecs are correct?) > > > > On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > David Govea, > > > > Still fails; both extensions rang. However, before I can answer either > one, I heard the same verbal error code: ?231?. > > > > How do I track down the meaning of ?231?? > > > > 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:* Monday, January 12, 2015 6:14 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > You can also try: > > bridge user/1001:_:user/1002 > > On Jan 12, 2015 12:04 PM, "George F. Phelps" > wrote: > > David Govea, > > > > That syntax, with more than one extension specified, causes the following > Freeswitch warning log message: > > > > [WARNING] switch_ivr_originate.c:2531 Only calling the first element in > the list in this mode. > > > > However, the call ? to only the first extension on the list ? does work. > > > > 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:* Monday, January 12, 2015 3:21 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > try this: > > > > > > > > > > On Jan 12, 2015 4:33 AM, "George F. Phelps" > wrote: > > Here you go: > > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > > > 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, January 11, 2015 10:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > > _________________________________________________________________________ > 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 > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/992fa2bf/attachment-0001.html From GeorgePhelps at gfphelps.com Tue Jan 13 15:49:28 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 13 Jan 2015 07:49:28 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <14 3801d02f24$31ccae10$95660a30$@gfphelps.com> Message-ID: <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 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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 -- DVG -- Imagination is more important than knowledge Albert Einstein -- DVG -- Imagination is more important than knowledge Albert Einstein 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/361b7ec8/attachment-0001.html From david.villasmil at gmail.com Tue Jan 13 15:58:02 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 13 Jan 2015 13:58:02 +0100 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> Message-ID: Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" wrote: > David Govea, > > > > It appears that the essence of the problem is: > > > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] > > > > Various Freeswitch web comments, related to the same problem, indicate > that I should: ?*Ok. Setting it per leg didn't help > [ignore_early_media=true], but per channel {ignore_early_media=true} worked* > ?. > > > > What dialplan(?) syntax do I use to correctly ?set > ignore_early_media=true? on a per channel basis? I tried, within my > dialplan? > > > > > > data="{ignore_early_media=true}user/1000:_:{ignore_early_media=true}user/1001"/> > > > > ?but without success. Or does setting ignore_early_media have to be done > somewhere else? > > > > 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:* Tuesday, January 13, 2015 6:36 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > You need to have both extensions registered. Register both and try again > and paste de log. > > On Jan 13, 2015 12:30 PM, "George F. Phelps" > wrote: > > For the most recent test/logfile, only extension 1001 was registered ? to > reduce the number of debug messages in the logfile. > > > > 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:* Monday, January 12, 2015 6:16 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Is 1000 registered? The log says it's not registered... > > > > On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > David Govea, > > > > I uploaded a new Freeswitch debug logfile at: > > > > *http://pastebin.com/v17SyXhh * > > > > *Notes* > > > > Only extension 1001 was registered for this test. > > > > Dialstring segment: data="{ignore_early_media=true}user/1000:_:user/1001"/> > > > > I?m guessing that ?*verbal error code 231*? is being generated by my VoIP > service provider. > > > > I am running Freeswitch with (mostly) the default configuration. Changed > passphrases, added my gateway, etc. > > > > I downloaded the source code from git and built it unmodified, from > scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit > (git 1965b3b 2014-12-30 15:06:32Z 64bit)? > > > > My effective codec is G711U ? fully supported throughout the call chain. > > > > 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:* Monday, January 12, 2015 7:15 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > BTW, I've never heard of verbal error code 231, that's why I ask whether > you downloaded and freeswitch from the git... > > > > On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Are you using freeswitch with its default config or did you install > something like fusionpbx? > > Can you please post your log now? the log for the last dial string, where > calls go out and then get hung up. > > (Are you sure your codecs are correct?) > > > > On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > David Govea, > > > > Still fails; both extensions rang. However, before I can answer either > one, I heard the same verbal error code: ?231?. > > > > How do I track down the meaning of ?231?? > > > > 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:* Monday, January 12, 2015 6:14 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > You can also try: > > bridge user/1001:_:user/1002 > > On Jan 12, 2015 12:04 PM, "George F. Phelps" > wrote: > > David Govea, > > > > That syntax, with more than one extension specified, causes the following > Freeswitch warning log message: > > > > [WARNING] switch_ivr_originate.c:2531 Only calling the first element in > the list in this mode. > > > > However, the call ? to only the first extension on the list ? does work. > > > > 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:* Monday, January 12, 2015 3:21 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > try this: > > > > > > > > > > On Jan 12, 2015 4:33 AM, "George F. Phelps" > wrote: > > Here you go: > > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > > > 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, January 11, 2015 10:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > > _________________________________________________________________________ > 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 > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/bf55ece8/attachment-0001.html From GeorgePhelps at gfphelps.com Tue Jan 13 16:09:51 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 13 Jan 2015 08:09:51 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> Message-ID: <147601d02f32$37a9f5a0$a6fde0e0$@gfphelps.com> I tried? ?but that did not resolve the problem. 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: Tuesday, January 13, 2015 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" wrote: David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/241dee04/attachment-0001.html From alhakeem at gmail.com Tue Jan 13 16:11:48 2015 From: alhakeem at gmail.com (Abdul Hakeem) Date: Tue, 13 Jan 2015 13:11:48 -0000 Subject: [Freeswitch-users] Fork( ) and Exec ( ) functions Message-ID: Hello, I understand FS makes system calls for sending mails, voicemail and fax. Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on system calls & also, a list of functions which require FS to make system calls ? Thanks in advance. Cheers, Abdul Hakeem From rbarroetavena at gmail.com Tue Jan 13 16:58:30 2015 From: rbarroetavena at gmail.com (Ricardo Barroetavena) Date: Tue, 13 Jan 2015 10:58:30 -0300 Subject: [Freeswitch-users] Multitenant subscription restriction In-Reply-To: References: Message-ID: Hi Vik, Thanks for the answer. We tested auth-subscriptions and as you've said it requires auth but it still allows subscriptions to different domains. On Mon, Jan 12, 2015 at 11:22 AM, Vik Killa wrote: > Hi Ricardo, > I believe you can turn SUBSCRIBE authentication on which would require a > user/password to complete SUBSCRIBE, but if they have a valid user/pass, > they could probably still SUBSCRIBE to other users in other domains. > I hope this helps. > Thanks, > /V > > On Mon, Jan 12, 2015 at 8:59 AM, Ricardo Barroetavena < > rbarroetavena at gmail.com> wrote: > >> Hi, >> In a multitenant environment, is there a way to restrict one tenant UA >> from subscribing to another tenant events? >> >> For example, if I've got bob at domain1 and alice at domain2, is there a way >> to prevent bob from subscribing to let's say alice message-summary events? >> >> Thanks for the hints >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/940cef92/attachment.html From max at nysolutions.com Tue Jan 13 17:15:07 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Tue, 13 Jan 2015 14:15:07 +0000 Subject: [Freeswitch-users] Multitenant subscription restriction In-Reply-To: References: Message-ID: Do you have this on your (internal) profile? force-subscription-domain=$${domain} Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ricardo Barroetavena Sent: Tuesday, January 13, 2015 8:59 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multitenant subscription restriction Hi Vik, Thanks for the answer. We tested auth-subscriptions and as you've said it requires auth but it still allows subscriptions to different domains. On Mon, Jan 12, 2015 at 11:22 AM, Vik Killa > wrote: Hi Ricardo, I believe you can turn SUBSCRIBE authentication on which would require a user/password to complete SUBSCRIBE, but if they have a valid user/pass, they could probably still SUBSCRIBE to other users in other domains. I hope this helps. Thanks, /V On Mon, Jan 12, 2015 at 8:59 AM, Ricardo Barroetavena > wrote: Hi, In a multitenant environment, is there a way to restrict one tenant UA from subscribing to another tenant events? For example, if I've got bob at domain1 and alice at domain2, is there a way to prevent bob from subscribing to let's say alice message-summary events? Thanks for the hints _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/70921aac/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/70921aac/attachment-0001.jpg From steveayre at gmail.com Tue Jan 13 17:18:51 2015 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 13 Jan 2015 14:18:51 +0000 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: Message-ID: Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) It's stable and uses FreeSWITCH's connection pools. It supports any FreeSWITCH DSNs - so sqlite, ODBC or postgresql native driver. https://wiki.freeswitch.org/wiki/DSN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/a09cc508/attachment.html From brian at freeswitch.org Tue Jan 13 19:18:41 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 13 Jan 2015 10:18:41 -0600 Subject: [Freeswitch-users] Fork( ) and Exec ( ) functions In-Reply-To: References: Message-ID: What problem are you trying to solve? On Tue, Jan 13, 2015 at 7:11 AM, Abdul Hakeem wrote: > Hello, > > I understand FS makes system calls for sending mails, voicemail and fax. > Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on > system calls & also, a list of functions which require FS to make system > calls > ? > Thanks in advance. > Cheers, > Abdul Hakeem > > > > _________________________________________________________________________ > 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/20150113/a5cf4831/attachment.html From sharath.kumar at mezocliq.com Tue Jan 13 20:05:25 2015 From: sharath.kumar at mezocliq.com (Kumar, Sharath) Date: Tue, 13 Jan 2015 12:05:25 -0500 Subject: [Freeswitch-users] Convert existing call to a conference and hangup the old call Message-ID: Hello, I am new to Freeswitch and wanted some input on how best to achieve the below scenario. 1. A and B are in a call. 2. A adds user C to the call. - At this point user A and B are added to a dynamically created conf and C is added as well. 3 A adds user D to the call. D is added to the conf. The fact that step 2 is a conf call and users C and D are provisioned and available to the FS. I am using FS_CURL so the PHP has access to the DB. I know how to add A,B,C and D directly into a conference using mod_conference. I made it work. However, I don't understand how I can access an existing call i.e between A and B and turn that into a conference. and also I guess the original call should be hungup when B accepts the conf invite. Is there a way to do this in the dial-plan ? i.e when A calls C, the PHP knows that this is a conf call, how do I kill the old call though. I guess I can use the uuid of the previous call, I don't know what to issue to hang up the first call. My clients are webrtc based and currently they don't natively support either hold/resume or transfer(using refer). They do however support multiple calls from/to the same user at the same time. i.e 2 lines. I guess both will be active. Thank you. Any help would be greatly appreciated!! -Shaks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/5bf061b1/attachment.html From moises.silva at gmail.com Tue Jan 13 20:11:16 2015 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 13 Jan 2015 12:11:16 -0500 Subject: [Freeswitch-users] Fork( ) and Exec ( ) functions In-Reply-To: References: Message-ID: On Tue, Jan 13, 2015 at 8:11 AM, Abdul Hakeem wrote: > Hello, > > I understand FS makes system calls for sending mails, voicemail and fax. > Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on > system calls & also, a list of functions which require FS to make system > calls > ? > Not making much sense here. Everything in FS relies heavily on system calls as it's a multi-threaded-I/O-driven system for the most part. I think you're asking the wrong question. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/20ff5cb4/attachment.html From steveayre at gmail.com Tue Jan 13 20:39:41 2015 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 13 Jan 2015 17:39:41 +0000 Subject: [Freeswitch-users] Convert existing call to a conference and hangup the old call In-Reply-To: References: Message-ID: Try using https://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast On 13 January 2015 at 17:05, Kumar, Sharath wrote: > Hello, > > I am new to Freeswitch and wanted some input on how best to achieve the > below scenario. > > 1. A and B are in a call. > > 2. A adds user C to the call. - At this point user A and B are added to a > dynamically created conf and C is added as well. > > 3 A adds user D to the call. D is added to the conf. > > > The fact that step 2 is a conf call and users C and D are provisioned and > available to the FS. I am using FS_CURL so the PHP has access to the DB. > > > I know how to add A,B,C and D directly into a conference using > mod_conference. I made it work. However, I don't understand how I can > access an existing call i.e between A and B and turn that into a > conference. and also I guess the original call should be hungup when B > accepts the conf invite. > > Is there a way to do this in the dial-plan ? i.e when A calls C, the PHP > knows that this is a conf call, how do I kill the old call though. I guess > I can use the uuid of the previous call, I don't know what to issue to hang > up the first call. > > My clients are webrtc based and currently they don't natively support > either hold/resume or transfer(using refer). They do however support > multiple calls from/to the same user at the same time. i.e 2 lines. I guess > both will be active. > > > Thank you. Any help would be greatly appreciated!! > > -Shaks. > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/407a20ff/attachment.html From bote_radio at botecomm.com Tue Jan 13 22:32:39 2015 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 13 Jan 2015 14:32:39 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <147601d02f32$37a9f5a0$a6fde0e0$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <147601d02f32$37a9f5a0$a6fde0e0$@gfphelps.com> Message-ID: <02bb01d02f67$b21fcd70$165f6850$@com> I suggest you configure and register 3 total local phones to your FS installation, configure 2 of them as the target of your simultaneous ring group, and call them with the 3rd phone. Until you can get that working, calling through a carrier is adding another layer of complexity to the problem and confusing the issue. Out of the box FreeSWITCH does not utter voice codes, they must be coming from your carrier. Also, the debug-level logs very likely tell you exactly what is happening, even though they can be staggering to decipher as a newcomer to FS. Learning how to read them pays off in so many ways, though. I find the color-coded logs on the console or viewed via FS_cli to be helpful in these instances. Bote From: George F. Phelps Sent: Tuesday, 13 January, 2015 08:10 Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I tried? ?but that did not resolve the problem. 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: Tuesday, January 13, 2015 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" wrote: David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/9637a7ab/attachment-0001.html From bote_radio at botecomm.com Tue Jan 13 23:36:24 2015 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 13 Jan 2015 15:36:24 -0500 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: Message-ID: <02c901d02f70$9a2dcd80$ce896880$@com> Surely you mean the Confluence pages at https://freeswitch.org/confluence/display/FREESWITCH/Lua+freeswitch+dbh https://freeswitch.org/confluence/display/FREESWITCH/Databases (formerly DSN) That nasty old wiki is deprecated, I don't know why people are still linking to it. Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Tuesday, 13 January, 2015 09:19 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Database query from dialplan Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) It's stable and uses FreeSWITCH's connection pools. It supports any FreeSWITCH DSNs - so sqlite, ODBC or postgresql native driver. https://wiki.freeswitch.org/wiki/DSN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/8b61dfcd/attachment.html From david.villasmil at gmail.com Tue Jan 13 23:54:59 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Tue, 13 Jan 2015 21:54:59 +0100 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: <02c901d02f70$9a2dcd80$ce896880$@com> References: <02c901d02f70$9a2dcd80$ce896880$@com> Message-ID: Simple, comes up on Google before confluence :) On Jan 13, 2015 9:37 PM, "Bote Man" wrote: > Surely you mean the Confluence pages at > > https://freeswitch.org/confluence/display/FREESWITCH/Lua+freeswitch+dbh > > > > https://freeswitch.org/confluence/display/FREESWITCH/Databases (formerly > DSN) > > > > That nasty old wiki is deprecated, I don't know why people are still > linking to it. > > > > Bote > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* Tuesday, 13 January, 2015 09:19 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Database query from dialplan > > > > Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) > > > > It's stable and uses FreeSWITCH's connection pools. > > > > It supports any FreeSWITCH DSNs - so sqlite, ODBC or postgresql native > driver. > > https://wiki.freeswitch.org/wiki/DSN > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/ed6cb621/attachment.html From sharath.kumar at mezocliq.com Wed Jan 14 00:07:05 2015 From: sharath.kumar at mezocliq.com (Kumar, Sharath) Date: Tue, 13 Jan 2015 16:07:05 -0500 Subject: [Freeswitch-users] Convert existing call to a conference and hangup the old call In-Reply-To: References: Message-ID: Thanks. I found another way by using uuid_kill. On Tue, Jan 13, 2015 at 12:39 PM, Steven Ayre wrote: > Try using https://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast > > > > On 13 January 2015 at 17:05, Kumar, Sharath > wrote: > >> Hello, >> >> I am new to Freeswitch and wanted some input on how best to achieve the >> below scenario. >> >> 1. A and B are in a call. >> >> 2. A adds user C to the call. - At this point user A and B are added to a >> dynamically created conf and C is added as well. >> >> 3 A adds user D to the call. D is added to the conf. >> >> >> The fact that step 2 is a conf call and users C and D are provisioned and >> available to the FS. I am using FS_CURL so the PHP has access to the DB. >> >> >> I know how to add A,B,C and D directly into a conference using >> mod_conference. I made it work. However, I don't understand how I can >> access an existing call i.e between A and B and turn that into a >> conference. and also I guess the original call should be hungup when B >> accepts the conf invite. >> >> Is there a way to do this in the dial-plan ? i.e when A calls C, the PHP >> knows that this is a conf call, how do I kill the old call though. I guess >> I can use the uuid of the previous call, I don't know what to issue to hang >> up the first call. >> >> My clients are webrtc based and currently they don't natively support >> either hold/resume or transfer(using refer). They do however support >> multiple calls from/to the same user at the same time. i.e 2 lines. I guess >> both will be active. >> >> >> Thank you. Any help would be greatly appreciated!! >> >> -Shaks. >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/ec708069/attachment.html From bote_radio at botecomm.com Wed Jan 14 00:38:59 2015 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 13 Jan 2015 16:38:59 -0500 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: <02c901d02f70$9a2dcd80$ce896880$@com> Message-ID: <02e601d02f79$581035b0$0830a110$@com> Actually, the search utility within Confluence is pretty good. If you just send people to https://confluence.freeswitch.org They can find what they want rather quickly, with the proper pointer. If something is still missing, we welcome the help in moving pages over to Confluence from the old wiki. Cheers! Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Tuesday, 13 January, 2015 15:55 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Database query from dialplan Simple, comes up on Google before confluence :) On Jan 13, 2015 9:37 PM, "Bote Man" wrote: Surely you mean the Confluence pages at https://freeswitch.org/confluence/display/FREESWITCH/Lua+freeswitch+dbh https://freeswitch.org/confluence/display/FREESWITCH/Databases (formerly DSN) That nasty old wiki is deprecated, I don't know why people are still linking to it. Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Tuesday, 13 January, 2015 09:19 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Database query from dialplan Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) It's stable and uses FreeSWITCH's connection pools. It supports any FreeSWITCH DSNs - so sqlite, ODBC or postgresql native driver. https://wiki.freeswitch.org/wiki/DSN _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/01e03047/attachment-0001.html From bote_radio at botecomm.com Wed Jan 14 00:49:52 2015 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 13 Jan 2015 16:49:52 -0500 Subject: [Freeswitch-users] Convert existing call to a conference and hangup the old call In-Reply-To: References: Message-ID: <02eb01d02f7a$dd15e6a0$9741b3e0$@com> Each "leg" of the call is represented by a uuid so you would be moving a uuid into the conference bridge, not hanging up and placing a separate call. I imagine this should be trivial from your PHP script. https://confluence.freeswitch.org Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kumar, Sharath Sent: Tuesday, 13 January, 2015 16:07 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Convert existing call to a conference and hangup the old call Thanks. I found another way by using uuid_kill. On Tue, Jan 13, 2015 at 12:39 PM, Steven Ayre wrote: Try using https://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast On 13 January 2015 at 17:05, Kumar, Sharath wrote: Hello, I am new to Freeswitch and wanted some input on how best to achieve the below scenario. 1. A and B are in a call. 2. A adds user C to the call. - At this point user A and B are added to a dynamically created conf and C is added as well. 3 A adds user D to the call. D is added to the conf. The fact that step 2 is a conf call and users C and D are provisioned and available to the FS. I am using FS_CURL so the PHP has access to the DB. I know how to add A,B,C and D directly into a conference using mod_conference. I made it work. However, I don't understand how I can access an existing call i.e between A and B and turn that into a conference. and also I guess the original call should be hungup when B accepts the conf invite. Is there a way to do this in the dial-plan ? i.e when A calls C, the PHP knows that this is a conf call, how do I kill the old call though. I guess I can use the uuid of the previous call, I don't know what to issue to hang up the first call. My clients are webrtc based and currently they don't natively support either hold/resume or transfer(using refer). They do however support multiple calls from/to the same user at the same time. i.e 2 lines. I guess both will be active. Thank you. Any help would be greatly appreciated!! -Shaks. _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/3feb510e/attachment.html From sharath.kumar at mezocliq.com Wed Jan 14 01:24:18 2015 From: sharath.kumar at mezocliq.com (Kumar, Sharath) Date: Tue, 13 Jan 2015 17:24:18 -0500 Subject: [Freeswitch-users] Convert existing call to a conference and hangup the old call In-Reply-To: <02eb01d02f7a$dd15e6a0$9741b3e0$@com> References: <02eb01d02f7a$dd15e6a0$9741b3e0$@com> Message-ID: When you say "moving" do you imply a uuid_transfer ? I am not clear how to "move" the call legs that are answered silently to a conf-extension. Can you please enlighten me ? Shaks On Tue, Jan 13, 2015 at 4:49 PM, Bote Man wrote: > Each "leg" of the call is represented by a uuid so you would be moving a > uuid into the conference bridge, not hanging up and placing a separate > call. I imagine this should be trivial from your PHP script. > > > > https://confluence.freeswitch.org > > > > Bote > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Kumar, > Sharath > *Sent:* Tuesday, 13 January, 2015 16:07 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Convert existing call to a conference > and hangup the old call > > > > Thanks. I found another way by using uuid_kill. > > > > On Tue, Jan 13, 2015 at 12:39 PM, Steven Ayre wrote: > > Try using https://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast > > > > > > > > On 13 January 2015 at 17:05, Kumar, Sharath > wrote: > > Hello, > > > > I am new to Freeswitch and wanted some input on how best to achieve the > below scenario. > > > > 1. A and B are in a call. > > > > 2. A adds user C to the call. - At this point user A and B are added to a > dynamically created conf and C is added as well. > > > > 3 A adds user D to the call. D is added to the conf. > > > > > > The fact that step 2 is a conf call and users C and D are provisioned and > available to the FS. I am using FS_CURL so the PHP has access to the DB. > > > > > > I know how to add A,B,C and D directly into a conference using > mod_conference. I made it work. However, I don't understand how I can > access an existing call i.e between A and B and turn that into a > conference. and also I guess the original call should be hungup when B > accepts the conf invite. > > > > Is there a way to do this in the dial-plan ? i.e when A calls C, the PHP > knows that this is a conf call, how do I kill the old call though. I guess > I can use the uuid of the previous call, I don't know what to issue to hang > up the first call. > > > > My clients are webrtc based and currently they don't natively support > either hold/resume or transfer(using refer). They do however support > multiple calls from/to the same user at the same time. i.e 2 lines. I guess > both will be active. > > > > > > Thank you. Any help would be greatly appreciated!! > > > > -Shaks. > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/7821d8c3/attachment-0001.html From GeorgePhelps at gfphelps.com Wed Jan 14 01:42:56 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Tue, 13 Jan 2015 17:42:56 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <02bb01d02f67$b21fcd70$165f6850$@com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <147601d02f32$37a9f5a0$a6fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> Message-ID: <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> New logfile uploaded to: http://pastebin.com/CFFvVarS The log contains default Freeswitch console log messages, plus a SIP trace of a failed call. BTW, both extensions were ringing ? prior to the CANCEL message (see context below). In the log I see the INVITE from my VoIP service provider: recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: ------------------------------------------------------------------------ INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: ------------------------------------------------------------------------ SIP/2.0 100 Trying (Then, subsequent INVITE messages to my two extensions. But other no messages to/from my VoIP service provider.) And then, a spontaneous CANCEL from my VoIP service provider, approximately 10 seconds after the initial INVITE message. Due to a SIP ?Timer B? timeout? Seems way too short. recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: ------------------------------------------------------------------------ CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (Freeswitch cleanup of SIP sessions to my extensions?) Bote Man--> I have two local extensions. Individually, the extensions can make and receive both internal and external calls. It?s only the simultaneous ringing for external, inbound calls that is not working at the moment. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Tuesday, January 13, 2015 2:33 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I suggest you configure and register 3 total local phones to your FS installation, configure 2 of them as the target of your simultaneous ring group, and call them with the 3rd phone. Until you can get that working, calling through a carrier is adding another layer of complexity to the problem and confusing the issue. Out of the box FreeSWITCH does not utter voice codes, they must be coming from your carrier. Also, the debug-level logs very likely tell you exactly what is happening, even though they can be staggering to decipher as a newcomer to FS. Learning how to read them pays off in so many ways, though. I find the color-coded logs on the console or viewed via FS_cli to be helpful in these instances. Bote From: George F. Phelps Sent: Tuesday, 13 January, 2015 08:10 Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I tried? ?but that did not resolve the problem. 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: Tuesday, January 13, 2015 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" wrote: David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/804a1454/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/20150113/804a1454/attachment-0001.bin From msc at freeswitch.org Wed Jan 14 03:52:07 2015 From: msc at freeswitch.org (Michael Collins) Date: Tue, 13 Jan 2015 16:52:07 -0800 Subject: [Freeswitch-users] How can i hangup call if there is no audio or dead air on channel. In-Reply-To: References: Message-ID: Hi Aqs, Just for clarification - what are the conditions where you want to hang up? And is this something automatic that you want to do? In other words, are you looking to have "something" listening in on a channel, detect a certain condition, and then tear it down? I just want to make sure that we understand what you are trying to accomplish. Thanks, MC On Mon, Jan 12, 2015 at 11:34 AM, Aqs Younas wrote: > Hi, list > > I am playing stream and sometimes streams went down and i hear on audio on > my soft phone. i want to hangup call on whenever this happens. > > But don't know how can i do so. > > Any help would be much appreciated. > > Thanks > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/8e6e6664/attachment.html From msc at freeswitch.org Wed Jan 14 03:57:39 2015 From: msc at freeswitch.org (Michael Collins) Date: Tue, 13 Jan 2015 16:57:39 -0800 Subject: [Freeswitch-users] postgresql In-Reply-To: <1420922004.999033200@f404.i.mail.ru> References: <1420922004.999033200@f404.i.mail.ru> Message-ID: Is it an absolute requirement to log directly to the database? There has been thorough and intense discussion on this list over the years as to why this is/isn't a good idea. IMHO the best option is to use mod_xml_cdr (see https://wiki.freeswitch.org/wiki/Mod_xml_cdr). It lets you post to a database and store on disk in case of failure. It's also pretty easy to use and lets you do logic/validation prior to storing in the db. -MC On Sat, Jan 10, 2015 at 12:33 PM, Nick Zaitsev wrote: > Good day to you, > Please,advise me, how can i use postgres schema in freeswitch config > (cdr_pg_csv.conf.xml)? > I'd like to use cdr schema instead public, for example here > > > -- > Nick Zaitsev > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/7fc7aeba/attachment.html From msc at freeswitch.org Wed Jan 14 04:02:30 2015 From: msc at freeswitch.org (Michael Collins) Date: Tue, 13 Jan 2015 17:02:30 -0800 Subject: [Freeswitch-users] Freeswitch only records empty files In-Reply-To: References: <1420812082840-7596139.post@n2.nabble.com> Message-ID: Bypass media means the endpoints send their RTP directly to each other, bypassing the server (FreeSWITCH), however I'm not sure that's exactly what's going on here. The only way to know for sure is to get a look at the FreeSWITCH debug log. Can you paste a debug log of the code that generates the call and the call log at pastebin.freeswitch.org? -MC On Mon, Jan 12, 2015 at 4:15 AM, bee wrote: > What do you mean? > > Stephen Wilde writes: > > > > > > > This can happens if you bypass media > > > > On Fri, Jan 9, 2015 at 3:01 PM, bee XR3nNK59k0GoYr4blSSd5g at public.gmane.org> wrote:Hey, i got an script > which makes a call to an outbound number (so only 1 > > channel is open) and then i want to record the input. But the record > > function only create empty 44kb files. When i join the channel over a > > softphone the recorded files have input. > > Any ideas? > > -- > > View this message in context: http://freeswitch- > users.2379917.n2.nabble.com/Freeswitch-only-records-empty-files- > tp7596139.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > ___________________________________________________________________ > ______ > > Professional FreeSWITCH Consulting Services:consulting- > YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsoluti > ons.com > > Official FreeSWITCH > Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.c > luecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users > lists.freeswitch.orghttp:// > lists.freeswitch.org/mailman/listinfo/freeswitch- > users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > usershttp://www.freeswitch.org > > > > > > > > > > > > > > > ___________________________________________________________________ > ______ > > Professional FreeSWITCH Consulting Services: > > consulting at ... > > 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 ... > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/6b57fc25/attachment.html From rstevens at robcoit.com Wed Jan 14 04:06:42 2015 From: rstevens at robcoit.com (Robert Stevens) Date: Wed, 14 Jan 2015 01:06:42 +0000 Subject: [Freeswitch-users] Cisco SPA + BLF behavior In-Reply-To: References: Message-ID: <1421197605115.7086@robcoit.com> Hello Gabriel, What revision of Freeswitch are you running currently? Best regards, Robert Stevens rstevens at robcoit.com ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Gabriel Kuri Sent: Tuesday, January 13, 2015 2:13 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Cisco SPA + BLF behavior I noticed that on a Cisco SPA500 series phone, when configured to do BLF+SD+CP for an extension, the LED for that line on the SPA does not turn red when the line goes off-hook on another phone. It turns RED when the call is actually in progress (ie after the number has been dialed and it's ringing). This behavior is different than when the same line is configured as a shared line appearance on the SPA, the LED turns red immedialtely when the line goes off hook on another phone. It appears FreeSWITCH is sending the same SIP NOTIFY event in both cases with line showing seized (appearance-index=1;appearance-state=seized), so I'm think this might be a bug in the phone's firmware, which is running the latest version, 7.5.6a. I did try a couple older version and had the same results. Am I missing something or is this the behavior of this phone? Should I bother to open a TAC case with Cisco, the behavior seems buggy to me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/5119778e/attachment-0001.html From luis.daniel.lucio at gmail.com Wed Jan 14 05:39:13 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 13 Jan 2015 21:39:13 -0500 Subject: [Freeswitch-users] Mod LCR SQL sub-query for better routing In-Reply-To: References: <52AA1192.6020005@gmail.com> <52AB750E.10405@freeswitch.org> Message-ID: You may want to homogenize the rates. As i see, rating list is a tree, and the digits 999 it is only the branch len. Homogenizing lenghts will solve your problem. You may need to code and offline program to do that. Its not that complex. 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 2014-11-28 20:05 GMT-05:00 DP . : > Victor, I take back my initial response on this old email. > > We ran into a couple of cases with some carriers and multiple matching > prefixes. After actually trying your sub query (modified for Mysql), it > actually does return the true lowest rate from a carrier while importantly > respecting the longest match per carrier. Unlike the reorder_by_rate > function that does not respect the longest match per carrier. That function > simply returns the lowest rate, period. > > So thanks! > > ________________________________ > From: hi-tecc at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: RE: [Freeswitch-users] Mod LCR SQL sub-query for better routing > Date: Mon, 16 Dec 2013 18:36:06 -0500 > > > I agree. This definitely sounds like he simply needed the "reorder_by_rate" > param. It will reorder the initial sql results strictly by rate: > > reorder_by_rate - Forces the LCR module to re-order the query strictly on > rate basis. By default this is turned off, but enabling this will always > prefer rate over anything else. > > > Beware this may have an adverse effect! I initially had this turned on then > quickly realized it would sometimes try to route ALL calls by the lowest > rate found. > > Ex: flowroute lists all calls for the US with a default NPA of "1" at .0098. > Now a user trying to call Jamaica with "1876" at a rate of 0.19 (or > whatever) will get both flowroute rates returned. The reorder by rate will > assume 0.0098 is a valid rate since it will now be the "cheapest" in the > list and send the call along its way to flowroute, whom will now bill you at > 0.19. Now if you have another carrier in your list with 1876 at 0.15 you can > see why this would be a problem. > > In this case you will always want the longest matched NPANXX rate. > > ________________________________ > Date: Fri, 13 Dec 2013 15:58:54 -0500 > From: intralanman at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Mod LCR SQL sub-query for better routing > > On 12/12/2013 02:42 PM, Victor Chukalovskiy wrote: > > Hello, > > For those interested, I added a piece to mod lcr wiki. It makes sorting > / routing logic better than default logic: > > http://wiki.freeswitch.org/wiki/Mod_lcr#Custom_SQL_with_sub-query_-_for_real-life_ratesheet_complexities > > Why it helps: > > Rates rates can often be given both on per-NPA or per-NPANXX level > depending on the carrier and on the NPA. Moreover, some carriers may > have NPANXX rate lower than the corresponding NPA rate, while others > will have it inverse. Neither simple ORDER BY rate, prefix; nor ORDER BY > prefix, rate; give the truly cheapest route. The LCR logic should be > two-step process to accommodate this. > > Cheers, > -Victor > > Unless I misunderstand what you're saying, this is what the reorder_by_rate > param does. > > You'll always want to pick the longest digit match per carrier. Then you > probably want to grab the cheapest overall rate of the matches you got back. > > -Ray > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server http://www.cudatel.com Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.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 From david.villasmil at gmail.com Wed Jan 14 06:35:39 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 14 Jan 2015 04:35:39 +0100 Subject: [Freeswitch-users] postgresql In-Reply-To: References: <1420922004.999033200@f404.i.mail.ru> Message-ID: You can even do the billing with it... I've had it running in production for years, not a single issue... Is it an absolute requirement to log directly to the database? There has been thorough and intense discussion on this list over the years as to why this is/isn't a good idea. IMHO the best option is to use mod_xml_cdr (see https://wiki.freeswitch.org/wiki/Mod_xml_cdr). It lets you post to a database and store on disk in case of failure. It's also pretty easy to use and lets you do logic/validation prior to storing in the db. -MC On Sat, Jan 10, 2015 at 12:33 PM, Nick Zaitsev wrote: > Good day to you, > Please,advise me, how can i use postgres schema in freeswitch config > (cdr_pg_csv.conf.xml)? > I'd like to use cdr schema instead public, for example here > > > -- > Nick Zaitsev > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/1e7b0bc3/attachment.html From jonlederman at gmail.com Wed Jan 14 08:01:57 2015 From: jonlederman at gmail.com (Jon Lederman) Date: Wed, 14 Jan 2015 00:01:57 -0500 Subject: [Freeswitch-users] SDP Renegotiation Message-ID: Hi, I am trying to get my client to work with Freeswitch in the following context to no avail. If I initiate a video call, the video works fine. However, if I start an audio call and then try to add video later in the call, no video is received. I tried adding this line to vars.xml: ( Message-ID: FreeSWITCH is Licensed under the MPL1.1 ... View any of the source file headers to see the particulars. On 1/13/15, 11:10 PM, "Craig Stevenson" wrote: > > I am trying to track down current FreeSWITCH licensing information. > > I found http://wiki.freeswitch.org/wiki/Licensing but that was last updated > two years ago and is in the obsolete wiki.? I suspect there are a few > additional components added over the past few years. > > I'm sure it is there somewhere on the new site.? But searching for "License" > or "Licensing" on freeswtich.org page doesn't find it > for me. > > Thanks, > Craig > > > > _________________________________________________________________________ > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/43a20c60/attachment-0001.html From daveh at beachdognet.com Wed Jan 14 09:05:51 2015 From: daveh at beachdognet.com (Dave Horton) Date: Wed, 14 Jan 2015 01:05:51 -0500 Subject: [Freeswitch-users] freeswitch reuses RTP port when bridging second call to user Message-ID: <97E00FD6-6BE6-4C56-9652-FE78EA4F4513@beachdognet.com> I have a problem relating the handling of a second incoming call to same extension/user. The scenario is this: - call from PSTN comes to freeswitch; freeswitch bridges the call to the identified registered device; call is set up fine (two-way audio) - a second call from the PSTN to the same DID comes to freeswitch; freeswitch bridges the call to the same identified registered device; call has no audio In looking at the traces, the problem stems from the fact that for the second call freeswitch is sending an INVITE offering an SDP with the same source ip and port as the first call. This causes the subsequent INVITE on hold from the phone (which is accepted by the freeswitch) to cause the phone not to send any RTP; meanwhile, the RTP sent from the freeswitch to the phone for the second call is rejected with an ICMP port unreachable since the phone has successfully invited that port on hold. In other words, by re-using the same port for call 1 and call 2, when call 1 is invited on hold the result means no media for call 2. Is this known, or expected behavior? Is this a bug? Has this been fixed in a more recent release (I'm on a very old release) Dave From mishehu at freeswitch.org Wed Jan 14 09:21:43 2015 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Wed, 14 Jan 2015 00:21:43 -0600 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: <02e601d02f79$581035b0$0830a110$@com> References: <02c901d02f70$9a2dcd80$ce896880$@com> <02e601d02f79$581035b0$0830a110$@com> Message-ID: <54B60AF7.5090103@freeswitch.org> Thought that per Ken it is http://freeswitch.org/confluence now... Yossi On 01/13/2015 03:38 PM, Bote Man wrote: > Actually, the search utility within Confluence is pretty good. If you > just send people to > > https://confluence.freeswitch.org > > They can find what they want rather quickly, with the proper pointer. > > If something is still missing, we welcome the help in moving pages > over to Confluence from the old wiki. > > Cheers! > > Bote > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *David Villasmil Govea > *Sent:* Tuesday, 13 January, 2015 15:55 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Database query from dialplan > > Simple, comes up on Google before confluence :) > > On Jan 13, 2015 9:37 PM, "Bote Man" > wrote: > > Surely you mean the Confluence pages at > > https://freeswitch.org/confluence/display/FREESWITCH/Lua+freeswitch+dbh > > https://freeswitch.org/confluence/display/FREESWITCH/Databases > (formerly DSN) > > That nasty old wiki is deprecated, I don't know why people are still > linking to it. > > Bote > > *From:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of > *Steven Ayre > *Sent:* Tuesday, 13 January, 2015 09:19 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Database query from dialplan > > Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) > > It's stable and uses FreeSWITCH's connection pools. > > It supports any FreeSWITCH DSNs - so sqlite, ODBC or postgresql native > driver. > > https://wiki.freeswitch.org/wiki/DSN > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/9ad531b7/attachment.html From gkuri at ieee.org Wed Jan 14 09:36:29 2015 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 13 Jan 2015 22:36:29 -0800 Subject: [Freeswitch-users] Cisco SPA + BLF behavior In-Reply-To: <1421197605115.7086@robcoit.com> References: <1421197605115.7086@robcoit.com> Message-ID: The latest "release", 1.4.15. I don't think it's Freeswitch, as I do see the the appropriate SIP NOTIFY messages coming from Freeswitch to tell the phone the line has been taken off hook (seized), but the phone just doesn't change the light to red, until an actual call is in progress. However when the line on the SPA is setup as a shared line appearance, it works fine when a remote phone takes the line off hook, it's seems to be the BLF mode that doesn't work correctly. On Tue, Jan 13, 2015 at 5:06 PM, Robert Stevens wrote: > Hello Gabriel, > > > What revision of Freeswitch are you running currently? > > > > Best regards, > > Robert Stevens > *rstevens at robcoit.com * > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org < > freeswitch-users-bounces at lists.freeswitch.org> on behalf of Gabriel Kuri < > gkuri at ieee.org> > *Sent:* Tuesday, January 13, 2015 2:13 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Cisco SPA + BLF behavior > > I noticed that on a Cisco SPA500 series phone, when configured to do > BLF+SD+CP for an extension, the LED for that line on the SPA does not turn > red when the line goes off-hook on another phone. It turns RED when the > call is actually in progress (ie after the number has been dialed and it's > ringing). This behavior is different than when the same line is configured > as a shared line appearance on the SPA, the LED turns red immedialtely when > the line goes off hook on another phone. It appears FreeSWITCH is sending > the same SIP NOTIFY event in both cases with line showing seized > (appearance-index=1;appearance-state=seized), so I'm think this might be a > bug in the phone's firmware, which is running the latest version, 7.5.6a. I > did try a couple older version and had the same results. Am I missing > something or is this the behavior of this phone? Should I bother to open a > TAC case with Cisco, the behavior seems buggy to me? > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/78a55b9f/attachment-0001.html From john.nash778 at gmail.com Wed Jan 14 10:24:16 2015 From: john.nash778 at gmail.com (John Nash) Date: Wed, 14 Jan 2015 12:54:16 +0530 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: <02c901d02f70$9a2dcd80$ce896880$@com> References: <02c901d02f70$9a2dcd80$ce896880$@com> Message-ID: Thank you. I have compiled freeswitch with postgresql core as per https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core I read many documentation pages but am still confused on few points... 1- Can freeswitch.dbh be used directly as core dsn (without ODBC Like the DSN used at freeswitch startup). If yes would it support connection pooling? 2- Can freeswitch.dbh be used to connect to some different database (Then core database) in lua scripts? On Wed, Jan 14, 2015 at 2:06 AM, Bote Man wrote: > Surely you mean the Confluence pages at > > https://freeswitch.org/confluence/display/FREESWITCH/Lua+freeswitch+dbh > > > > https://freeswitch.org/confluence/display/FREESWITCH/Databases (formerly > DSN) > > > > That nasty old wiki is deprecated, I don't know why people are still > linking to it. > > > > Bote > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* Tuesday, 13 January, 2015 09:19 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Database query from dialplan > > > > Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) > > > > It's stable and uses FreeSWITCH's connection pools. > > > > It supports any FreeSWITCH DSNs - so sqlite, ODBC or postgresql native > driver. > > https://wiki.freeswitch.org/wiki/DSN > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/77901cfe/attachment.html From john.nash778 at gmail.com Wed Jan 14 10:33:14 2015 From: john.nash778 at gmail.com (John Nash) Date: Wed, 14 Jan 2015 13:03:14 +0530 Subject: [Freeswitch-users] postgresql In-Reply-To: References: <1420922004.999033200@f404.i.mail.ru> Message-ID: Hello David and Michael, As per this page https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_pg_csv we can define disk spooling in case of insert failure. What are the other advantages of xml_cdr over cdr_pg_csv? Also does cdr_pg_csv use connection pooling? Regards Manoj On Wed, Jan 14, 2015 at 9:05 AM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > You can even do the billing with it... I've had it running in production > for years, not a single issue... > Is it an absolute requirement to log directly to the database? There has > been thorough and intense discussion on this list over the years as to why > this is/isn't a good idea. IMHO the best option is to use mod_xml_cdr (see > https://wiki.freeswitch.org/wiki/Mod_xml_cdr). It lets you post to a > database and store on disk in case of failure. It's also pretty easy to use > and lets you do logic/validation prior to storing in the db. > > -MC > > > On Sat, Jan 10, 2015 at 12:33 PM, Nick Zaitsev > wrote: > >> Good day to you, >> Please,advise me, how can i use postgres schema in freeswitch config >> (cdr_pg_csv.conf.xml)? >> I'd like to use cdr schema instead public, for example here >> >> >> -- >> Nick Zaitsev >> >> _________________________________________________________________________ >> 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/556f65dd/attachment.html From msc at freeswitch.org Wed Jan 14 10:40:51 2015 From: msc at freeswitch.org (Michael Collins) Date: Tue, 13 Jan 2015 23:40:51 -0800 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <147601d02f32$37a9f5a0$a6fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> Message-ID: We covered this nicely in chapter 1 of the FreeSWITCH Cookbook I'm sorry that I'm late to the party so I am missing some information. Can you pastebin not only the call log but also the dialplan code for the example in question? One other tip: it appears that the log that you are pasting is coming directly from the FreeSWITCH console. By default the console does not have debug level output enabled. Try entering the command "console loglevel debug" and you'll see way more log lines, mostly yellow text. Those lines will most likely contain the clues needed to unravel this mystery. Thanks, MC On Tue, Jan 13, 2015 at 2:42 PM, George F. Phelps wrote: > New logfile uploaded to: > > > > *http://pastebin.com/CFFvVarS * > > > > The log contains default Freeswitch console log messages, plus a SIP trace > of a failed call. BTW, both extensions were ringing ? prior to the CANCEL > message (see context below). > > > > In the log I see the INVITE from my VoIP service provider: > > > > recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: > > ------------------------------------------------------------------------ > > INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 > > > > send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > > > (Then, subsequent INVITE messages to my two extensions. But other no > messages to/from my VoIP service provider.) > > > > And then, a spontaneous CANCEL from my VoIP service provider, > approximately 10 seconds after the initial INVITE message. Due to a SIP > ?Timer B? timeout? Seems way too short. > > > > recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: > > ------------------------------------------------------------------------ > > CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 > > > > (Freeswitch cleanup of SIP sessions to my extensions?) > > > > > > Bote Man--> I have two local extensions. Individually, the extensions > can make and receive both internal and external calls. It?s only the > simultaneous ringing for external, inbound calls that is not working at the > moment. > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bote Man > *Sent:* Tuesday, January 13, 2015 2:33 PM > > *To:* 'FreeSWITCH Users Help' > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > I suggest you configure and register 3 total local phones to your FS > installation, configure 2 of them as the target of your simultaneous ring > group, and call them with the 3rd phone. Until you can get that working, > calling through a carrier is adding another layer of complexity to the > problem and confusing the issue. > > > Out of the box FreeSWITCH does not utter voice codes, they must be coming > from your carrier. > > > > Also, the debug-level logs very likely tell you exactly what is happening, > even though they can be staggering to decipher as a newcomer to FS. > Learning how to read them pays off in so many ways, though. I find the > color-coded logs on the console or viewed via FS_cli to be helpful in these > instances. > > > > Bote > > > > > > *From:* George F. Phelps > *Sent:* Tuesday, 13 January, 2015 08:10 > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > I tried? > > > > ?but that did not resolve the problem. > > > > 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:* Tuesday, January 13, 2015 7:58 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Correct, first endpoint providing audio wins, but you're using > ignore_early_media... > Try using > > Which is global. And I believe in the dial string also is. > But try it anyway. > > On Jan 13, 2015 1:50 PM, "George F. Phelps" > wrote: > > David Govea, > > > > It appears that the essence of the problem is: > > > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] > > > > Various Freeswitch web comments, related to the same problem, indicate > that I should: ?*Ok. Setting it per leg didn't help > [ignore_early_media=true], but per channel {ignore_early_media=true} worked* > ?. > > > > What dialplan(?) syntax do I use to correctly ?set > ignore_early_media=true? on a per channel basis? I tried, within my > dialplan? > > > > > > data="{ignore_early_media=true}user/1000:_:{ignore_early_media=true}user/1001"/> > > > > ?but without success. Or does setting ignore_early_media have to be done > somewhere else? > > > > 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:* Tuesday, January 13, 2015 6:36 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > You need to have both extensions registered. Register both and try again > and paste de log. > > On Jan 13, 2015 12:30 PM, "George F. Phelps" > wrote: > > For the most recent test/logfile, only extension 1001 was registered ? to > reduce the number of debug messages in the logfile. > > > > 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:* Monday, January 12, 2015 6:16 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Is 1000 registered? The log says it's not registered... > > > > On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > David Govea, > > > > I uploaded a new Freeswitch debug logfile at: > > > > *http://pastebin.com/v17SyXhh * > > > > *Notes* > > > > Only extension 1001 was registered for this test. > > > > Dialstring segment: data="{ignore_early_media=true}user/1000:_:user/1001"/> > > > > I?m guessing that ?*verbal error code 231*? is being generated by my VoIP > service provider. > > > > I am running Freeswitch with (mostly) the default configuration. Changed > passphrases, added my gateway, etc. > > > > I downloaded the source code from git and built it unmodified, from > scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit > (git 1965b3b 2014-12-30 15:06:32Z 64bit)? > > > > My effective codec is G711U ? fully supported throughout the call chain. > > > > 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:* Monday, January 12, 2015 7:15 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > BTW, I've never heard of verbal error code 231, that's why I ask whether > you downloaded and freeswitch from the git... > > > > On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Are you using freeswitch with its default config or did you install > something like fusionpbx? > > Can you please post your log now? the log for the last dial string, where > calls go out and then get hung up. > > (Are you sure your codecs are correct?) > > > > On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > David Govea, > > > > Still fails; both extensions rang. However, before I can answer either > one, I heard the same verbal error code: ?231?. > > > > How do I track down the meaning of ?231?? > > > > 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:* Monday, January 12, 2015 6:14 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > You can also try: > > bridge user/1001:_:user/1002 > > On Jan 12, 2015 12:04 PM, "George F. Phelps" > wrote: > > David Govea, > > > > That syntax, with more than one extension specified, causes the following > Freeswitch warning log message: > > > > [WARNING] switch_ivr_originate.c:2531 Only calling the first element in > the list in this mode. > > > > However, the call ? to only the first extension on the list ? does work. > > > > 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:* Monday, January 12, 2015 3:21 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > try this: > > > > > > > > > > On Jan 12, 2015 4:33 AM, "George F. Phelps" > wrote: > > Here you go: > > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > > > 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, January 11, 2015 10:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > > _________________________________________________________________________ > 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 > > > > > > -- > > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > > _________________________________________________________________________ > 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/ccc58b36/attachment-0001.html From msc at freeswitch.org Wed Jan 14 10:45:27 2015 From: msc at freeswitch.org (Michael Collins) Date: Tue, 13 Jan 2015 23:45:27 -0800 Subject: [Freeswitch-users] postgresql In-Reply-To: References: <1420922004.999033200@f404.i.mail.ru> Message-ID: I think David and I agree on these two basic points regarding mod_xml_curl: You can validate/manipulate/massage the data prior to insertion It's uncomplicated, flexible, and reliable As for connection pooling I'll have to defer to those who use mod_cdr_pg_csv. -MC On Tue, Jan 13, 2015 at 11:33 PM, John Nash wrote: > Hello David and Michael, > > As per this page > https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_pg_csv we > can define disk spooling in case of insert failure. What are the other > advantages of xml_cdr over cdr_pg_csv? > > Also does cdr_pg_csv use connection pooling? > > Regards > > Manoj > > On Wed, Jan 14, 2015 at 9:05 AM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> You can even do the billing with it... I've had it running in production >> for years, not a single issue... >> Is it an absolute requirement to log directly to the database? There has >> been thorough and intense discussion on this list over the years as to why >> this is/isn't a good idea. IMHO the best option is to use mod_xml_cdr (see >> https://wiki.freeswitch.org/wiki/Mod_xml_cdr). It lets you post to a >> database and store on disk in case of failure. It's also pretty easy to use >> and lets you do logic/validation prior to storing in the db. >> >> -MC >> >> >> On Sat, Jan 10, 2015 at 12:33 PM, Nick Zaitsev >> wrote: >> >>> Good day to you, >>> Please,advise me, how can i use postgres schema in freeswitch config >>> (cdr_pg_csv.conf.xml)? >>> I'd like to use cdr schema instead public, for example here >>> >>> >>> -- >>> Nick Zaitsev >>> >>> _________________________________________________________________________ >>> 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 >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150113/0a400588/attachment.html From john.nash778 at gmail.com Wed Jan 14 10:53:37 2015 From: john.nash778 at gmail.com (John Nash) Date: Wed, 14 Jan 2015 13:23:37 +0530 Subject: [Freeswitch-users] postgresql In-Reply-To: References: <1420922004.999033200@f404.i.mail.ru> Message-ID: OK I see the point of manipulating CDR data (Even bill CDR as you said). I am just trying to figure out best approach. But I also see a downside of http post as it will keep web server busy too for each call made specially in high volume systems (Assuming user portal/reporting also are being used on same web server) On the other hand if pg_cdr module is used with postgres function we can manipulating CDR too without keeping web server busy. On Wed, Jan 14, 2015 at 1:15 PM, Michael Collins wrote: > I think David and I agree on these two basic points regarding mod_xml_curl: > You can validate/manipulate/massage the data prior to insertion > It's uncomplicated, flexible, and reliable > > As for connection pooling I'll have to defer to those who use > mod_cdr_pg_csv. > -MC > > > On Tue, Jan 13, 2015 at 11:33 PM, John Nash > wrote: > >> Hello David and Michael, >> >> As per this page >> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_pg_csv we >> can define disk spooling in case of insert failure. What are the other >> advantages of xml_cdr over cdr_pg_csv? >> >> Also does cdr_pg_csv use connection pooling? >> >> Regards >> >> Manoj >> >> On Wed, Jan 14, 2015 at 9:05 AM, David Villasmil Govea < >> david.villasmil at gmail.com> wrote: >> >>> You can even do the billing with it... I've had it running in production >>> for years, not a single issue... >>> Is it an absolute requirement to log directly to the database? There has >>> been thorough and intense discussion on this list over the years as to why >>> this is/isn't a good idea. IMHO the best option is to use mod_xml_cdr (see >>> https://wiki.freeswitch.org/wiki/Mod_xml_cdr). It lets you post to a >>> database and store on disk in case of failure. It's also pretty easy to use >>> and lets you do logic/validation prior to storing in the db. >>> >>> -MC >>> >>> >>> On Sat, Jan 10, 2015 at 12:33 PM, Nick Zaitsev >>> wrote: >>> >>>> Good day to you, >>>> Please,advise me, how can i use postgres schema in freeswitch config >>>> (cdr_pg_csv.conf.xml)? >>>> I'd like to use cdr schema instead public, for example here >>>> >>>> >>>> -- >>>> Nick Zaitsev >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >>> _________________________________________________________________________ >>> 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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/e59f023c/attachment-0001.html From GeorgePhelps at gfphelps.com Wed Jan 14 15:27:01 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Wed, 14 Jan 2015 07:27:01 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <147601d02f32$37a9f5a0$a6fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> Message-ID: <007901d02ff5$67075090$3515f1b0$@gfphelps.com> Michael Collins, I already have the book. Thanks! Here?s my dialplan: New log file uploaded to: http://pastebin.com/gnEpPzk9 To me, the most significant event in the log file is the SIP CANCEL message ? starting at line #321: tport.c:3023 tport_deliver() tport_deliver(0x95daa0): msg 0xad8fb0 (437 bytes) from udp/169.XX.XX.XX:5080/sip next=(nil) nta.c:2880 agent_recv_request() nta: received CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (CSeq 1) nta.c:3026 agent_recv_request() nta: CANCEL (1) is going to INVITE (1) I don?t think it?s related, but I am also curious about log file line #285: sres.c:2987 sres_query_report_error() sres(q=0x98b050): reporting error NAME_ERR for SRV _sip._udp.sip.switch2voip.us Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 2:41 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? We covered this nicely in chapter 1 of the FreeSWITCH Cookbook I'm sorry that I'm late to the party so I am missing some information. Can you pastebin not only the call log but also the dialplan code for the example in question? One other tip: it appears that the log that you are pasting is coming directly from the FreeSWITCH console. By default the console does not have debug level output enabled. Try entering the command "console loglevel debug" and you'll see way more log lines, mostly yellow text. Those lines will most likely contain the clues needed to unravel this mystery. Thanks, MC On Tue, Jan 13, 2015 at 2:42 PM, George F. Phelps wrote: New logfile uploaded to: http://pastebin.com/CFFvVarS The log contains default Freeswitch console log messages, plus a SIP trace of a failed call. BTW, both extensions were ringing ? prior to the CANCEL message (see context below). In the log I see the INVITE from my VoIP service provider: recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: ------------------------------------------------------------------------ INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: ------------------------------------------------------------------------ SIP/2.0 100 Trying (Then, subsequent INVITE messages to my two extensions. But other no messages to/from my VoIP service provider.) And then, a spontaneous CANCEL from my VoIP service provider, approximately 10 seconds after the initial INVITE message. Due to a SIP ?Timer B? timeout? Seems way too short. recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: ------------------------------------------------------------------------ CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (Freeswitch cleanup of SIP sessions to my extensions?) Bote Man--> I have two local extensions. Individually, the extensions can make and receive both internal and external calls. It?s only the simultaneous ringing for external, inbound calls that is not working at the moment. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Tuesday, January 13, 2015 2:33 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I suggest you configure and register 3 total local phones to your FS installation, configure 2 of them as the target of your simultaneous ring group, and call them with the 3rd phone. Until you can get that working, calling through a carrier is adding another layer of complexity to the problem and confusing the issue. Out of the box FreeSWITCH does not utter voice codes, they must be coming from your carrier. Also, the debug-level logs very likely tell you exactly what is happening, even though they can be staggering to decipher as a newcomer to FS. Learning how to read them pays off in so many ways, though. I find the color-coded logs on the console or viewed via FS_cli to be helpful in these instances. Bote From: George F. Phelps Sent: Tuesday, 13 January, 2015 08:10 Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I tried? ?but that did not resolve the problem. 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: Tuesday, January 13, 2015 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" wrote: David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein _________________________________________________________________________ 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/5f080608/attachment-0001.html From david.villasmil at gmail.com Wed Jan 14 17:05:36 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 14 Jan 2015 15:05:36 +0100 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: <02c901d02f70$9a2dcd80$ce896880$@com> Message-ID: hello, 1- Can freeswitch.dbh be used directly as core dsn (without ODBC Like the DSN used at freeswitch startup). If yes would it support connection pooling? I believe you must use odbc. As someone said before, yes, it will use connection pooling. 2- Can freeswitch.dbh be used to connect to some different database (Then core database) in lua scripts? Yes, you can connect to any db. In the case of MySQL it's like so: /etc/odbc.ini [freeswitch] Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so SERVER = [SERVER-IP] PORT = 3306 DATABASE = [DATABASE] OPTION = 67108864 USER = [USER] PASSWORD = [PASSWORD] [my_other_db] Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so SERVER = [SERVER-IP] PORT = 3306 DATABASE = [DATABASE] OPTION = 67108864 USER = [USER] PASSWORD = [PASSWORD] /etc/odbcinst.ini [MySQL] Description = MySQL driver Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so Setup = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so UsageCount = 1 FileUsage = 1 Threading = 0 function dbConnect() conn = freeswitch.Dbh("odbc://my_other_db:user:passw"); assert(conn:connected()) end On Wed, Jan 14, 2015 at 8:24 AM, John Nash wrote: > Thank you. > > I have compiled freeswitch with postgresql core as per > https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core > I read many documentation pages but am still confused on few points... > > 1- Can freeswitch.dbh be used directly as core dsn (without ODBC Like the > DSN used at freeswitch startup). If yes would it support connection > pooling? > > 2- Can freeswitch.dbh be used to connect to some different database (Then > core database) in lua scripts? > > > > > > > > On Wed, Jan 14, 2015 at 2:06 AM, Bote Man wrote: > >> Surely you mean the Confluence pages at >> >> https://freeswitch.org/confluence/display/FREESWITCH/Lua+freeswitch+dbh >> >> >> >> https://freeswitch.org/confluence/display/FREESWITCH/Databases (formerly >> DSN) >> >> >> >> That nasty old wiki is deprecated, I don't know why people are still >> linking to it. >> >> >> >> Bote >> >> >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre >> *Sent:* Tuesday, 13 January, 2015 09:19 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Database query from dialplan >> >> >> >> Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) >> >> >> >> It's stable and uses FreeSWITCH's connection pools. >> >> >> >> It supports any FreeSWITCH DSNs - so sqlite, ODBC or postgresql native >> driver. >> >> https://wiki.freeswitch.org/wiki/DSN >> >> _________________________________________________________________________ >> 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 > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/4b90afff/attachment.html From rbarroetavena at gmail.com Wed Jan 14 18:09:28 2015 From: rbarroetavena at gmail.com (Ricardo Barroetavena) Date: Wed, 14 Jan 2015 12:09:28 -0300 Subject: [Freeswitch-users] Multitenant subscription restriction In-Reply-To: References: Message-ID: Hi, We're not using force-subscription-domain. We understood it's meant for specifying a given domain to all subscriptions. We'd need each tenant to use its own domain. Thanks On Tue, Jan 13, 2015 at 11:15 AM, Moishe Grunstein wrote: > Do you have this on your (internal) profile? > > force-subscription-domain=$${domain} > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ricardo > Barroetavena > *Sent:* Tuesday, January 13, 2015 8:59 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Multitenant subscription restriction > > > > Hi Vik, > > Thanks for the answer. We tested auth-subscriptions and as you've said it > requires auth but it still allows subscriptions to different domains. > > > > > > On Mon, Jan 12, 2015 at 11:22 AM, Vik Killa wrote: > > Hi Ricardo, > > I believe you can turn SUBSCRIBE authentication on which would require a > user/password to complete SUBSCRIBE, but if they have a valid user/pass, > they could probably still SUBSCRIBE to other users in other domains. > > I hope this helps. > > Thanks, > > /V > > > > On Mon, Jan 12, 2015 at 8:59 AM, Ricardo Barroetavena < > rbarroetavena at gmail.com> wrote: > > Hi, > > In a multitenant environment, is there a way to restrict one tenant UA > from subscribing to another tenant events? > > > > For example, if I've got bob at domain1 and alice at domain2, is there a way to > prevent bob from subscribing to let's say alice message-summary events? > > > > Thanks for the hints > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/afce3ff5/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/afce3ff5/attachment-0001.jpg From max at nysolutions.com Wed Jan 14 18:19:59 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 14 Jan 2015 15:19:59 +0000 Subject: [Freeswitch-users] Multitenant subscription restriction In-Reply-To: References: Message-ID: That is correct, wanted to make sure you don?t have that set. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ricardo Barroetavena Sent: Wednesday, January 14, 2015 10:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multitenant subscription restriction Hi, We're not using force-subscription-domain. We understood it's meant for specifying a given domain to all subscriptions. We'd need each tenant to use its own domain. Thanks On Tue, Jan 13, 2015 at 11:15 AM, Moishe Grunstein > wrote: Do you have this on your (internal) profile? force-subscription-domain=$${domain} Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ricardo Barroetavena Sent: Tuesday, January 13, 2015 8:59 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multitenant subscription restriction Hi Vik, Thanks for the answer. We tested auth-subscriptions and as you've said it requires auth but it still allows subscriptions to different domains. On Mon, Jan 12, 2015 at 11:22 AM, Vik Killa > wrote: Hi Ricardo, I believe you can turn SUBSCRIBE authentication on which would require a user/password to complete SUBSCRIBE, but if they have a valid user/pass, they could probably still SUBSCRIBE to other users in other domains. I hope this helps. Thanks, /V On Mon, Jan 12, 2015 at 8:59 AM, Ricardo Barroetavena > wrote: Hi, In a multitenant environment, is there a way to restrict one tenant UA from subscribing to another tenant events? For example, if I've got bob at domain1 and alice at domain2, is there a way to prevent bob from subscribing to let's say alice message-summary events? Thanks for the hints _________________________________________________________________________ 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 _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/f20e1858/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/f20e1858/attachment-0001.jpg From alhakeem at gmail.com Wed Jan 14 18:19:16 2015 From: alhakeem at gmail.com (Abdul Hakeem) Date: Wed, 14 Jan 2015 15:19:16 -0000 Subject: [Freeswitch-users] Database query from dialplan In-Reply-To: References: <02c901d02f70$9a2dcd80$ce896880$@com> Message-ID: Hello, Just out of curiosity, has anyone managed to connect to MySQL via Handlersocket ? If yes, would appreciate tips. Cheers, Abdul Hakeem From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Wednesday, January 14, 2015 2:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Database query from dialplan hello, 1- Can freeswitch.dbh be used directly as core dsn (without ODBC Like the DSN used at freeswitch startup). If yes would it support connection pooling? I believe you must use odbc. As someone said before, yes, it will use connection pooling. 2- Can freeswitch.dbh be used to connect to some different database (Then core database) in lua scripts? Yes, you can connect to any db. In the case of MySQL it's like so: /etc/odbc.ini [freeswitch] Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so SERVER = [SERVER-IP] PORT = 3306 DATABASE = [DATABASE] OPTION = 67108864 USER = [USER] PASSWORD = [PASSWORD] [my_other_db] Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so SERVER = [SERVER-IP] PORT = 3306 DATABASE = [DATABASE] OPTION = 67108864 USER = [USER] PASSWORD = [PASSWORD] /etc/odbcinst.ini [MySQL] Description = MySQL driver Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so Setup = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so UsageCount = 1 FileUsage = 1 Threading = 0 function dbConnect() conn = freeswitch.Dbh("odbc://my_other_db:user:passw"); assert(conn:connected()) end On Wed, Jan 14, 2015 at 8:24 AM, John Nash wrote: Thank you. I have compiled freeswitch with postgresql core as per https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core I read many documentation pages but am still confused on few points... 1- Can freeswitch.dbh be used directly as core dsn (without ODBC Like the DSN used at freeswitch startup). If yes would it support connection pooling? 2- Can freeswitch.dbh be used to connect to some different database (Then core database) in lua scripts? On Wed, Jan 14, 2015 at 2:06 AM, Bote Man wrote: Surely you mean the Confluence pages at https://freeswitch.org/confluence/display/FREESWITCH/Lua+freeswitch+dbh https://freeswitch.org/confluence/display/FREESWITCH/Databases (formerly DSN) That nasty old wiki is deprecated, I don't know why people are still linking to it. Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Tuesday, 13 January, 2015 09:19 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Database query from dialplan Use freeswitch.Dbh (https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh) It's stable and uses FreeSWITCH's connection pools. It supports any FreeSWITCH DSNs - so sqlite, ODBC or postgresql native driver. https://wiki.freeswitch.org/wiki/DSN _________________________________________________________________________ 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 -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/690895a9/attachment-0001.html From brian at freeswitch.org Wed Jan 14 18:37:05 2015 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jan 2015 09:37:05 -0600 Subject: [Freeswitch-users] SDP Renegotiation In-Reply-To: References: Message-ID: "I tried adding this line to vars.xml: ( wrote: > Hi, > > I am trying to get my client to work with Freeswitch in the following > context to no avail. If I initiate a video call, the video works fine. > However, if I start an audio call and then try to add video later in the > call, no video is received. I tried adding this line to vars.xml: ( name="renegotiate-codec-on-reinvite" value="true?/> > That, however, did not fix the problem. > > Any help would be greatly appreciated. The SDP negotiation is indicated > below. Thanks in advance. > > I do see in fs_cli that FS has received a new SDP and is trying to > renegotiate as follows: > > INITIAL SDP (AUDIO ONLY): > v=0 > o=FreeSWITCH 1421171138 1421171140 IN IP4 146.148.54.28 > s=FreeSWITCH > c=IN IP4 146.148.54.28 > t=0 0 > m=audio 25746 RTP/AVP 124 101 > a=rtpmap:124 opus/48000/2 > a=fmtp:124 useinbandfec=1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > m=video 26852 RTP/AVP 98 > a=rtpmap:98 VP8/90000 > > NEW SDP WITH ATTEMPT TO ADD VIDEO TO EXISTING SESSIOn: > 2015-01-14 00:55:29.401667 [DEBUG] sofia.c:5828 Remote SDP: > v=0 > o=1000 3968 2153 IN IP4 10.1.10.10 > s=Talk > c=IN IP4 10.1.10.10 > b=AS:380 > t=0 0 > a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics > m=audio 53368 RTP/AVP 124 101 > a=rtpmap:124 opus/48000/2 > a=fmtp:124 useinbandfec=1; stereo=0; sprop-stereo=0 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > m=video 9078 RTP/AVP 103 > a=rtpmap:103 VP8/90000 > > > > _________________________________________________________________________ > 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/20150114/3c776f0b/attachment.html From ahabiba at gmail.com Wed Jan 14 19:08:09 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Wed, 14 Jan 2015 19:08:09 +0300 Subject: [Freeswitch-users] Security Issue Message-ID: Dears, Kindly I noticed a very strange behaviour on Freeswitch that may allow non authorised users to make call through the system below is the log and my notice highlighted, you help will be appreciated. 1-Below is a request coming from not authored IP. 2-However the originating IP is ?142.54.179.218? the from is as below as if it is from the same server: freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at 16:41:34.211099: ------------------------------------------------------------------------ INVITE sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 To: 9009972599796504 From: 1000;tag=e8473b10 Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE User-Agent: sipcli/v1.8 Content-Type: application/sdp Content-Length: 285 v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv 3-Accordingly Freeswitch start to deal with the call normally ------------------------------------------------------------------------ send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 From: 1000;tag=e8473b10 To: 9009972599796504 Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Content-Length: 0 ???????????????????????????????????? 4-as we can see below Freeswitch consider the call coming from my server IP not from the remote IP(My server IP = 177.31.245.177) 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1000 at 177.31.245.177 [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177) Running State Change CS_NEW 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 sofia/external/1000 at 177.31.245.177 receiving invite from 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z 64bit 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel sofia/external/1000 at 177.31.245.177 entering state [received][100] 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 (sofia/external/1000 at 177.31.245.177) State Change CS_NEW -> CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1000 at 177.31.245.177) State NEW 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177) Running State Change CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177) State INIT 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 sofia/external/1000 at 177.31.245.177 SOFIA INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 sofia/external/1000 at 177.31.245.177 Standard INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 (sofia/external/1000 at 177.31.245.177) State Change CS_INIT -> CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177) State INIT going to sleep 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177) Running State Change CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 (sofia/external/1000 at 177.31.245.177) Callstate Change DOWN -> RINGING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 (sofia/external/1000 at 177.31.245.177) State ROUTING 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 sofia/external/1000 at 177.31.245.177 SOFIA ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 sofia/external/1000 at 177.31.245.177 Standard ROUTING 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->9009972599796504 in context public -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/c8b193e7/attachment-0001.html From mike at jerris.com Wed Jan 14 19:19:29 2015 From: mike at jerris.com (Michael Jerris) Date: Wed, 14 Jan 2015 11:19:29 -0500 Subject: [Freeswitch-users] Current FreeSWITCH licensing page? In-Reply-To: References: Message-ID: <5456AD7B-1BDD-4FDA-A834-1B9FDECE08D0@jerris.com> The license info in the debian packages is probably the best maintained list that doesn't require reviewing code to be sure. > On Jan 14, 2015, at 12:26 AM, Ken Rice wrote: > > FreeSWITCH is Licensed under the MPL1.1 ... View any of the source file headers to see the particulars. > > > > On 1/13/15, 11:10 PM, "Craig Stevenson" > wrote: > >> >> I am trying to track down current FreeSWITCH licensing information. >> >> I found http://wiki.freeswitch.org/wiki/Licensing but that was last updated two years ago and is in the obsolete wiki. I suspect there are a few additional components added over the past few years. >> >> I'm sure it is there somewhere on the new site. But searching for "License" or "Licensing" on freeswtich.org > page doesn't find it for me. >> >> Thanks, >> Craig -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/f10400b8/attachment.html From sdevoy at bizfocused.com Wed Jan 14 19:27:20 2015 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 14 Jan 2015 16:27:20 +0000 Subject: [Freeswitch-users] robo caller Message-ID: Does anyone have a sample RoboCaller script? Perhaps I am using the wrong name and that is why I can't find one. I have a doctor's office that wants to automate the reminder calls about appointments to their patients. I am curious how people handle answering machine detection as well? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/457b0812/attachment.html From brian at freeswitch.org Wed Jan 14 19:29:28 2015 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jan 2015 10:29:28 -0600 Subject: [Freeswitch-users] Security Issue In-Reply-To: References: Message-ID: You have some learning to do about VoIP and various security models. See comments inline. On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba wrote: > Dears, > > Kindly I noticed a very strange behaviour on Freeswitch that may allow non > authorised users to make call through the system below is the log and my > notice highlighted, you help will be appreciated. > > 1-Below is a request coming from not authored IP. > 2-However the originating IP is ?142.54.179.218? the from is as below as > if it is from the same server: > The IP was 142.54.179.218... thats the real IP we received the request from. > > freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at > 16:41:34.211099: > ------------------------------------------------------------------------ > INVITE sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 > To: 9009972599796504 > From: 1000;tag=e8473b10 > Via: SIP/2.0/UDP 142.54.179.218:5070 > ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport > Call-ID: d6e1ddab827448435f49ecaf6e613e2e > CSeq: 1 INVITE > Contact: > Max-Forwards: 70 > Allow: INVITE, ACK, CANCEL, BYE > User-Agent: sipcli/v1.8 > Content-Type: application/sdp > Content-Length: 285 > > > v=0 > o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 > s=sipcli > c=IN IP4 142.54.179.218 > t=0 0 > m=audio 5072 RTP/AVP 18 0 8 101 > a=fmtp:101 0-15 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > a=sendrecv > > 3-Accordingly Freeswitch start to deal with the call normally > As it should, Its hitting the external profile which doesn't have authentication on it. > > > ------------------------------------------------------------------------ > send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 142.54.179.218:5070 > ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 > From: 1000;tag=e8473b10 > To: 9009972599796504 > Call-ID: d6e1ddab827448435f49ecaf6e613e2e > CSeq: 1 INVITE > Content-Length: 0 > > > ???????????????????????????????????? > > 4-as we can see below Freeswitch consider the call coming from my server > IP not from the remote IP(My server IP = 177.31.245.177) > No thats false, This is just a channel named thats formed from various bits of data, Its meaningless data and can be set, overridden or changed. This has no bearing on what freeswitch thinks. Most likely its using the host element from the From header. > > 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel > sofia/external/1000 at 177.31.245.177 [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal > sofia/external/1000 at 177.31.245.177 [BREAK] > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal > sofia/external/1000 at 177.31.245.177 [BREAK] > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( > sofia/external/1000 at 177.31.245.177) Running State Change CS_NEW > 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 > sofia/external/1000 at 177.31.245.177 receiving invite from > 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z 64bit > 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel > sofia/external/1000 at 177.31.245.177 entering state [received][100] > 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: > v=0 > o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 > s=sipcli > c=IN IP4 142.54.179.218 > t=0 0 > m=audio 5072 RTP/AVP 18 0 8 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 ( > sofia/external/1000 at 177.31.245.177) State Change CS_NEW -> CS_INIT > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal > sofia/external/1000 at 177.31.245.177 [BREAK] > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 ( > sofia/external/1000 at 177.31.245.177) State NEW > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( > sofia/external/1000 at 177.31.245.177) Running State Change CS_INIT > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( > sofia/external/1000 at 177.31.245.177) State INIT > 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 > sofia/external/1000 at 177.31.245.177 SOFIA INIT > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 > sofia/external/1000 at 177.31.245.177 Standard INIT > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 ( > sofia/external/1000 at 177.31.245.177) State Change CS_INIT -> CS_ROUTING > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal > sofia/external/1000 at 177.31.245.177 [BREAK] > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( > sofia/external/1000 at 177.31.245.177) State INIT going to sleep > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( > sofia/external/1000 at 177.31.245.177) Running State Change CS_ROUTING > 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 ( > sofia/external/1000 at 177.31.245.177) Callstate Change DOWN -> RINGING > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 ( > sofia/external/1000 at 177.31.245.177) State ROUTING > 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 > sofia/external/1000 at 177.31.245.177 SOFIA ROUTING > 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 > sofia/external/1000 at 177.31.245.177 Standard ROUTING > This is someone from outside calling in to your system, the public context is a sandbox to allow you to isolate and route non-authenticated traffic to your internal contexts via the transfer app, per the vanilla config examples. This is also whey you need to understand how to secure FreeSWITCH. > 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing 1000 > <1000>->9009972599796504 in context public > > > > _________________________________________________________________________ > 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/20150114/cbd0dc3a/attachment-0001.html From david.villasmil at gmail.com Wed Jan 14 19:30:50 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 14 Jan 2015 17:30:50 +0100 Subject: [Freeswitch-users] robo caller In-Reply-To: References: Message-ID: That's fairly simple to implement. Look into at mod_amd, orignate and ivr. On Jan 14, 2015 5:28 PM, "Sean Devoy" wrote: > Does anyone have a sample RoboCaller script? Perhaps I am using the > wrong name and that is why I can?t find one. I have a doctor?s office that > wants to automate the reminder calls about appointments to their patients. > > > > I am curious how people handle answering machine detection as well? > > > > Thanks, > > Sean > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/980e711d/attachment.html From max at nysolutions.com Wed Jan 14 19:37:46 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 14 Jan 2015 16:37:46 +0000 Subject: [Freeswitch-users] robo caller In-Reply-To: References: Message-ID: You can also have a look at http://www.newfies-dialer.org/ Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Govea Sent: Wednesday, January 14, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] robo caller That's fairly simple to implement. Look into at mod_amd, orignate and ivr. On Jan 14, 2015 5:28 PM, "Sean Devoy" > wrote: Does anyone have a sample RoboCaller script? Perhaps I am using the wrong name and that is why I can?t find one. I have a doctor?s office that wants to automate the reminder calls about appointments to their patients. I am curious how people handle answering machine detection as well? Thanks, Sean _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/9abdafd0/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/9abdafd0/attachment.jpg From ahabiba at gmail.com Wed Jan 14 19:39:12 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Wed, 14 Jan 2015 19:39:12 +0300 Subject: [Freeswitch-users] Security Issue In-Reply-To: References: Message-ID: Dear Brian, Thank you really for your feedback, what you mentioned is correct, however I understand that any call to comes from external system to public context to processed , the external system IP shall be allowed via ACL, accordingly we shouldn't expect any traffic to processed from a server/IP that is not defined in the ACL, am I correct? Thanks, Ahmed Habiba. From: Brian West > To: FreeSWITCH Users Help > Date: January 14, 2015 at 7:29:28 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue You have some learning to do about VoIP and various security models. See comments inline. On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba > wrote: Dears, Kindly I noticed a very strange behaviour on Freeswitch that may allow non authorised users to make call through the system below is the log and my notice highlighted, you help will be appreciated. 1-Below is a request coming from not authored IP. 2-However the originating IP is ?142.54.179.218? the from is as below as if it is from the same server: The IP was 142.54.179.218... thats the real IP we received the request from. freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at 16:41:34.211099: ------------------------------------------------------------------------ INVITE <>sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 To: 9009972599796504< <>sip:9009972599796504 at 177.31.245.177 > From: 1000< <>sip:1000 at 177.31.245.177 >;tag=e8473b10 Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Contact: < <>sip:1000 at 142.54.179.218:5070 > Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE User-Agent: sipcli/v1.8 Content-Type: application/sdp Content-Length: 285 v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv 3-Accordingly Freeswitch start to deal with the call normally As it should, Its hitting the external profile which doesn't have authentication on it. ------------------------------------------------------------------------ send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 From: 1000< <>sip:1000 at 177.31.245.177 >;tag=e8473b10 To: 9009972599796504< <>sip:9009972599796504 at 177.31.245.177 > Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Content-Length: 0 ???????????????????????????????????? 4-as we can see below Freeswitch consider the call coming from my server IP not from the remote IP(My server IP = 177.31.245.177) No thats false, This is just a channel named thats formed from various bits of data, Its meaningless data and can be set, overridden or changed. This has no bearing on what freeswitch thinks. Most likely its using the host element from the From header. 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1000 at 177.31.245.177 [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_NEW 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 sofia/external/1000 at 177.31.245.177 receiving invite from 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z 64bit 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel sofia/external/1000 at 177.31.245.177 entering state [received][100] 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 (sofia/external/1000 at 177.31.245.177 ) State Change CS_NEW -> CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1000 at 177.31.245.177 ) State NEW 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 sofia/external/1000 at 177.31.245.177 SOFIA INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 sofia/external/1000 at 177.31.245.177 Standard INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 (sofia/external/1000 at 177.31.245.177 ) State Change CS_INIT -> CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT going to sleep 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 (sofia/external/1000 at 177.31.245.177 ) Callstate Change DOWN -> RINGING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 (sofia/external/1000 at 177.31.245.177 ) State ROUTING 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 sofia/external/1000 at 177.31.245.177 SOFIA ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 sofia/external/1000 at 177.31.245.177 Standard ROUTING This is someone from outside calling in to your system, the public context is a sandbox to allow you to isolate and route non-authenticated traffic to your internal contexts via the transfer app, per the vanilla config examples. This is also whey you need to understand how to secure FreeSWITCH. 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->9009972599796504 in context public _________________________________________________________________________ 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/20150114/b2362f01/attachment-0001.html From areski at gmail.com Wed Jan 14 19:44:43 2015 From: areski at gmail.com (Areski) Date: Wed, 14 Jan 2015 17:44:43 +0100 Subject: [Freeswitch-users] robo caller In-Reply-To: References: Message-ID: Newfies-Dialer (http://www.newfies-dialer.org/) might help and obviously it's built on top of FreeSWITCH. We built a flexible module for appointment reminders: http://docs.newfies-dialer.org/en/latest/user-guide-doc/appointment.html If you want to code this yourself, we use ESL ( https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library) to originate the calls and Lua to build the IVR part ( https://freeswitch.org/confluence/display/FREESWITCH/mod_lua). On Wed, Jan 14, 2015 at 5:37 PM, Moishe Grunstein wrote: > You can also have a look at http://www.newfies-dialer.org/ > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Wednesday, January 14, 2015 11:31 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] robo caller > > > > That's fairly simple to implement. Look into at mod_amd, orignate and ivr. > > On Jan 14, 2015 5:28 PM, "Sean Devoy" wrote: > > Does anyone have a sample RoboCaller script? Perhaps I am using the > wrong name and that is why I can?t find one. I have a doctor?s office that > wants to automate the reminder calls about appointments to their patients. > > > > I am curious how people handle answering machine detection as well? > > > > Thanks, > > Sean > > > _________________________________________________________________________ > 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 > -- Kind regards, /Areski ---- Arezqui Belaid, Founder at Star2Billing (www.star2billing.com) Tel: +34650784355 Twitter: http://twitter.com/areskib LinkedIn: http://www.linkedin.com/in/areski -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/73fc9cee/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/73fc9cee/attachment.jpg From brian at freeswitch.org Wed Jan 14 19:49:32 2015 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jan 2015 10:49:32 -0600 Subject: [Freeswitch-users] Security Issue In-Reply-To: References: Message-ID: Unless you've changed the configs its not setup to do that out of the box. On Wed, Jan 14, 2015 at 10:39 AM, Ahmed Habiba wrote: > *Dear Brian,* > > *Thank you really for your feedback, what you mentioned is correct, > however I understand that any call to comes from external system to public > context to processed , the external system IP shall be allowed via ACL, > accordingly we shouldn't expect any traffic to processed from a **server/IP > that is not defined in the ACL, am I correct?* > > *Thanks,* > > *Ahmed Habiba.* > *From: *Brian West > *To: *FreeSWITCH Users Help > *Date: *January 14, 2015 at 7:29:28 PM GMT+3 > *Reply-To: *FreeSWITCH Users Help > *Subject: **Re: [Freeswitch-users] Security Issue* > > > You have some learning to do about VoIP and various security models. See > comments inline. > > On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba wrote: > >> Dears, >> >> Kindly I noticed a very strange behaviour on Freeswitch that may allow >> non authorised users to make call through the system below is the log and >> my notice highlighted, you help will be appreciated. >> >> 1-Below is a request coming from not authored IP. >> 2-However the originating IP is ?142.54.179.218? the from is as below as >> if it is from the same server: >> > > The IP was 142.54.179.218... thats the real IP we received the request > from. > > >> >> freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at >> 16:41:34.211099: >> >> ------------------------------------------------------------------------ >> INVITE sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 >> To: 9009972599796504 >> From: 1000;tag=e8473b10 >> Via: SIP/2.0/UDP 142.54.179.218:5070 >> ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport >> Call-ID: d6e1ddab827448435f49ecaf6e613e2e >> CSeq: 1 INVITE >> Contact: >> Max-Forwards: 70 >> Allow: INVITE, ACK, CANCEL, BYE >> User-Agent: sipcli/v1.8 >> Content-Type: application/sdp >> Content-Length: 285 >> >> v=0 >> o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 >> s=sipcli >> c=IN IP4 142.54.179.218 >> t=0 0 >> m=audio 5072 RTP/AVP 18 0 8 101 >> a=fmtp:101 0-15 >> a=rtpmap:18 G729/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=ptime:20 >> a=sendrecv >> >> 3-Accordingly Freeswitch start to deal with the call normally >> > > As it should, Its hitting the external profile which doesn't have > authentication on it. > > >> >> >> >> ------------------------------------------------------------------------ >> send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 142.54.179.218:5070 >> ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 >> From: 1000;tag=e8473b10 >> To: 9009972599796504 >> Call-ID: d6e1ddab827448435f49ecaf6e613e2e >> CSeq: 1 INVITE >> Content-Length: 0 >> >> ???????????????????????????????????? >> >> 4-as we can see below Freeswitch consider the call coming from my server >> IP not from the remote IP(My server IP = 177.31.245.177) >> > > No thats false, This is just a channel named thats formed from various > bits of data, Its meaningless data and can be set, overridden or changed. > This has no bearing on what freeswitch thinks. Most likely its using the > host element from the From header. > > >> >> 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel >> sofia/external/1000 at 177.31.245.177 [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal >> sofia/external/1000 at 177.31.245.177 [BREAK] >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal >> sofia/external/1000 at 177.31.245.177 [BREAK] >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >> sofia/external/1000 at 177.31.245.177) Running State Change CS_NEW >> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 >> sofia/external/1000 at 177.31.245.177 receiving invite from >> 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z >> 64bit >> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel >> sofia/external/1000 at 177.31.245.177 entering state [received][100] >> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: >> v=0 >> o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 >> s=sipcli >> c=IN IP4 142.54.179.218 >> t=0 0 >> m=audio 5072 RTP/AVP 18 0 8 101 >> a=rtpmap:18 G729/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 ( >> sofia/external/1000 at 177.31.245.177) State Change CS_NEW -> CS_INIT >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal >> sofia/external/1000 at 177.31.245.177 [BREAK] >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 ( >> sofia/external/1000 at 177.31.245.177) State NEW >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >> sofia/external/1000 at 177.31.245.177) Running State Change CS_INIT >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( >> sofia/external/1000 at 177.31.245.177) State INIT >> 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 >> sofia/external/1000 at 177.31.245.177 SOFIA INIT >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 >> sofia/external/1000 at 177.31.245.177 Standard INIT >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 ( >> sofia/external/1000 at 177.31.245.177) State Change CS_INIT -> CS_ROUTING >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal >> sofia/external/1000 at 177.31.245.177 [BREAK] >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( >> sofia/external/1000 at 177.31.245.177) State INIT going to sleep >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >> sofia/external/1000 at 177.31.245.177) Running State Change CS_ROUTING >> 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 ( >> sofia/external/1000 at 177.31.245.177) Callstate Change DOWN -> RINGING >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 ( >> sofia/external/1000 at 177.31.245.177) State ROUTING >> 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 >> sofia/external/1000 at 177.31.245.177 SOFIA ROUTING >> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 >> sofia/external/1000 at 177.31.245.177 Standard ROUTING >> > > This is someone from outside calling in to your system, the public context > is a sandbox to allow you to isolate and route non-authenticated traffic to > your internal contexts via the transfer app, per the vanilla config > examples. This is also whey you need to understand how to secure > FreeSWITCH. > > > >> 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing 1000 >> <1000>->9009972599796504 in context public >> >> >> >> _________________________________________________________________________ >> 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 > -- *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/20150114/e408ba6a/attachment-0001.html From david.villasmil at gmail.com Wed Jan 14 19:49:37 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 14 Jan 2015 17:49:37 +0100 Subject: [Freeswitch-users] robo caller In-Reply-To: References: Message-ID: I tried using that, couldn't do what I needed at the time. Probably ok for what Sean needs, though. In any case, you're in for a great time learning about freeSWITCH! On Jan 14, 2015 5:38 PM, "Moishe Grunstein" wrote: > You can also have a look at http://www.newfies-dialer.org/ > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil Govea > *Sent:* Wednesday, January 14, 2015 11:31 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] robo caller > > > > That's fairly simple to implement. Look into at mod_amd, orignate and ivr. > > On Jan 14, 2015 5:28 PM, "Sean Devoy" wrote: > > Does anyone have a sample RoboCaller script? Perhaps I am using the > wrong name and that is why I can?t find one. I have a doctor?s office that > wants to automate the reminder calls about appointments to their patients. > > > > I am curious how people handle answering machine detection as well? > > > > Thanks, > > Sean > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/2318b529/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/2318b529/attachment.jpg From victor.chukalovskiy at gmail.com Wed Jan 14 19:51:20 2015 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 14 Jan 2015 11:51:20 -0500 Subject: [Freeswitch-users] Mod LCR SQL sub-query for better routing In-Reply-To: References: <52AA1192.6020005@gmail.com>, <52AB750E.10405@freeswitch.org>, , Message-ID: <54B69E88.3050304@gmail.com> Alright, I'm glad it helped someone :) On 14-11-28 08:05 PM, DP . wrote: > Victor, I take back my initial response on this old email. > > We ran into a couple of cases with some carriers and multiple matching > prefixes. After actually trying your sub query (modified for Mysql), > it actually does return the true lowest rate from a carrier while > importantly respecting the longest match per carrier. Unlike the > reorder_by_rate function that does not respect the longest match per > carrier. That function simply returns the lowest rate, period. > > So thanks! > > ------------------------------------------------------------------------ > From: hi-tecc at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: RE: [Freeswitch-users] Mod LCR SQL sub-query for better routing > Date: Mon, 16 Dec 2013 18:36:06 -0500 > > I agree. This definitely sounds like he simply needed the > "reorder_by_rate" param. It will reorder the initial sql results > strictly by rate: > > * reorder_by_rate - Forces the LCR module to re-order the query > strictly on rate basis. By default this is turned off, but > enabling this will always prefer rate over anything else. > > > Beware this may have an adverse effect! I initially had this turned on > then quickly realized it would sometimes try to route ALL calls by the > lowest rate found. > > Ex: flowroute lists all calls for the US with a default NPA of "1" at > .0098. Now a user trying to call Jamaica with "1876" at a rate of 0.19 > (or whatever) will get both flowroute rates returned. The reorder by > rate will assume 0.0098 is a valid rate since it will now be the > "cheapest" in the list and send the call along its way to flowroute, > whom will now bill you at 0.19. Now if you have another carrier in > your list with 1876 at 0.15 you can see why this would be a problem. > > In this case you will always want the longest matched NPANXX rate. > > ------------------------------------------------------------------------ > Date: Fri, 13 Dec 2013 15:58:54 -0500 > From: intralanman at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Mod LCR SQL sub-query for better routing > > On 12/12/2013 02:42 PM, Victor Chukalovskiy wrote: > > Hello, > > For those interested, I added a piece to mod lcr wiki. It makes sorting > / routing logic better than default logic: > > http://wiki.freeswitch.org/wiki/Mod_lcr#Custom_SQL_with_sub-query_-_for_real-life_ratesheet_complexities > > Why it helps: > > Rates rates can often be given both on per-NPA or per-NPANXX level > depending on the carrier and on the NPA. Moreover, some carriers may > have NPANXX rate lower than the corresponding NPA rate, while others > will have it inverse. Neither simple ORDER BY rate, prefix; nor ORDER BY > prefix, rate; give the truly cheapest route. The LCR logic should be > two-step process to accommodate this. > > Cheers, > -Victor > > Unless I misunderstand what you're saying, this is what the > reorder_by_rate param does. > > You'll always want to pick the longest digit match per carrier. Then > you probably want to grab the cheapest overall rate of the matches you > got back. > > -Ray > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The > CudaTel Communication Server http://www.cudatel.com Official > FreeSWITCH Sites http://www.freeswitch.org http://wiki.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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/393c7d49/attachment-0001.html From olegstolyar at gmail.com Wed Jan 14 20:05:34 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 14 Jan 2015 09:05:34 -0800 Subject: [Freeswitch-users] Detecting originate failures from dialplan Message-ID: Hi guys, I have a line like this in my dialplan: wrote: > Hi guys, > > I have a line like this in my dialplan: > > data="api_result=${originate(sofia/external/@2.3.4.5:5060 > XML > > How can I know if the originate failed? For example the number is > unreacheable or invalid or the trunk is down. In this case the > will not be executed. > > It seems that my dialplan continues to execute even if originate failed > but is there something I can do to check for that? > > _________________________________________________________________________ > 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 > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/4ee14514/attachment.html From david.villasmil at gmail.com Wed Jan 14 20:16:38 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 14 Jan 2015 18:16:38 +0100 Subject: [Freeswitch-users] Detecting originate failures from dialplan In-Reply-To: References: Message-ID: Take a look at "continue_on_fail":Hangup Causesbridge_hangup_cause This is set to the hangup cause of the last bridged B leg of the call. If you have continue_on_fail=true and hangup_after_bridge=false you can do checks on this to see what "really" happened to the call. You can for instance do execute_extension after bridge, do a condition check on ${bridge_hangup_cause} to see if it contains MEDIA_TIMEOUT and then trigger a redial of the call or transfer to a cell phone. For a list of hangup causes, see Hangup Causes . On Wed, Jan 14, 2015 at 6:16 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > take a look at "continue_on_fail": > > > On Wed, Jan 14, 2015 at 6:05 PM, Oleg Stolyar > wrote: > >> Hi guys, >> >> I have a line like this in my dialplan: >> >> > data="api_result=${originate(sofia/external/@2.3.4.5:5060 >> XML >> >> How can I know if the originate failed? For example the number is >> unreacheable or invalid or the trunk is down. In this case the >> will not be executed. >> >> It seems that my dialplan continues to execute even if originate failed >> but is there something I can do to check for that? >> >> _________________________________________________________________________ >> 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 >> > > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/1c82b1a5/attachment.html From ahabiba at gmail.com Wed Jan 14 20:21:29 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Wed, 14 Jan 2015 20:21:29 +0300 Subject: [Freeswitch-users] Security Issue In-Reply-To: References: Message-ID: Thanks Brian, What I got from the below link that unless having a record like the below in acl.conf.xml, Digest Authentication(i.e. username/password) will be required, please correct me. https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes From: Brian West > To: FreeSWITCH Users Help > Date: January 14, 2015 at 7:49:32 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue Unless you've changed the configs its not setup to do that out of the box. On Wed, Jan 14, 2015 at 10:39 AM, Ahmed Habiba > wrote: Dear Brian, Thank you really for your feedback, what you mentioned is correct, however I understand that any call to comes from external system to public context to processed , the external system IP shall be allowed via ACL, accordingly we shouldn't expect any traffic to processed from a server/IP that is not defined in the ACL, am I correct? Thanks, Ahmed Habiba. From: Brian West > To: FreeSWITCH Users Help > Date: January 14, 2015 at 7:29:28 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue You have some learning to do about VoIP and various security models. See comments inline. On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba > wrote: Dears, Kindly I noticed a very strange behaviour on Freeswitch that may allow non authorised users to make call through the system below is the log and my notice highlighted, you help will be appreciated. 1-Below is a request coming from not authored IP. 2-However the originating IP is ?142.54.179.218? the from is as below as if it is from the same server: The IP was 142.54.179.218... thats the real IP we received the request from. freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at 16:41:34.211099: ------------------------------------------------------------------------ INVITE <> <>sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 To: 9009972599796504< <> <>sip:9009972599796504 at 177.31.245.177 > From: 1000< <> <>sip:1000 at 177.31.245.177 >;tag=e8473b10 Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Contact: < <> <>sip:1000 at 142.54.179.218:5070 > Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE User-Agent: sipcli/v1.8 Content-Type: application/sdp Content-Length: 285 v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv 3-Accordingly Freeswitch start to deal with the call normally As it should, Its hitting the external profile which doesn't have authentication on it. ------------------------------------------------------------------------ send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 From: 1000< <> <>sip:1000 at 177.31.245.177 >;tag=e8473b10 To: 9009972599796504< <> <>sip:9009972599796504 at 177.31.245.177 > Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Content-Length: 0 ???????????????????????????????????? 4-as we can see below Freeswitch consider the call coming from my server IP not from the remote IP(My server IP = 177.31.245.177) No thats false, This is just a channel named thats formed from various bits of data, Its meaningless data and can be set, overridden or changed. This has no bearing on what freeswitch thinks. Most likely its using the host element from the From header. 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1000 at 177.31.245.177 [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_NEW 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 sofia/external/1000 at 177.31.245.177 receiving invite from 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z 64bit 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel sofia/external/1000 at 177.31.245.177 entering state [received][100] 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 (sofia/external/1000 at 177.31.245.177 ) State Change CS_NEW -> CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1000 at 177.31.245.177 ) State NEW 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 sofia/external/1000 at 177.31.245.177 SOFIA INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 sofia/external/1000 at 177.31.245.177 Standard INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 (sofia/external/1000 at 177.31.245.177 ) State Change CS_INIT -> CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT going to sleep 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 (sofia/external/1000 at 177.31.245.177 ) Callstate Change DOWN -> RINGING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 (sofia/external/1000 at 177.31.245.177 ) State ROUTING 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 sofia/external/1000 at 177.31.245.177 SOFIA ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 sofia/external/1000 at 177.31.245.177 Standard ROUTING This is someone from outside calling in to your system, the public context is a sandbox to allow you to isolate and route non-authenticated traffic to your internal contexts via the transfer app, per the vanilla config examples. This is also whey you need to understand how to secure FreeSWITCH. 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->9009972599796504 in context public _________________________________________________________________________ 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/20150114/da806978/attachment-0001.html From ahabiba at gmail.com Wed Jan 14 20:21:29 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Wed, 14 Jan 2015 20:21:29 +0300 Subject: [Freeswitch-users] Security Issue In-Reply-To: References: Message-ID: Thanks Brian, What I got from the below link that unless having a record like the below in acl.conf.xml, Digest Authentication(i.e. username/password) will be required, please correct me. https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes From: Brian West > To: FreeSWITCH Users Help > Date: January 14, 2015 at 7:49:32 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue Unless you've changed the configs its not setup to do that out of the box. On Wed, Jan 14, 2015 at 10:39 AM, Ahmed Habiba > wrote: Dear Brian, Thank you really for your feedback, what you mentioned is correct, however I understand that any call to comes from external system to public context to processed , the external system IP shall be allowed via ACL, accordingly we shouldn't expect any traffic to processed from a server/IP that is not defined in the ACL, am I correct? Thanks, Ahmed Habiba. From: Brian West > To: FreeSWITCH Users Help > Date: January 14, 2015 at 7:29:28 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue You have some learning to do about VoIP and various security models. See comments inline. On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba > wrote: Dears, Kindly I noticed a very strange behaviour on Freeswitch that may allow non authorised users to make call through the system below is the log and my notice highlighted, you help will be appreciated. 1-Below is a request coming from not authored IP. 2-However the originating IP is ?142.54.179.218? the from is as below as if it is from the same server: The IP was 142.54.179.218... thats the real IP we received the request from. freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at 16:41:34.211099: ------------------------------------------------------------------------ INVITE <> <>sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 To: 9009972599796504< <> <>sip:9009972599796504 at 177.31.245.177 > From: 1000< <> <>sip:1000 at 177.31.245.177 >;tag=e8473b10 Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Contact: < <> <>sip:1000 at 142.54.179.218:5070 > Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE User-Agent: sipcli/v1.8 Content-Type: application/sdp Content-Length: 285 v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv 3-Accordingly Freeswitch start to deal with the call normally As it should, Its hitting the external profile which doesn't have authentication on it. ------------------------------------------------------------------------ send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 From: 1000< <> <>sip:1000 at 177.31.245.177 >;tag=e8473b10 To: 9009972599796504< <> <>sip:9009972599796504 at 177.31.245.177 > Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Content-Length: 0 ???????????????????????????????????? 4-as we can see below Freeswitch consider the call coming from my server IP not from the remote IP(My server IP = 177.31.245.177) No thats false, This is just a channel named thats formed from various bits of data, Its meaningless data and can be set, overridden or changed. This has no bearing on what freeswitch thinks. Most likely its using the host element from the From header. 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1000 at 177.31.245.177 [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_NEW 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 sofia/external/1000 at 177.31.245.177 receiving invite from 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z 64bit 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel sofia/external/1000 at 177.31.245.177 entering state [received][100] 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 (sofia/external/1000 at 177.31.245.177 ) State Change CS_NEW -> CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1000 at 177.31.245.177 ) State NEW 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 sofia/external/1000 at 177.31.245.177 SOFIA INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 sofia/external/1000 at 177.31.245.177 Standard INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 (sofia/external/1000 at 177.31.245.177 ) State Change CS_INIT -> CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT going to sleep 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 (sofia/external/1000 at 177.31.245.177 ) Callstate Change DOWN -> RINGING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 (sofia/external/1000 at 177.31.245.177 ) State ROUTING 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 sofia/external/1000 at 177.31.245.177 SOFIA ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 sofia/external/1000 at 177.31.245.177 Standard ROUTING This is someone from outside calling in to your system, the public context is a sandbox to allow you to isolate and route non-authenticated traffic to your internal contexts via the transfer app, per the vanilla config examples. This is also whey you need to understand how to secure FreeSWITCH. 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->9009972599796504 in context public _________________________________________________________________________ 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/20150114/da806978/attachment-0003.html From david.villasmil at gmail.com Wed Jan 14 20:30:35 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 14 Jan 2015 18:30:35 +0100 Subject: [Freeswitch-users] Security Issue In-Reply-To: References: Message-ID: Authorization is done if you configure your sip profile to do it. By default 5060 (internal) requires authentication, 5080 (external) doesn't but it does use the ACL to allow or not calls. On Jan 14, 2015 6:22 PM, "Ahmed Habiba" wrote: > > Thanks Brian, > > What I got from the below link that unless having a record like the below > in acl.conf.xml, Digest Authentication(i.e. username/password) will be > required, please correct me. > > https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes > > > > > > *From: *Brian West > *To: *FreeSWITCH Users Help > *Date: *January 14, 2015 at 7:49:32 PM GMT+3 > *Reply-To: *FreeSWITCH Users Help > *Subject: **Re: [Freeswitch-users] Security Issue* > > > Unless you've changed the configs its not setup to do that out of the box. > > On Wed, Jan 14, 2015 at 10:39 AM, Ahmed Habiba wrote: > >> *Dear Brian,* >> >> *Thank you really for your feedback, what you mentioned is correct, >> however I understand that any call to comes from external system to public >> context to processed , the external system IP shall be allowed via ACL, >> accordingly we shouldn't expect any traffic to processed from a **server/IP >> that is not defined in the ACL, am I correct?* >> >> *Thanks,* >> >> *Ahmed Habiba.* >> *From: *Brian West >> *To: *FreeSWITCH Users Help >> *Date: *January 14, 2015 at 7:29:28 PM GMT+3 >> *Reply-To: *FreeSWITCH Users Help >> *Subject: **Re: [Freeswitch-users] Security Issue* >> >> >> You have some learning to do about VoIP and various security models. See >> comments inline. >> >> On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba wrote: >> >>> Dears, >>> >>> Kindly I noticed a very strange behaviour on Freeswitch that may allow >>> non authorised users to make call through the system below is the log and >>> my notice highlighted, you help will be appreciated. >>> >>> 1-Below is a request coming from not authored IP. >>> 2-However the originating IP is ?142.54.179.218? the from is as below >>> as if it is from the same server: >>> >> >> The IP was 142.54.179.218... thats the real IP we received the request >> from. >> >> >>> >>> freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at >>> 16:41:34.211099: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 >>> To: 9009972599796504 >>> From: 1000;tag=e8473b10 >>> Via: SIP/2.0/UDP 142.54.179.218:5070 >>> ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport >>> Call-ID: d6e1ddab827448435f49ecaf6e613e2e >>> CSeq: 1 INVITE >>> Contact: >>> Max-Forwards: 70 >>> Allow: INVITE, ACK, CANCEL, BYE >>> User-Agent: sipcli/v1.8 >>> Content-Type: application/sdp >>> Content-Length: 285 >>> >>> v=0 >>> o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 >>> s=sipcli >>> c=IN IP4 142.54.179.218 >>> t=0 0 >>> m=audio 5072 RTP/AVP 18 0 8 101 >>> a=fmtp:101 0-15 >>> a=rtpmap:18 G729/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=ptime:20 >>> a=sendrecv >>> >>> 3-Accordingly Freeswitch start to deal with the call normally >>> >> >> As it should, Its hitting the external profile which doesn't have >> authentication on it. >> >> >>> >>> >>> >>> ------------------------------------------------------------------------ >>> send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 142.54.179.218:5070 >>> ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 >>> From: 1000;tag=e8473b10 >>> To: 9009972599796504 >>> Call-ID: d6e1ddab827448435f49ecaf6e613e2e >>> CSeq: 1 INVITE >>> Content-Length: 0 >>> >>> ???????????????????????????????????? >>> >>> 4-as we can see below Freeswitch consider the call coming from my server >>> IP not from the remote IP(My server IP = 177.31.245.177) >>> >> >> No thats false, This is just a channel named thats formed from various >> bits of data, Its meaningless data and can be set, overridden or changed. >> This has no bearing on what freeswitch thinks. Most likely its using the >> host element from the From header. >> >> >>> >>> 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel >>> sofia/external/1000 at 177.31.245.177 >>> [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send >>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send >>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >>> sofia/external/1000 at 177.31.245.177) Running State Change CS_NEW >>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 >>> sofia/external/1000 at 177.31.245.177 receiving invite from >>> 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z >>> 64bit >>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel >>> sofia/external/1000 at 177.31.245.177 entering state [received][100] >>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: >>> v=0 >>> o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 >>> s=sipcli >>> c=IN IP4 142.54.179.218 >>> t=0 0 >>> m=audio 5072 RTP/AVP 18 0 8 101 >>> a=rtpmap:18 G729/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=ptime:20 >>> >>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 ( >>> sofia/external/1000 at 177.31.245.177) State Change CS_NEW -> CS_INIT >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send >>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 ( >>> sofia/external/1000 at 177.31.245.177) State NEW >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >>> sofia/external/1000 at 177.31.245.177) Running State Change CS_INIT >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( >>> sofia/external/1000 at 177.31.245.177) State INIT >>> 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 >>> sofia/external/1000 at 177.31.245.177 SOFIA INIT >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 >>> sofia/external/1000 at 177.31.245.177 Standard INIT >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 ( >>> sofia/external/1000 at 177.31.245.177) State Change CS_INIT -> CS_ROUTING >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send >>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( >>> sofia/external/1000 at 177.31.245.177) State INIT going to sleep >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >>> sofia/external/1000 at 177.31.245.177) Running State Change CS_ROUTING >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 ( >>> sofia/external/1000 at 177.31.245.177) Callstate Change DOWN -> RINGING >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 ( >>> sofia/external/1000 at 177.31.245.177) State ROUTING >>> 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 >>> sofia/external/1000 at 177.31.245.177 SOFIA ROUTING >>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 >>> sofia/external/1000 at 177.31.245.177 Standard ROUTING >>> >> >> This is someone from outside calling in to your system, the public >> context is a sandbox to allow you to isolate and route non-authenticated >> traffic to your internal contexts via the transfer app, per the vanilla >> config examples. This is also whey you need to understand how to secure >> FreeSWITCH. >> >> >> >>> 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing 1000 >>> <1000>->9009972599796504 in context public >>> >>> >>> >>> _________________________________________________________________________ >>> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/6f4f6b9a/attachment-0001.html From krice at freeswitch.org Wed Jan 14 20:33:35 2015 From: krice at freeswitch.org (Ken Rice) Date: Wed, 14 Jan 2015 17:33:35 +0000 Subject: [Freeswitch-users] FreeSWITCH Launches New Platform on 10th Anniversary Message-ID: <54b6a86fd963d_3e7b05338740d@ip-10-169-78-55.mail> New Post on freeswitch.org from anthm check it out at http://ift.tt/1IKASRe FreeSWITCH Launches New Platform on 10th Anniversary The first line of code in FreeSWITCH was drafted Spring of 2005, nearly a decade ago. 2015 marks the 10-year anniversary of the core code and the 9-year anniversary of the FreeSWITCH Public Community. Over the years, the core developers of the project have worked hard to provide top notch community support. FreeSWITCH Solutions, the consulting firm owned and operated by the core FreeSWITCH team, have been offering commercial support for many years for companies who take things to the next level and use FreeSWITCH to power their products. Today we are proud to announce the unveiling of FreeSWITCH.com and the FreeSWITCH Advantage? platform and support. Built on top of Debian Linux, FreeSWITCH Advantage takes our proven support services and couples it with a reliable operating system distribution and the result is a uniform packaged system tuned specifically for running FreeSWITCH. New releases will be simple to apply and any changes to the other packages on the system that have an impact on FreeSWITCH will be applied and managed automatically. Customers will have the best environment possible in place to run their systems as well as the support they need to ensure they stay up and running. The FreeSWITCH Advantage provides critical early-access to releases to make sure any problems encountered are addressed ASAP. All of the code used with FreeSWITCH Advantage is taken straight from the project?s public git repository and packaged with testing and stability in mind so the community continues to benefit as a result. FreeSWITCH Solutions also continues to offer the most comprehensive consulting services available with the experience and knowledge to create almost anything imaginable that pertains to interfacing with FreeSWITCH. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/c84cbaaf/attachment.html From ahabiba at gmail.com Wed Jan 14 20:39:26 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Wed, 14 Jan 2015 20:39:26 +0300 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 103, Issue 177 In-Reply-To: References: Message-ID: Thank you really David, Here is my point, the sip-trace in the first mail shows that, the call comes to public context mainly through port 5080, and however the originator IP was not defined in my ACL list Freeswitch continue to process the call for some reason. even if it come to 5060, I was expecting some request for digest authentication, which is not shown in the log. From: David Villasmil Govea > To: FreeSWITCH Users Help > Date: January 14, 2015 at 8:30:35 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue Authorization is done if you configure your sip profile to do it. By default 5060 (internal) requires authentication, 5080 (external) doesn't but it does use the ACL to allow or not calls. On Jan 14, 2015 6:22 PM, "Ahmed Habiba" > wrote: Thanks Brian, What I got from the below link that unless having a record like the below in acl.conf.xml, Digest Authentication(i.e. username/password) will be required, please correct me. https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes From: Brian West > To: FreeSWITCH Users Help > Date: January 14, 2015 at 7:49:32 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue Unless you've changed the configs its not setup to do that out of the box. On Wed, Jan 14, 2015 at 10:39 AM, Ahmed Habiba > wrote: Dear Brian, Thank you really for your feedback, what you mentioned is correct, however I understand that any call to comes from external system to public context to processed , the external system IP shall be allowed via ACL, accordingly we shouldn't expect any traffic to processed from a server/IP that is not defined in the ACL, am I correct? Thanks, Ahmed Habiba. From: Brian West > To: FreeSWITCH Users Help > Date: January 14, 2015 at 7:29:28 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue You have some learning to do about VoIP and various security models. See comments inline. On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba > wrote: Dears, Kindly I noticed a very strange behaviour on Freeswitch that may allow non authorised users to make call through the system below is the log and my notice highlighted, you help will be appreciated. 1-Below is a request coming from not authored IP. 2-However the originating IP is ?142.54.179.218? the from is as below as if it is from the same server: The IP was 142.54.179.218... thats the real IP we received the request from. freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at 16:41:34.211099: ------------------------------------------------------------------------ INVITE <> <> <>sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 To: 9009972599796504< <> <> <>sip:9009972599796504 at 177.31.245.177 > From: 1000< <> <> <>sip:1000 at 177.31.245.177 >;tag=e8473b10 Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Contact: < <> <> <>sip:1000 at 142.54.179.218:5070 > Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE User-Agent: sipcli/v1.8 Content-Type: application/sdp Content-Length: 285 v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv 3-Accordingly Freeswitch start to deal with the call normally As it should, Its hitting the external profile which doesn't have authentication on it. ------------------------------------------------------------------------ send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 From: 1000< <> <> <>sip:1000 at 177.31.245.177 >;tag=e8473b10 To: 9009972599796504< <> <> <>sip:9009972599796504 at 177.31.245.177 > Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Content-Length: 0 ???????????????????????????????????? 4-as we can see below Freeswitch consider the call coming from my server IP not from the remote IP(My server IP = 177.31.245.177) No thats false, This is just a channel named thats formed from various bits of data, Its meaningless data and can be set, overridden or changed. This has no bearing on what freeswitch thinks. Most likely its using the host element from the From header. 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1000 at 177.31.245.177 [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_NEW 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 sofia/external/1000 at 177.31.245.177 receiving invite from 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z 64bit 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel sofia/external/1000 at 177.31.245.177 entering state [received][100] 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 (sofia/external/1000 at 177.31.245.177 ) State Change CS_NEW -> CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1000 at 177.31.245.177 ) State NEW 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 sofia/external/1000 at 177.31.245.177 SOFIA INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 sofia/external/1000 at 177.31.245.177 Standard INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 (sofia/external/1000 at 177.31.245.177 ) State Change CS_INIT -> CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT going to sleep 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 (sofia/external/1000 at 177.31.245.177 ) Callstate Change DOWN -> RINGING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 (sofia/external/1000 at 177.31.245.177 ) State ROUTING 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 sofia/external/1000 at 177.31.245.177 SOFIA ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 sofia/external/1000 at 177.31.245.177 Standard ROUTING This is someone from outside calling in to your system, the public context is a sandbox to allow you to isolate and route non-authenticated traffic to your internal contexts via the transfer app, per the vanilla config examples. This is also whey you need to understand how to secure FreeSWITCH. 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->9009972599796504 in context public _________________________________________________________________________ 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/20150114/f4f669d3/attachment-0001.html From ahabiba at gmail.com Wed Jan 14 20:40:09 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Wed, 14 Jan 2015 20:40:09 +0300 Subject: [Freeswitch-users] Security Issue In-Reply-To: References: Message-ID: <4BFC98A1-CF17-4ACF-9E4C-EF8D6B447FA5@gmail.com> Thank you really David, Here is my point, the sip-trace in the first mail shows that, the call comes to public context mainly through port 5080, and however the originator IP was not defined in my ACL list Freeswitch continue to process the call for some reason. even if it come to 5060, I was expecting some request for digest authentication, which is not shown in the log. From: David Villasmil Govea > To: FreeSWITCH Users Help > Date: January 14, 2015 at 8:30:35 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue Authorization is done if you configure your sip profile to do it. By default 5060 (internal) requires authentication, 5080 (external) doesn't but it does use the ACL to allow or not calls. On Jan 14, 2015 6:22 PM, "Ahmed Habiba" > wrote: Thanks Brian, What I got from the below link that unless having a record like the below in acl.conf.xml, Digest Authentication(i.e. username/password) will be required, please correct me. https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes From: Brian West > To: FreeSWITCH Users Help > Date: January 14, 2015 at 7:49:32 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue Unless you've changed the configs its not setup to do that out of the box. On Wed, Jan 14, 2015 at 10:39 AM, Ahmed Habiba > wrote: Dear Brian, Thank you really for your feedback, what you mentioned is correct, however I understand that any call to comes from external system to public context to processed , the external system IP shall be allowed via ACL, accordingly we shouldn't expect any traffic to processed from a server/IP that is not defined in the ACL, am I correct? Thanks, Ahmed Habiba. From: Brian West > To: FreeSWITCH Users Help > Date: January 14, 2015 at 7:29:28 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue You have some learning to do about VoIP and various security models. See comments inline. On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba > wrote: Dears, Kindly I noticed a very strange behaviour on Freeswitch that may allow non authorised users to make call through the system below is the log and my notice highlighted, you help will be appreciated. 1-Below is a request coming from not authored IP. 2-However the originating IP is ?142.54.179.218? the from is as below as if it is from the same server: The IP was 142.54.179.218... thats the real IP we received the request from. freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at 16:41:34.211099: ------------------------------------------------------------------------ INVITE <> <> <>sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 To: 9009972599796504< <> <> <>sip:9009972599796504 at 177.31.245.177 > From: 1000< <> <> <>sip:1000 at 177.31.245.177 >;tag=e8473b10 Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Contact: < <> <> <>sip:1000 at 142.54.179.218:5070 > Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE User-Agent: sipcli/v1.8 Content-Type: application/sdp Content-Length: 285 v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv 3-Accordingly Freeswitch start to deal with the call normally As it should, Its hitting the external profile which doesn't have authentication on it. ------------------------------------------------------------------------ send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 142.54.179.218:5070;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 From: 1000< <> <> <>sip:1000 at 177.31.245.177 >;tag=e8473b10 To: 9009972599796504< <> <> <>sip:9009972599796504 at 177.31.245.177 > Call-ID: d6e1ddab827448435f49ecaf6e613e2e CSeq: 1 INVITE Content-Length: 0 ???????????????????????????????????? 4-as we can see below Freeswitch consider the call coming from my server IP not from the remote IP(My server IP = 177.31.245.177) No thats false, This is just a channel named thats formed from various bits of data, Its meaningless data and can be set, overridden or changed. This has no bearing on what freeswitch thinks. Most likely its using the host element from the From header. 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1000 at 177.31.245.177 [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_NEW 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 sofia/external/1000 at 177.31.245.177 receiving invite from 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z 64bit 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel sofia/external/1000 at 177.31.245.177 entering state [received][100] 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: v=0 o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 s=sipcli c=IN IP4 142.54.179.218 t=0 0 m=audio 5072 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 (sofia/external/1000 at 177.31.245.177 ) State Change CS_NEW -> CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1000 at 177.31.245.177 ) State NEW 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 sofia/external/1000 at 177.31.245.177 SOFIA INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 sofia/external/1000 at 177.31.245.177 Standard INIT 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 (sofia/external/1000 at 177.31.245.177 ) State Change CS_INIT -> CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1000 at 177.31.245.177 [BREAK] 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 (sofia/external/1000 at 177.31.245.177 ) State INIT going to sleep 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1000 at 177.31.245.177 ) Running State Change CS_ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 (sofia/external/1000 at 177.31.245.177 ) Callstate Change DOWN -> RINGING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 (sofia/external/1000 at 177.31.245.177 ) State ROUTING 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 sofia/external/1000 at 177.31.245.177 SOFIA ROUTING 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 sofia/external/1000 at 177.31.245.177 Standard ROUTING This is someone from outside calling in to your system, the public context is a sandbox to allow you to isolate and route non-authenticated traffic to your internal contexts via the transfer app, per the vanilla config examples. This is also whey you need to understand how to secure FreeSWITCH. 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->9009972599796504 in context public _________________________________________________________________________ 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/20150114/79c8648a/attachment-0001.html From brian at freeswitch.org Wed Jan 14 20:43:41 2015 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jan 2015 11:43:41 -0600 Subject: [Freeswitch-users] Security Issue In-Reply-To: References: Message-ID: ACLs are NOT used in the external profile at all. On the external profile will help. On Wed, Jan 14, 2015 at 11:30 AM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Authorization is done if you configure your sip profile to do it. By > default 5060 (internal) requires authentication, 5080 (external) doesn't > but it does use the ACL to allow or not calls. > On Jan 14, 2015 6:22 PM, "Ahmed Habiba" wrote: > >> >> Thanks Brian, >> >> What I got from the below link that unless having a record like the below >> in acl.conf.xml, Digest Authentication(i.e. username/password) will be >> required, please correct me. >> >> https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes >> >> >> >> >> >> *From: *Brian West >> *To: *FreeSWITCH Users Help >> *Date: *January 14, 2015 at 7:49:32 PM GMT+3 >> *Reply-To: *FreeSWITCH Users Help >> *Subject: **Re: [Freeswitch-users] Security Issue* >> >> >> Unless you've changed the configs its not setup to do that out of the box. >> >> On Wed, Jan 14, 2015 at 10:39 AM, Ahmed Habiba wrote: >> >>> *Dear Brian,* >>> >>> *Thank you really for your feedback, what you mentioned is correct, >>> however I understand that any call to comes from external system to public >>> context to processed , the external system IP shall be allowed via ACL, >>> accordingly we shouldn't expect any traffic to processed from a **server/IP >>> that is not defined in the ACL, am I correct?* >>> >>> *Thanks,* >>> >>> *Ahmed Habiba.* >>> *From: *Brian West >>> *To: *FreeSWITCH Users Help >>> *Date: *January 14, 2015 at 7:29:28 PM GMT+3 >>> *Reply-To: *FreeSWITCH Users Help >> > >>> *Subject: **Re: [Freeswitch-users] Security Issue* >>> >>> >>> You have some learning to do about VoIP and various security models. >>> See comments inline. >>> >>> On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba >>> wrote: >>> >>>> Dears, >>>> >>>> Kindly I noticed a very strange behaviour on Freeswitch that may allow >>>> non authorised users to make call through the system below is the log and >>>> my notice highlighted, you help will be appreciated. >>>> >>>> 1-Below is a request coming from not authored IP. >>>> 2-However the originating IP is ?142.54.179.218? the from is as below >>>> as if it is from the same server: >>>> >>> >>> The IP was 142.54.179.218... thats the real IP we received the request >>> from. >>> >>> >>>> >>>> freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at >>>> 16:41:34.211099: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 >>>> To: 9009972599796504 >>>> From: 1000;tag=e8473b10 >>>> Via: SIP/2.0/UDP 142.54.179.218:5070 >>>> ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport >>>> Call-ID: d6e1ddab827448435f49ecaf6e613e2e >>>> CSeq: 1 INVITE >>>> Contact: >>>> Max-Forwards: 70 >>>> Allow: INVITE, ACK, CANCEL, BYE >>>> User-Agent: sipcli/v1.8 >>>> Content-Type: application/sdp >>>> Content-Length: 285 >>>> >>>> v=0 >>>> o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 >>>> s=sipcli >>>> c=IN IP4 142.54.179.218 >>>> t=0 0 >>>> m=audio 5072 RTP/AVP 18 0 8 101 >>>> a=fmtp:101 0-15 >>>> a=rtpmap:18 G729/8000 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=ptime:20 >>>> a=sendrecv >>>> >>>> 3-Accordingly Freeswitch start to deal with the call normally >>>> >>> >>> As it should, Its hitting the external profile which doesn't have >>> authentication on it. >>> >>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 100 Trying >>>> Via: SIP/2.0/UDP 142.54.179.218:5070 >>>> ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 >>>> From: 1000;tag=e8473b10 >>>> To: 9009972599796504 >>>> Call-ID: d6e1ddab827448435f49ecaf6e613e2e >>>> CSeq: 1 INVITE >>>> Content-Length: 0 >>>> >>>> ???????????????????????????????????? >>>> >>>> 4-as we can see below Freeswitch consider the call coming from my >>>> server IP not from the remote IP(My server IP = 177.31.245.177) >>>> >>> >>> No thats false, This is just a channel named thats formed from various >>> bits of data, Its meaningless data and can be set, overridden or changed. >>> This has no bearing on what freeswitch thinks. Most likely its using the >>> host element from the From header. >>> >>> >>>> >>>> 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel >>>> sofia/external/1000 at 177.31.245.177 >>>> [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send >>>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send >>>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >>>> sofia/external/1000 at 177.31.245.177) Running State Change CS_NEW >>>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 >>>> sofia/external/1000 at 177.31.245.177 receiving invite from >>>> 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z >>>> 64bit >>>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel >>>> sofia/external/1000 at 177.31.245.177 entering state [received][100] >>>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: >>>> v=0 >>>> o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 >>>> s=sipcli >>>> c=IN IP4 142.54.179.218 >>>> t=0 0 >>>> m=audio 5072 RTP/AVP 18 0 8 101 >>>> a=rtpmap:18 G729/8000 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> a=ptime:20 >>>> >>>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 ( >>>> sofia/external/1000 at 177.31.245.177) State Change CS_NEW -> CS_INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send >>>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 ( >>>> sofia/external/1000 at 177.31.245.177) State NEW >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >>>> sofia/external/1000 at 177.31.245.177) Running State Change CS_INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( >>>> sofia/external/1000 at 177.31.245.177) State INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 >>>> sofia/external/1000 at 177.31.245.177 SOFIA INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 >>>> sofia/external/1000 at 177.31.245.177 Standard INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 ( >>>> sofia/external/1000 at 177.31.245.177) State Change CS_INIT -> CS_ROUTING >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send >>>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( >>>> sofia/external/1000 at 177.31.245.177) State INIT going to sleep >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >>>> sofia/external/1000 at 177.31.245.177) Running State Change CS_ROUTING >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 ( >>>> sofia/external/1000 at 177.31.245.177) Callstate Change DOWN -> RINGING >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 ( >>>> sofia/external/1000 at 177.31.245.177) State ROUTING >>>> 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 >>>> sofia/external/1000 at 177.31.245.177 SOFIA ROUTING >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 >>>> sofia/external/1000 at 177.31.245.177 Standard ROUTING >>>> >>> >>> This is someone from outside calling in to your system, the public >>> context is a sandbox to allow you to isolate and route non-authenticated >>> traffic to your internal contexts via the transfer app, per the vanilla >>> config examples. This is also whey you need to understand how to secure >>> FreeSWITCH. >>> >>> >>> >>>> 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing >>>> 1000 <1000>->9009972599796504 in context public >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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/20150114/428ebfe2/attachment-0001.html From david.villasmil at gmail.com Wed Jan 14 20:51:37 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Wed, 14 Jan 2015 18:51:37 +0100 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 103, Issue 177 In-Reply-To: References: Message-ID: On 5060 is not asking for authentication??? If it's not, something's wrong with your config. It's not the default config. Can you paste your sip profile both internal and external? On Jan 14, 2015 6:40 PM, "Ahmed Habiba" wrote: > Thank you really David, > > Here is my point, the sip-trace in the first mail shows that, the call > comes to public context mainly through port 5080, and however the > originator IP was not defined in my ACL list Freeswitch continue to process > the call for some reason. > > even if it come to 5060, I was expecting some request for digest > authentication, which is not shown in the log. > > *From: *David Villasmil Govea > *To: *FreeSWITCH Users Help > *Date: *January 14, 2015 at 8:30:35 PM GMT+3 > *Reply-To: *FreeSWITCH Users Help > *Subject: **Re: [Freeswitch-users] Security Issue* > > > Authorization is done if you configure your sip profile to do it. By > default 5060 (internal) requires authentication, 5080 (external) doesn't > but it does use the ACL to allow or not calls. > On Jan 14, 2015 6:22 PM, "Ahmed Habiba" wrote: > >> >> Thanks Brian, >> >> What I got from the below link that unless having a record like the below >> in acl.conf.xml, Digest Authentication(i.e. username/password) will be >> required, please correct me. >> >> https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes >> >> >> >> >> >> *From: *Brian West >> *To: *FreeSWITCH Users Help >> *Date: *January 14, 2015 at 7:49:32 PM GMT+3 >> *Reply-To: *FreeSWITCH Users Help >> *Subject: **Re: [Freeswitch-users] Security Issue* >> >> >> Unless you've changed the configs its not setup to do that out of the box. >> >> On Wed, Jan 14, 2015 at 10:39 AM, Ahmed Habiba wrote: >> >>> *Dear Brian,* >>> >>> *Thank you really for your feedback, what you mentioned is correct, >>> however I understand that any call to comes from external system to public >>> context to processed , the external system IP shall be allowed via ACL, >>> accordingly we shouldn't expect any traffic to processed from a **server/IP >>> that is not defined in the ACL, am I correct?* >>> >>> *Thanks,* >>> >>> *Ahmed Habiba.* >>> *From: *Brian West >>> *To: *FreeSWITCH Users Help >>> *Date: *January 14, 2015 at 7:29:28 PM GMT+3 >>> *Reply-To: *FreeSWITCH Users Help >> > >>> *Subject: **Re: [Freeswitch-users] Security Issue* >>> >>> >>> You have some learning to do about VoIP and various security models. >>> See comments inline. >>> >>> On Wed, Jan 14, 2015 at 10:08 AM, Ahmed Habiba >>> wrote: >>> >>>> Dears, >>>> >>>> Kindly I noticed a very strange behaviour on Freeswitch that may allow >>>> non authorised users to make call through the system below is the log and >>>> my notice highlighted, you help will be appreciated. >>>> >>>> 1-Below is a request coming from not authored IP. >>>> 2-However the originating IP is ?142.54.179.218? the from is as below >>>> as if it is from the same server: >>>> >>> >>> The IP was 142.54.179.218... thats the real IP we received the request >>> from. >>> >>> >>>> >>>> freeswitch at internal> recv 770 bytes from udp/[142.54.179.218]:5070 at >>>> 16:41:34.211099: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE sip:9009972599796504 at 177.31.245.177:5080 SIP/2.0 >>>> To: 9009972599796504 >>>> From: 1000;tag=e8473b10 >>>> Via: SIP/2.0/UDP 142.54.179.218:5070 >>>> ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport >>>> Call-ID: d6e1ddab827448435f49ecaf6e613e2e >>>> CSeq: 1 INVITE >>>> Contact: >>>> Max-Forwards: 70 >>>> Allow: INVITE, ACK, CANCEL, BYE >>>> User-Agent: sipcli/v1.8 >>>> Content-Type: application/sdp >>>> Content-Length: 285 >>>> >>>> v=0 >>>> o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 >>>> s=sipcli >>>> c=IN IP4 142.54.179.218 >>>> t=0 0 >>>> m=audio 5072 RTP/AVP 18 0 8 101 >>>> a=fmtp:101 0-15 >>>> a=rtpmap:18 G729/8000 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=ptime:20 >>>> a=sendrecv >>>> >>>> 3-Accordingly Freeswitch start to deal with the call normally >>>> >>> >>> As it should, Its hitting the external profile which doesn't have >>> authentication on it. >>> >>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> send 333 bytes to udp/[142.54.179.218]:5070 at 16:41:34.211442: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 100 Trying >>>> Via: SIP/2.0/UDP 142.54.179.218:5070 >>>> ;branch=z9hG4bK-d6e1ddab827448435f49ecaf6e613e2e;rport=5070 >>>> From: 1000;tag=e8473b10 >>>> To: 9009972599796504 >>>> Call-ID: d6e1ddab827448435f49ecaf6e613e2e >>>> CSeq: 1 INVITE >>>> Content-Length: 0 >>>> >>>> ???????????????????????????????????? >>>> >>>> 4-as we can see below Freeswitch consider the call coming from my >>>> server IP not from the remote IP(My server IP = 177.31.245.177) >>>> >>> >>> No thats false, This is just a channel named thats formed from various >>> bits of data, Its meaningless data and can be set, overridden or changed. >>> This has no bearing on what freeswitch thinks. Most likely its using the >>> host element from the From header. >>> >>> >>>> >>>> 2015-01-14 16:41:34.203196 [NOTICE] switch_channel.c:1055 New Channel >>>> sofia/external/1000 at 177.31.245.177 >>>> [d1879400-9c03-11e4-8cd6-2f1eb174d7b4] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send >>>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1053 Send >>>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >>>> sofia/external/1000 at 177.31.245.177) Running State Change CS_NEW >>>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:8812 >>>> sofia/external/1000 at 177.31.245.177 receiving invite from >>>> 142.54.179.218:5070 version: 1.4.13 git b942d0f 2014-11-03 19:53:00Z >>>> 64bit >>>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6606 Channel >>>> sofia/external/1000 at 177.31.245.177 entering state [received][100] >>>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6616 Remote SDP: >>>> v=0 >>>> o=sipcli-Session 1883669566 1798766211 IN IP4 142.54.179.218 >>>> s=sipcli >>>> c=IN IP4 142.54.179.218 >>>> t=0 0 >>>> m=audio 5072 RTP/AVP 18 0 8 101 >>>> a=rtpmap:18 G729/8000 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> a=ptime:20 >>>> >>>> 2015-01-14 16:41:34.203196 [DEBUG] sofia.c:6868 ( >>>> sofia/external/1000 at 177.31.245.177) State Change CS_NEW -> CS_INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send >>>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:491 ( >>>> sofia/external/1000 at 177.31.245.177) State NEW >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >>>> sofia/external/1000 at 177.31.245.177) Running State Change CS_INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( >>>> sofia/external/1000 at 177.31.245.177) State INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:87 >>>> sofia/external/1000 at 177.31.245.177 SOFIA INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:40 >>>> sofia/external/1000 at 177.31.245.177 Standard INIT >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:48 ( >>>> sofia/external/1000 at 177.31.245.177) State Change CS_INIT -> CS_ROUTING >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_session.c:1388 Send >>>> signal sofia/external/1000 at 177.31.245.177 [BREAK] >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:512 ( >>>> sofia/external/1000 at 177.31.245.177) State INIT going to sleep >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:472 ( >>>> sofia/external/1000 at 177.31.245.177) Running State Change CS_ROUTING >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_channel.c:2184 ( >>>> sofia/external/1000 at 177.31.245.177) Callstate Change DOWN -> RINGING >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:528 ( >>>> sofia/external/1000 at 177.31.245.177) State ROUTING >>>> 2015-01-14 16:41:34.203196 [DEBUG] mod_sofia.c:123 >>>> sofia/external/1000 at 177.31.245.177 SOFIA ROUTING >>>> 2015-01-14 16:41:34.203196 [DEBUG] switch_core_state_machine.c:166 >>>> sofia/external/1000 at 177.31.245.177 Standard ROUTING >>>> >>> >>> This is someone from outside calling in to your system, the public >>> context is a sandbox to allow you to isolate and route non-authenticated >>> traffic to your internal contexts via the transfer app, per the vanilla >>> config examples. This is also whey you need to understand how to secure >>> FreeSWITCH. >>> >>> >>> >>>> 2015-01-14 16:41:34.203196 [INFO] mod_dialplan_xml.c:558 Processing >>>> 1000 <1000>->9009972599796504 in context public >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/9fd6e10b/attachment-0001.html From olegstolyar at gmail.com Wed Jan 14 22:25:19 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 14 Jan 2015 11:25:19 -0800 Subject: [Freeswitch-users] Detecting originate failures from dialplan In-Reply-To: References: Message-ID: Thanks David! I did look at continue_on_fail before asking but it doesn't see to work for originate. I did find a solution though. I made the originate call to execute inline and then check for the value of api_result variable which in my dialplan gets assigned the return of the originate function. This conditions seems to tell me that the originate was successful On Wed, Jan 14, 2015 at 9:16 AM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Take a look at "continue_on_fail":Hangup Causesbridge_hangup_cause > > > This is set to the hangup cause of the last bridged B leg of the call. If > you have continue_on_fail=true and hangup_after_bridge=false you can do > checks on this to see what "really" happened to the call. You can for > instance do execute_extension after bridge, do a condition check on > ${bridge_hangup_cause} to see if it contains MEDIA_TIMEOUT and then trigger > a redial of the call or transfer to a cell phone. For a list of hangup > causes, see Hangup Causes > . > > > On Wed, Jan 14, 2015 at 6:16 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> take a look at "continue_on_fail": >> >> >> On Wed, Jan 14, 2015 at 6:05 PM, Oleg Stolyar >> wrote: >> >>> Hi guys, >>> >>> I have a line like this in my dialplan: >>> >>> >> data="api_result=${originate(sofia/external/@2.3.4.5:5060 >>> XML >>> >>> How can I know if the originate failed? For example the number is >>> unreacheable or invalid or the trunk is down. In this case the >>> will not be executed. >>> >>> It seems that my dialplan continues to execute even if originate failed >>> but is there something I can do to check for that? >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> DVG >> >> -- >> Imagination is more important than knowledge >> Albert Einstein >> > > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/b4b6a8ca/attachment.html From frederick at targointernet.com Thu Jan 15 00:13:45 2015 From: frederick at targointernet.com (Frederick Pruneau) Date: Wed, 14 Jan 2015 16:13:45 -0500 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: Sorry for the delay. I have attached a PNG file of my topology. 2014-12-19 17:01 GMT-05:00 Brian West : > Describe your topology a little bit. > > On Fri, Dec 19, 2014 at 2:31 PM, Frederick Pruneau < > frederick at targointernet.com> wrote: >> >> Sorry, I pasted all my log file. I have a new pastebin: >> >> https://pastebin.freeswitch.org/23771 >> >> I tested it and I can open it. >> >> 2014-12-19 13:05 GMT-05:00 Brian West : >> >> Never able to load your pastebin, it would timeout and not load, what >>> exactly did you paste in there? >>> >>> On Fri, Dec 19, 2014 at 8:56 AM, Frederick Pruneau < >>> frederick at targointernet.com> wrote: >>>> >>>> Did you find something? >>>> >>>> 2014-12-17 10:36 GMT-05:00 Frederick Pruneau < >>>> frederick at targointernet.com>: >>>> >>>>> Here it is: https://pastebin.freeswitch.org/23746 >>>>> >>>>> It is my freeswitch log. I have followed this guide before: >>>>> >>>>> https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP >>>>> >>>>> I have enabled this and make a call: >>>>> >>>>> sofia global siptrace on >>>>> sofia loglevel all 9 >>>>> sofia tracelevel alert >>>>> console loglevel debug >>>>> fsctl debug_level 10 >>>>> >>>>> This is what you will get in my pastebin >>>>> >>>>> >>>>> 2014-12-17 9:27 GMT-05:00 Brian West : >>>>> >>>>>> sofia global siptrace on >>>>>> >>>>>> from fs_cli >>>>>> >>>>>> On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < >>>>>> frederick at targointernet.com> wrote: >>>>>>> >>>>>>> Sorry for this noob question but how can I see sip traffic? Is there >>>>>>> a specific command to show this? Is it what we find in freeswitch.log? If >>>>>>> so, I attached my log file in my first post. >>>>>>> >>>>>>> Thanks for you help >>>>>>> >>>>>>> 2014-12-16 15:44 GMT-05:00 Brian West : >>>>>>> >>>>>>>> have you looked at the signalling? What does the sip traffic >>>>>>>> show? Please pastebin that. >>>>>>>> >>>>>>>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>> >>>>>>>>> Same problem... >>>>>>>>> >>>>>>>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>>>>>>> >>>>>>>>>> Guessing you don't have UPNP or NAT-PMP on your network, there >>>>>>>>>> for that won't work, >>>>>>>>>> >>>>>>>>>> ext-sip-ip=autonat:x.x.x.x >>>>>>>>>> ext-rtp-ip=autonat:x.x.x.x >>>>>>>>>> >>>>>>>>>> Set local-network-ac to rfc1918.auto >>>>>>>>>> >>>>>>>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>>>>>>> support at targointernet.com> wrote: >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>>>>>>> >>>>>>>>>>>> On the system behind nat what do you have ext-rtp-ip, >>>>>>>>>>>> ext-sip-ip and local-network-acl set to? >>>>>>>>>>>> >>>>>>>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>> >>>>>>>>>>>>> We have an issue with one freeswitch server behind nat. We >>>>>>>>>>>>> have a setup like this: >>>>>>>>>>>>> >>>>>>>>>>>>> -One master Freeswitch server >>>>>>>>>>>>> >>>>>>>>>>>>> -One freeswitch server connected to the master (Public IP) - >>>>>>>>>>>>> Server A >>>>>>>>>>>>> >>>>>>>>>>>>> -One freeswitch server connected to the master (behind nat) - >>>>>>>>>>>>> Server B >>>>>>>>>>>>> >>>>>>>>>>>>> If server A call server B, nothing happens. There is no sound. >>>>>>>>>>>>> After 30 sec, it times out. We have done a tcpdump. From server A to master >>>>>>>>>>>>> packets are ok. From Master to server B, we have seen that there is no >>>>>>>>>>>>> source and no destination ports for sip invite. >>>>>>>>>>>>> >>>>>>>>>>>>> If we use our cellphone and we call server B, there is no >>>>>>>>>>>>> problem. >>>>>>>>>>>>> >>>>>>>>>>>>> I have attached the failed call pcap file and freeswitch's log >>>>>>>>>>>>> file so you can take a look at them. >>>>>>>>>>>>> >>>>>>>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release >>>>>>>>>>>>> since we have issues with this server) >>>>>>>>>>>>> >>>>>>>>>>>>> Thanks in advance. >>>>>>>>>>>>> >>>>>>>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> 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 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> 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 >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> >> Fr?d?rick Pruneau >> Administrateur r?seau | Network administrator >> Targo Communications >> Ste-Clotilde : (450) 826-0031 >> Montr?al : *(514) 448-0773 <%28514%29%20448-0773> * >> >> >> _________________________________________________________________________ >> 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 > -- Fr?d?rick Pruneau Administrateur r?seau | Network administrator Targo Communications Ste-Clotilde : (450) 826-0031 Montr?al : *(514) 448-0773 * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/52eb873d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: topo.png Type: image/png Size: 75715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/52eb873d/attachment-0001.png From pasha at prosperity4ever.com Thu Jan 15 00:30:58 2015 From: pasha at prosperity4ever.com (Pasha) Date: Wed, 14 Jan 2015 13:30:58 -0800 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: Message-ID: <54B6E012.6020105@prosperity4ever.com> When you're behind NAT you have to make sure that your firewall rewrites the packets and allows them through. It sounds like your port 5060 is open and is being sent to the correct internal machine (server B), but your media is not (RTP), you need to make sure that you are allowing and forwarding the RTP ports as well (I can't remember the exact range, but I believe they are specified somewhere in freeswitch config) but it's something like 10,000-22,000 (it's a range), then your audio should be working. Depending on your setup you might have to setup source NAT as well as destination NAT. I have a working environment in production where all my servers are behind NAT as well as all my clients connecting to them are behind NAT as well via openvpn (vpn NAT too hehe), so I know it works, just need to play with it enough until you get it working properly. Hope that helped. Paul On 15-01-14 01:13 PM, Frederick Pruneau wrote: > Sorry for the delay. > > I have attached a PNG file of my topology. > > 2014-12-19 17:01 GMT-05:00 Brian West >: > > Describe your topology a little bit. > > On Fri, Dec 19, 2014 at 2:31 PM, Frederick Pruneau > > > wrote: > > Sorry, I pasted all my log file. I have a new pastebin: > > https://pastebin.freeswitch.org/23771 > > I tested it and I can open it. > > 2014-12-19 13:05 GMT-05:00 Brian West >: > > Never able to load your pastebin, it would timeout and not > load, what exactly did you paste in there? > > On Fri, Dec 19, 2014 at 8:56 AM, Frederick Pruneau > > wrote: > > Did you find something? > > 2014-12-17 10:36 GMT-05:00 Frederick Pruneau > >: > > Here it is: https://pastebin.freeswitch.org/23746 > > It is my freeswitch log. I have followed this > guide before: > > https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP > > I have enabled this and make a call: > > sofia global siptrace on > sofia loglevel all 9 > sofia tracelevel alert > console loglevel debug > fsctl debug_level 10 > > This is what you will get in my pastebin > > > 2014-12-17 9:27 GMT-05:00 Brian West > >: > > sofia global siptrace on > > from fs_cli > > On Wed, Dec 17, 2014 at 7:44 AM, Frederick > Pruneau > wrote: > > Sorry for this noob question but how can I > see sip traffic? Is there a specific > command to show this? Is it what we find > in freeswitch.log? If so, I attached my > log file in my first post. > > Thanks for you help > > 2014-12-16 15:44 GMT-05:00 Brian West > >: > > have you looked at the signalling? > What does the sip traffic show? > Please pastebin that. > > On Tue, Dec 16, 2014 at 2:37 PM, > Frederick Pruneau > > > wrote: > > Same problem... > > 2014-12-16 13:55 GMT-05:00 Brian > West >: > > Guessing you don't have UPNP > or NAT-PMP on your network, > there for that won't work, > > ext-sip-ip=autonat:x.x.x.x > ext-rtp-ip=autonat:x.x.x.x > > Set local-network-ac to > rfc1918.auto > > On Tue, Dec 16, 2014 at 12:15 > PM, Support Technique > > > wrote: > > value="auto-nat"/> > value="auto-nat"/> > name="local-network-acl" > value="localnet.auto"/> > > 2014-12-16 12:25 GMT-05:00 > Brian West > >: > > > On the system behind > nat what do you have > ext-rtp-ip, ext-sip-ip > and local-network-acl > set to? > > On Tue, Dec 16, 2014 > at 10:41 AM, Frederick > Pruneau > > > wrote: > > Hi guys, > > We have an issue > with one > freeswitch server > behind nat. We > have a setup like > this: > > -One master > Freeswitch server > > -One freeswitch > server connected > to the master > (Public IP) - Server A > > -One freeswitch > server connected > to the master > (behind nat) - > Server B > > If server A call > server B, nothing > happens. There is > no sound. After 30 > sec, it times out. > We have done a > tcpdump. From > server A to master > packets are ok. > From Master to > server B, we have > seen that there is > no source and no > destination ports > for sip invite. > > If we use our > cellphone and we > call server B, > there is no problem. > > I have attached > the failed call > pcap file and > freeswitch's log > file so you can > take a look at them. > > Master = > Freeswitch v1.4.13 > Server A = > Freeswitch v.1.4.13 > Server B = > Freeswitch > v.1.4.14 (Updated > to latest release > since we have > issues with this > server) > > Thanks in advance. > > PS: The failed > call is from > 514-448-0773 > . > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > > > -- > > Fr?d?rick Pruneau > Administrateur r?seau | Network administrator > Targo Communications > Ste-Clotilde :(450) 826-0031 > Montr?al :_(514) 448-0773 _ > > > _________________________________________________________________________ > 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 > > > > > -- > Fr?d?rick Pruneau > Administrateur r?seau | Network administrator > Targo Communications > Ste-Clotilde :(450) 826-0031 > Montr?al :_(514) 448-0773_ > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/a7970847/attachment-0001.html From krice at freeswitch.org Thu Jan 15 01:25:14 2015 From: krice at freeswitch.org (Ken Rice) Date: Wed, 14 Jan 2015 22:25:14 +0000 Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) January 4th-10th Message-ID: <54b6ecca7693b_6a2ed29330359d@ip-10-140-131-62.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1C1TXya FreeSWITCH Week in Review (Master Branch) January 4th-10th Hello, again. This week in the FreeSWITCH master branch we had 36 commits. Most of the features for this week involve mod_verto and include: fixes for recent firefox changes in mod_verto, work toward browser compatibility in mod_verto, some stuff to verto for introp, and some of the other commits worked toward updates for js, tweaking cdquality conference defaults, and adding utils. New features that were added: 8251218 Tweak cdquality conference defaults 66c425c Add utils 94bb460 Fixes for recent firefox changes in mod_verto 41bfc18 Add some stuff to verto for introp 9ca115c More work toward browser compatibility in mod_verto b79a7e1 Vid screen share placeholder ?args ?enable-usermedia-screen-capturing ?usermedia-screen-capturing b170e9e Update minified js 6c1bc0e Sync ws code Improvements to packaging: 2a05f8c Drop limit on stack size via systemd Improvements in cross platform build supports: 7c0c3ab Sofia rebuild 19a0a0f Sofia rebuild The following bugs were squashed: 4512dbc Fix multi screen bash history issues ece5cd5 FS-7088 #resolve Set flag to enable core dump from non-root users. [Jira: http://ift.tt/1C1TVXa] f48ec61 FS-7132 Resolve jsock hash overwrite issue in mod_verto [Jira: http://ift.tt/1u7zZf1] 165f542 FS-7137 Set sip_to_tag on ringing indication for inbound channels in mod_sofia 85b8631 FS-7136 Add conference member data to floor event in mod_conference 51f2442 FS-7122 Resolve an automake warning about subdirs on latest automake 16f7177 FS-7122 Resolve an automake warning about subdirs on latest automake 6afc2b5 FS-6688 Fix resubscribe through proxy with record route when the resub does not have a record route and the route has uri params [Jira: http://ift.tt/1yzQEuo] d199060 FS-7036 Fixed race condition [Jira: http://ift.tt/1u7zYrv] a2b5356 FS-7131 Fix for a segfault [Jira: http://ift.tt/1u7zZf5] d82611a FS-7142 Fix build of freetdm on CentOS c460c00 FS-7134 Improved Zoiper subscribes [Jira: http://ift.tt/1C1TWdy] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/306fcfe0/attachment.html From sharath.kumar at mezocliq.com Thu Jan 15 01:42:32 2015 From: sharath.kumar at mezocliq.com (Kumar, Sharath) Date: Wed, 14 Jan 2015 17:42:32 -0500 Subject: [Freeswitch-users] Video MCU Message-ID: All, Does anyone knows if the Freeswitch 1.6 beta will support an video MCU ? Any timelines? thank you, Shaks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/9d95c8d0/attachment.html From brian at freeswitch.org Thu Jan 15 02:50:18 2015 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jan 2015 17:50:18 -0600 Subject: [Freeswitch-users] Video MCU In-Reply-To: References: Message-ID: Please contact consulting at freeswitch.org for additional details. On Wed, Jan 14, 2015 at 4:42 PM, Kumar, Sharath wrote: > All, > Does anyone knows if the Freeswitch 1.6 beta will support an video MCU ? > Any timelines? > thank you, > Shaks. > > _________________________________________________________________________ > 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/20150114/7db980fc/attachment.html From msc at freeswitch.org Thu Jan 15 06:19:29 2015 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Jan 2015 19:19:29 -0800 Subject: [Freeswitch-users] robo caller In-Reply-To: References: Message-ID: Areski, Does Newfies support mod_com_amd? Just curious how you handle machines. -MC On Wed, Jan 14, 2015 at 8:44 AM, Areski wrote: > Newfies-Dialer (http://www.newfies-dialer.org/) might help and obviously > it's built on top of FreeSWITCH. > We built a flexible module for appointment reminders: > http://docs.newfies-dialer.org/en/latest/user-guide-doc/appointment.html > > If you want to code this yourself, we use ESL ( > https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library) > to originate the calls and Lua to build the IVR part ( > https://freeswitch.org/confluence/display/FREESWITCH/mod_lua). > > On Wed, Jan 14, 2015 at 5:37 PM, Moishe Grunstein > wrote: > >> You can also have a look at http://www.newfies-dialer.org/ >> >> >> >> Thanks, >> >> >> >> Moishe Grunstein >> >> Tornado Computer Systems, Inc. >> >> 212.400.7650 888.IPPBX.US >> *Service Request Email: support at nysolutions.com >> * >> >> [image: cid:image001.jpg at 01C72F94.9EE45D60] >> >> Computer Networking * Managed Services * IP Video Surveillance * Network >> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >> Security * Site Surveys * CMS >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David >> Villasmil Govea >> *Sent:* Wednesday, January 14, 2015 11:31 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] robo caller >> >> >> >> That's fairly simple to implement. Look into at mod_amd, orignate and >> ivr. >> >> On Jan 14, 2015 5:28 PM, "Sean Devoy" wrote: >> >> Does anyone have a sample RoboCaller script? Perhaps I am using the >> wrong name and that is why I can?t find one. I have a doctor?s office that >> wants to automate the reminder calls about appointments to their patients. >> >> >> >> I am curious how people handle answering machine detection as well? >> >> >> >> Thanks, >> >> Sean >> >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Kind regards, > /Areski > > ---- > Arezqui Belaid, > Founder at Star2Billing (www.star2billing.com) > > Tel: +34650784355 > Twitter: http://twitter.com/areskib > LinkedIn: http://www.linkedin.com/in/areski > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/c1b64411/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/c1b64411/attachment-0001.jpg From msc at freeswitch.org Thu Jan 15 06:25:59 2015 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Jan 2015 19:25:59 -0800 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <007901d02ff5$67075090$3515f1b0$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <147601d02f32$37a9f5a0$a6fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> <007901d02ff5$67075090$3515f1b0$@gfphelps.com> Message-ID: Glad you have the book. On page 19 it covers the use of enterprise originate. I think possibly you need to use the method as discussed on page 21. Try something like this: Could just be that early media is not being ignored on both user dialout attempts. -MC On Wed, Jan 14, 2015 at 4:27 AM, George F. Phelps wrote: > Michael Collins, > > > > I already have the book. Thanks! > > > > Here?s my dialplan: > > > > > > > > > > > > data="{ignore_early_media=true}user/1000:_:user/1001"/> > > > > > > > > > > New log file uploaded to: > > > > *http://pastebin.com/gnEpPzk9 * > > > > To me, the most significant event in the log file is the SIP CANCEL > message ? starting at line #321: > > > > tport.c:3023 tport_deliver() tport_deliver(0x95daa0): msg 0xad8fb0 (437 > bytes) from udp/169.XX.XX.XX:5080/sip next=(nil) > > nta.c:2880 agent_recv_request() nta: received CANCEL > sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (CSeq 1) > > nta.c:3026 agent_recv_request() nta: CANCEL (1) is going to INVITE (1) > > > > I don?t think it?s related, but I am also curious about log file line #285: > > > > sres.c:2987 sres_query_report_error() sres(q=0x98b050): reporting error > NAME_ERR for SRV _sip._udp.sip.switch2voip.us > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Wednesday, January 14, 2015 2:41 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > > We covered this nicely in chapter 1 of the FreeSWITCH Cookbook > > > > I'm sorry that I'm late to the party so I am missing some information. Can > you pastebin not only the call log but also the dialplan code for the > example in question? One other tip: it appears that the log that you are > pasting is coming directly from the FreeSWITCH console. By default the > console does not have debug level output enabled. Try entering the command > "console loglevel debug" and you'll see way more log lines, mostly yellow > text. Those lines will most likely contain the clues needed to unravel this > mystery. > > Thanks, > > MC > > > > On Tue, Jan 13, 2015 at 2:42 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > New logfile uploaded to: > > > > *http://pastebin.com/CFFvVarS * > > > > The log contains default Freeswitch console log messages, plus a SIP trace > of a failed call. BTW, both extensions were ringing ? prior to the CANCEL > message (see context below). > > > > In the log I see the INVITE from my VoIP service provider: > > > > recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: > > ------------------------------------------------------------------------ > > INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 > > > > send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > > > (Then, subsequent INVITE messages to my two extensions. But other no > messages to/from my VoIP service provider.) > > > > And then, a spontaneous CANCEL from my VoIP service provider, > approximately 10 seconds after the initial INVITE message. Due to a SIP > ?Timer B? timeout? Seems way too short. > > > > recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: > > ------------------------------------------------------------------------ > > CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 > > > > (Freeswitch cleanup of SIP sessions to my extensions?) > > > > > > Bote Man--> I have two local extensions. Individually, the extensions > can make and receive both internal and external calls. It?s only the > simultaneous ringing for external, inbound calls that is not working at the > moment. > > > > Thanks, > > > > George > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bote Man > *Sent:* Tuesday, January 13, 2015 2:33 PM > > > *To:* 'FreeSWITCH Users Help' > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > I suggest you configure and register 3 total local phones to your FS > installation, configure 2 of them as the target of your simultaneous ring > group, and call them with the 3rd phone. Until you can get that working, > calling through a carrier is adding another layer of complexity to the > problem and confusing the issue. > > > Out of the box FreeSWITCH does not utter voice codes, they must be coming > from your carrier. > > > > Also, the debug-level logs very likely tell you exactly what is happening, > even though they can be staggering to decipher as a newcomer to FS. > Learning how to read them pays off in so many ways, though. I find the > color-coded logs on the console or viewed via FS_cli to be helpful in these > instances. > > > > Bote > > > > > > *From:* George F. Phelps > *Sent:* Tuesday, 13 January, 2015 08:10 > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > I tried? > > > > ?but that did not resolve the problem. > > > > 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:* Tuesday, January 13, 2015 7:58 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Correct, first endpoint providing audio wins, but you're using > ignore_early_media... > Try using > > Which is global. And I believe in the dial string also is. > But try it anyway. > > On Jan 13, 2015 1:50 PM, "George F. Phelps" > wrote: > > David Govea, > > > > It appears that the essence of the problem is: > > > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] > > [NOTICE] switch_ivr_originate.c:3495 Hangup > sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] > > > > Various Freeswitch web comments, related to the same problem, indicate > that I should: ?*Ok. Setting it per leg didn't help > [ignore_early_media=true], but per channel {ignore_early_media=true} worked* > ?. > > > > What dialplan(?) syntax do I use to correctly ?set > ignore_early_media=true? on a per channel basis? I tried, within my > dialplan? > > > > > > data="{ignore_early_media=true}user/1000:_:{ignore_early_media=true}user/1001"/> > > > > ?but without success. Or does setting ignore_early_media have to be done > somewhere else? > > > > 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:* Tuesday, January 13, 2015 6:36 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > You need to have both extensions registered. Register both and try again > and paste de log. > > On Jan 13, 2015 12:30 PM, "George F. Phelps" > wrote: > > For the most recent test/logfile, only extension 1001 was registered ? to > reduce the number of debug messages in the logfile. > > > > 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:* Monday, January 12, 2015 6:16 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Is 1000 registered? The log says it's not registered... > > > > On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > David Govea, > > > > I uploaded a new Freeswitch debug logfile at: > > > > *http://pastebin.com/v17SyXhh * > > > > *Notes* > > > > Only extension 1001 was registered for this test. > > > > Dialstring segment: data="{ignore_early_media=true}user/1000:_:user/1001"/> > > > > I?m guessing that ?*verbal error code 231*? is being generated by my VoIP > service provider. > > > > I am running Freeswitch with (mostly) the default configuration. Changed > passphrases, added my gateway, etc. > > > > I downloaded the source code from git and built it unmodified, from > scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit > (git 1965b3b 2014-12-30 15:06:32Z 64bit)? > > > > My effective codec is G711U ? fully supported throughout the call chain. > > > > 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:* Monday, January 12, 2015 7:15 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > BTW, I've never heard of verbal error code 231, that's why I ask whether > you downloaded and freeswitch from the git... > > > > On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > > Are you using freeswitch with its default config or did you install > something like fusionpbx? > > Can you please post your log now? the log for the last dial string, where > calls go out and then get hung up. > > (Are you sure your codecs are correct?) > > > > On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps < > GeorgePhelps at gfphelps.com> wrote: > > David Govea, > > > > Still fails; both extensions rang. However, before I can answer either > one, I heard the same verbal error code: ?231?. > > > > How do I track down the meaning of ?231?? > > > > 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:* Monday, January 12, 2015 6:14 AM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > You can also try: > > bridge user/1001:_:user/1002 > > On Jan 12, 2015 12:04 PM, "George F. Phelps" > wrote: > > David Govea, > > > > That syntax, with more than one extension specified, causes the following > Freeswitch warning log message: > > > > [WARNING] switch_ivr_originate.c:2531 Only calling the first element in > the list in this mode. > > > > However, the call ? to only the first extension on the list ? does work. > > > > 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:* Monday, January 12, 2015 3:21 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > try this: > > > > > > > > > > On Jan 12, 2015 4:33 AM, "George F. Phelps" > wrote: > > Here you go: > > > > > > > > > > > > data="{ignore_early_media=true}sofia/internal/1001%${domain}"/> > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > > > 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, January 11, 2015 10:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > > Yes, I tested with that dialstring. My extension was registered, and > online. > > > > The call disconnects with verbal error code ?231?. The associated > logfile is at: > > > > http://pastebin.com/BeWhhgSU > > > > 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, January 11, 2015 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If > extension 1001 is registered they should get the call. What happens when > you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > > David Govea, > > > > I am attempting to implement simultaneous ringing ? where when one of my > inbound DIDs is called, then two SIP extensions and one outbound DID are > all rung at the same time. Simultaneous ringing is also referred, in the > Freeswitch documentation, as ?forked dialing? and ?calling multiple > destinations.? > > > > I am trying to get the first extension to work with ?bridge.? > > > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > > > *Calling multiple destinations > * > > By using commas to separate the addresses, bridge will dial them > simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate > multiple destinations to be dialed in a multi-threaded manner (this is > referred to as "Enterprise Origination") - this gives more flexibility (and > avoids the "Only calling the first element in the list in this mode" > warning) > > If you need to set different channel variables for each destination, you > may prefix the destinations with [] and the variables inside the brackets. > Example: > > data="[origination_caller_id_number=1234]sofia//, > [origination_caller_id_number=55555]sofia//"/> > > > > 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, January 11, 2015 7:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about > bridge. You can't "bridge" to an extension. Can you please explain in > detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > > David Govea, > > > > Thanks for your input. I tried that coding yesterday, and the call > failed. I wasn?t 100 percent sure I was using the correct coding. When I > call, I hear spoken error ?231? and then the call hangs up. > > > > I created a pastebin.com of the failed call log, at: > > > > http://pastebin.com/BeWhhgSU > > > > A reminder that this ?transfer? statement works: > > > > > > > > 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, January 11, 2015 4:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > > Can someone help me with my question? > > > > Thanks, > > > > George > > > > *From:* George F. Phelps [mailto:GeorgePhelps at gfphelps.com] > *Sent:* Saturday, January 10, 2015 12:02 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* How to Bridge To Local Extensions? > > > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The > following statement does not work for me: > > > > > > > > My extensions are effectively default values and in the default directory > location. For example: > > > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> > > > > Thanks, > > > > George > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/20c4a80a/attachment-0001.html From msc at freeswitch.org Thu Jan 15 06:30:31 2015 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Jan 2015 19:30:31 -0800 Subject: [Freeswitch-users] postgresql In-Reply-To: References: <1420922004.999033200@f404.i.mail.ru> Message-ID: Also, most web servers are designed to handle many multiple requests. If you have a robust web server/db backend then you can scale as needed and do failover, etc. -MC On Tue, Jan 13, 2015 at 11:53 PM, John Nash wrote: > OK I see the point of manipulating CDR data (Even bill CDR as you said). I > am just trying to figure out best approach. But I also see a downside of > http post as it will keep web server busy too for each call made specially > in high volume systems (Assuming user portal/reporting also are being used > on same web server) > > On the other hand if pg_cdr module is used with postgres function we can > manipulating CDR too without keeping web server busy. > > > > > > On Wed, Jan 14, 2015 at 1:15 PM, Michael Collins > wrote: > >> I think David and I agree on these two basic points regarding >> mod_xml_curl: >> You can validate/manipulate/massage the data prior to insertion >> It's uncomplicated, flexible, and reliable >> >> As for connection pooling I'll have to defer to those who use >> mod_cdr_pg_csv. >> -MC >> >> >> On Tue, Jan 13, 2015 at 11:33 PM, John Nash >> wrote: >> >>> Hello David and Michael, >>> >>> As per this page >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_pg_csv we >>> can define disk spooling in case of insert failure. What are the other >>> advantages of xml_cdr over cdr_pg_csv? >>> >>> Also does cdr_pg_csv use connection pooling? >>> >>> Regards >>> >>> Manoj >>> >>> On Wed, Jan 14, 2015 at 9:05 AM, David Villasmil Govea < >>> david.villasmil at gmail.com> wrote: >>> >>>> You can even do the billing with it... I've had it running in >>>> production for years, not a single issue... >>>> Is it an absolute requirement to log directly to the database? There >>>> has been thorough and intense discussion on this list over the years as to >>>> why this is/isn't a good idea. IMHO the best option is to use mod_xml_cdr >>>> (see https://wiki.freeswitch.org/wiki/Mod_xml_cdr). It lets you post >>>> to a database and store on disk in case of failure. It's also pretty easy >>>> to use and lets you do logic/validation prior to storing in the db. >>>> >>>> -MC >>>> >>>> >>>> On Sat, Jan 10, 2015 at 12:33 PM, Nick Zaitsev >>>> wrote: >>>> >>>>> Good day to you, >>>>> Please,advise me, how can i use postgres schema in freeswitch config >>>>> (cdr_pg_csv.conf.xml)? >>>>> I'd like to use cdr schema instead public, for example here >>>>> >>>>> >>>>> -- >>>>> Nick Zaitsev >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/bd46bbae/attachment.html From msc at freeswitch.org Thu Jan 15 07:19:07 2015 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Jan 2015 20:19:07 -0800 Subject: [Freeswitch-users] Security Issue In-Reply-To: <4BFC98A1-CF17-4ACF-9E4C-EF8D6B447FA5@gmail.com> References: <4BFC98A1-CF17-4ACF-9E4C-EF8D6B447FA5@gmail.com> Message-ID: On Wed, Jan 14, 2015 at 9:40 AM, Ahmed Habiba wrote: > Thank you really David, > > Here is my point, the sip-trace in the first mail shows that, the call > comes to public context mainly through port 5080, and however the > originator IP was not defined in my ACL list Freeswitch continue to process > the call for some reason. > Just an FYI, the external profile does not have auth-calls param set to true, so FS simply tries to route the call in the public context without sending back an auth challenge. Since the public context is pretty paranoid it's not exactly easy to dial out. Also, just because FS tries to route the call does not mean that FS considers the call to be "authenticated." If you want all traffic coming in to your server to be authenticated then either send it all to the internal profile (i.e. port 5060) or add auth-calls to your external profile. The bigger question you may want to ask is: why are these random IP even getting to your server? Do you allow public access to your system? If so, why? If not, then you need a firewall (iptables or whatnot) to block those SIP messages from ever getting to your FreeSWITCH. You may also be interested in something like fail2ban and voipbl.org. -MC > > even if it come to 5060, I was expecting some request for digest > authentication, which is not shown in the log. > > *From: *David Villasmil Govea > *To: *FreeSWITCH Users Help > *Date: *January 14, 2015 at 8:30:35 PM GMT+3 > *Reply-To: *FreeSWITCH Users Help > *Subject: **Re: [Freeswitch-users] Security Issue* > > > Authorization is done if you configure your sip profile to do it. By > default 5060 (internal) requires authentication, 5080 (external) doesn't > but it does use the ACL to allow or not calls. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/3a145fa2/attachment-0001.html From msc at freeswitch.org Thu Jan 15 07:20:57 2015 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Jan 2015 20:20:57 -0800 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: <54B6E012.6020105@prosperity4ever.com> References: <54B6E012.6020105@prosperity4ever.com> Message-ID: For posterity, RTP port range is specified in autoload_configs/switch.conf.xml: -MC On Wed, Jan 14, 2015 at 1:30 PM, Pasha wrote: > When you're behind NAT you have to make sure that your firewall rewrites > the packets and allows them through. It sounds like your port 5060 is open > and is being sent to the correct internal machine (server B), but your > media is not (RTP), you need to make sure that you are allowing and > forwarding the RTP ports as well (I can't remember the exact range, but I > believe they are specified somewhere in freeswitch config) but it's > something like 10,000-22,000 (it's a range), then your audio should be > working. Depending on your setup you might have to setup source NAT as well > as destination NAT. > > I have a working environment in production where all my servers are behind > NAT as well as all my clients connecting to them are behind NAT as well via > openvpn (vpn NAT too hehe), so I know it works, just need to play with it > enough until you get it working properly. > > Hope that helped. > > Paul > > > On 15-01-14 01:13 PM, Frederick Pruneau wrote: > > Sorry for the delay. > > I have attached a PNG file of my topology. > > 2014-12-19 17:01 GMT-05:00 Brian West : > >> Describe your topology a little bit. >> >> On Fri, Dec 19, 2014 at 2:31 PM, Frederick Pruneau < >> frederick at targointernet.com> wrote: >>> >>> Sorry, I pasted all my log file. I have a new pastebin: >>> >>> https://pastebin.freeswitch.org/23771 >>> >>> I tested it and I can open it. >>> >>> 2014-12-19 13:05 GMT-05:00 Brian West : >>> >>> Never able to load your pastebin, it would timeout and not load, what >>>> exactly did you paste in there? >>>> >>>> On Fri, Dec 19, 2014 at 8:56 AM, Frederick Pruneau < >>>> frederick at targointernet.com> wrote: >>>>> >>>>> Did you find something? >>>>> >>>>> 2014-12-17 10:36 GMT-05:00 Frederick Pruneau < >>>>> frederick at targointernet.com>: >>>>> >>>>>> Here it is: https://pastebin.freeswitch.org/23746 >>>>>> >>>>>> It is my freeswitch log. I have followed this guide before: >>>>>> >>>>>> https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP >>>>>> >>>>>> I have enabled this and make a call: >>>>>> >>>>>> sofia global siptrace on >>>>>> sofia loglevel all 9 >>>>>> sofia tracelevel alert >>>>>> console loglevel debug >>>>>> fsctl debug_level 10 >>>>>> >>>>>> This is what you will get in my pastebin >>>>>> >>>>>> >>>>>> 2014-12-17 9:27 GMT-05:00 Brian West : >>>>>> >>>>>>> sofia global siptrace on >>>>>>> >>>>>>> from fs_cli >>>>>>> >>>>>>> On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < >>>>>>> frederick at targointernet.com> wrote: >>>>>>>> >>>>>>>> Sorry for this noob question but how can I see sip traffic? Is >>>>>>>> there a specific command to show this? Is it what we find in >>>>>>>> freeswitch.log? If so, I attached my log file in my first post. >>>>>>>> >>>>>>>> Thanks for you help >>>>>>>> >>>>>>>> 2014-12-16 15:44 GMT-05:00 Brian West : >>>>>>>> >>>>>>>>> have you looked at the signalling? What does the sip traffic >>>>>>>>> show? Please pastebin that. >>>>>>>>> >>>>>>>>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>>> >>>>>>>>>> Same problem... >>>>>>>>>> >>>>>>>>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>>>>>>>> >>>>>>>>>>> Guessing you don't have UPNP or NAT-PMP on your network, there >>>>>>>>>>> for that won't work, >>>>>>>>>>> >>>>>>>>>>> ext-sip-ip=autonat:x.x.x.x >>>>>>>>>>> ext-rtp-ip=autonat:x.x.x.x >>>>>>>>>>> >>>>>>>>>>> Set local-network-ac to rfc1918.auto >>>>>>>>>>> >>>>>>>>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>>>>>>>> support at targointernet.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>>>>>>>> >>>>>>>>>>>>> On the system behind nat what do you have ext-rtp-ip, >>>>>>>>>>>>> ext-sip-ip and local-network-acl set to? >>>>>>>>>>>>> >>>>>>>>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>>> >>>>>>>>>>>>>> We have an issue with one freeswitch server behind nat. We >>>>>>>>>>>>>> have a setup like this: >>>>>>>>>>>>>> >>>>>>>>>>>>>> -One master Freeswitch server >>>>>>>>>>>>>> >>>>>>>>>>>>>> -One freeswitch server connected to the master (Public IP) >>>>>>>>>>>>>> - Server A >>>>>>>>>>>>>> >>>>>>>>>>>>>> -One freeswitch server connected to the master (behind nat) >>>>>>>>>>>>>> - Server B >>>>>>>>>>>>>> >>>>>>>>>>>>>> If server A call server B, nothing happens. There is no >>>>>>>>>>>>>> sound. After 30 sec, it times out. We have done a tcpdump. From server A to >>>>>>>>>>>>>> master packets are ok. From Master to server B, we have seen that there is >>>>>>>>>>>>>> no source and no destination ports for sip invite. >>>>>>>>>>>>>> >>>>>>>>>>>>>> If we use our cellphone and we call server B, there is no >>>>>>>>>>>>>> problem. >>>>>>>>>>>>>> >>>>>>>>>>>>>> I have attached the failed call pcap file and freeswitch's >>>>>>>>>>>>>> log file so you can take a look at them. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release >>>>>>>>>>>>>> since we have issues with this server) >>>>>>>>>>>>>> >>>>>>>>>>>>>> Thanks in advance. >>>>>>>>>>>>>> >>>>>>>>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>> 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 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> 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 >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> 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 >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>> >>> >>> >>> -- >>> >>> Fr?d?rick Pruneau >>> Administrateur r?seau | Network administrator >>> Targo Communications >>> Ste-Clotilde : (450) 826-0031 >>> Montr?al : *(514) 448-0773 <%28514%29%20448-0773> * >>> >>> >>> _________________________________________________________________________ >>> 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 >> > > > > -- > > Fr?d?rick Pruneau > Administrateur r?seau | Network administrator > Targo Communications > Ste-Clotilde : (450) 826-0031 > Montr?al : *(514) 448-0773 <%28514%29%20448-0773> * > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150114/47d3765c/attachment-0001.html From john.nash778 at gmail.com Thu Jan 15 12:17:09 2015 From: john.nash778 at gmail.com (John Nash) Date: Thu, 15 Jan 2015 14:47:09 +0530 Subject: [Freeswitch-users] Miscellaneous question Message-ID: I have some points where I request you to please comment... 1- When using sched_hangup to auto hangup call in XX seconds, does freeswitch uses some considerable additional resource? Like may be creating one extra thread to time call or such call limit is anyway part of the design whether someone uses or not. 2- I have read in some articles where it is mentioned that freeswitch is able to bridge media (all RTP packets pass through freeswitch) without degrading "quality". Is it different than using RTPPROXY/mediaproxy better or worse? (Assuming IVR/conference etc are not being used) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/fb182702/attachment.html From ax.lyb.lei at gmail.com Thu Jan 15 10:52:16 2015 From: ax.lyb.lei at gmail.com (ax lyb) Date: Thu, 15 Jan 2015 15:52:16 +0800 Subject: [Freeswitch-users] how can i keep A-LEG do not hangup Message-ID: All: recently i write a small program based freeswitch, in this program i only do things like callcenter mod, here is what i do: 1. originate a call (? originate sofia/gateway/gw01/xxxx my-out-call XML default?), in the default.xml, i already config it as follow: 65 66 67 72 73 2. when call-out-destination customer pick up the call, i control the flow to playback a wav file, then , bridge the call to a free agent while the wav file play over in CHANNEL_EXECUTE_COMPLETE, these steps run normally ok. 3. the question is : if i have two agent in one moment, and this time i have 3 or > 3 out-call to bridge, i?ll leave the other call do nothing,for i can not get more free agents to service; some special scene is i have two agents (A1,A2) service 3 call (C1,C2,C3), A1 service C1, A2 service C2,then C3 is only A-LEG when play over a wav file,about 100 seconds later, C3 is hangup(it?s not user hangup manually), how can i do make the C3 call not hangup until i get a free agent to service it? any suggest is appreciate, ax.lyb.lei at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/1bcf9524/attachment.html From vipkilla at gmail.com Thu Jan 15 16:12:19 2015 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 15 Jan 2015 08:12:19 -0500 Subject: [Freeswitch-users] postgresql In-Reply-To: References: <1420922004.999033200@f404.i.mail.ru> Message-ID: There is a fairly new module called mod_odbc_cdr. If you're looking to inject any channel variable directly into a database you can use that module. It supports multiple tables and field customization. On Wed, Jan 14, 2015 at 10:30 PM, Michael Collins wrote: > Also, most web servers are designed to handle many multiple requests. If > you have a robust web server/db backend then you can scale as needed and do > failover, etc. > > -MC > > On Tue, Jan 13, 2015 at 11:53 PM, John Nash > wrote: > >> OK I see the point of manipulating CDR data (Even bill CDR as you said). >> I am just trying to figure out best approach. But I also see a downside of >> http post as it will keep web server busy too for each call made specially >> in high volume systems (Assuming user portal/reporting also are being used >> on same web server) >> >> On the other hand if pg_cdr module is used with postgres function we can >> manipulating CDR too without keeping web server busy. >> >> >> >> >> >> On Wed, Jan 14, 2015 at 1:15 PM, Michael Collins >> wrote: >> >>> I think David and I agree on these two basic points regarding >>> mod_xml_curl: >>> You can validate/manipulate/massage the data prior to insertion >>> It's uncomplicated, flexible, and reliable >>> >>> As for connection pooling I'll have to defer to those who use >>> mod_cdr_pg_csv. >>> -MC >>> >>> >>> On Tue, Jan 13, 2015 at 11:33 PM, John Nash >>> wrote: >>> >>>> Hello David and Michael, >>>> >>>> As per this page >>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_pg_csv we >>>> can define disk spooling in case of insert failure. What are the other >>>> advantages of xml_cdr over cdr_pg_csv? >>>> >>>> Also does cdr_pg_csv use connection pooling? >>>> >>>> Regards >>>> >>>> Manoj >>>> >>>> On Wed, Jan 14, 2015 at 9:05 AM, David Villasmil Govea < >>>> david.villasmil at gmail.com> wrote: >>>> >>>>> You can even do the billing with it... I've had it running in >>>>> production for years, not a single issue... >>>>> Is it an absolute requirement to log directly to the database? There >>>>> has been thorough and intense discussion on this list over the years as to >>>>> why this is/isn't a good idea. IMHO the best option is to use mod_xml_cdr >>>>> (see https://wiki.freeswitch.org/wiki/Mod_xml_cdr). It lets you post >>>>> to a database and store on disk in case of failure. It's also pretty easy >>>>> to use and lets you do logic/validation prior to storing in the db. >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Sat, Jan 10, 2015 at 12:33 PM, Nick Zaitsev >>>>> wrote: >>>>> >>>>>> Good day to you, >>>>>> Please,advise me, how can i use postgres schema in freeswitch config >>>>>> (cdr_pg_csv.conf.xml)? >>>>>> I'd like to use cdr schema instead public, for example here >>>>>> >>>>>> >>>>>> -- >>>>>> Nick Zaitsev >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> 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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/c8474d32/attachment-0001.html From frederick at targointernet.com Thu Jan 15 16:26:29 2015 From: frederick at targointernet.com (Frederick Pruneau) Date: Thu, 15 Jan 2015 08:26:29 -0500 Subject: [Freeswitch-users] Issue with Freeswitch Behind nat In-Reply-To: References: <54B6E012.6020105@prosperity4ever.com> Message-ID: I have already done that and the issue is still there... :-/ 2015-01-14 23:20 GMT-05:00 Michael Collins : > For posterity, RTP port range is specified in > autoload_configs/switch.conf.xml: > > > > > > > -MC > > On Wed, Jan 14, 2015 at 1:30 PM, Pasha wrote: > >> When you're behind NAT you have to make sure that your firewall rewrites >> the packets and allows them through. It sounds like your port 5060 is open >> and is being sent to the correct internal machine (server B), but your >> media is not (RTP), you need to make sure that you are allowing and >> forwarding the RTP ports as well (I can't remember the exact range, but I >> believe they are specified somewhere in freeswitch config) but it's >> something like 10,000-22,000 (it's a range), then your audio should be >> working. Depending on your setup you might have to setup source NAT as well >> as destination NAT. >> >> I have a working environment in production where all my servers are >> behind NAT as well as all my clients connecting to them are behind NAT as >> well via openvpn (vpn NAT too hehe), so I know it works, just need to play >> with it enough until you get it working properly. >> >> Hope that helped. >> >> Paul >> >> >> On 15-01-14 01:13 PM, Frederick Pruneau wrote: >> >> Sorry for the delay. >> >> I have attached a PNG file of my topology. >> >> 2014-12-19 17:01 GMT-05:00 Brian West : >> >>> Describe your topology a little bit. >>> >>> On Fri, Dec 19, 2014 at 2:31 PM, Frederick Pruneau < >>> frederick at targointernet.com> wrote: >>>> >>>> Sorry, I pasted all my log file. I have a new pastebin: >>>> >>>> https://pastebin.freeswitch.org/23771 >>>> >>>> I tested it and I can open it. >>>> >>>> 2014-12-19 13:05 GMT-05:00 Brian West : >>>> >>>> Never able to load your pastebin, it would timeout and not load, what >>>>> exactly did you paste in there? >>>>> >>>>> On Fri, Dec 19, 2014 at 8:56 AM, Frederick Pruneau < >>>>> frederick at targointernet.com> wrote: >>>>>> >>>>>> Did you find something? >>>>>> >>>>>> 2014-12-17 10:36 GMT-05:00 Frederick Pruneau < >>>>>> frederick at targointernet.com>: >>>>>> >>>>>>> Here it is: https://pastebin.freeswitch.org/23746 >>>>>>> >>>>>>> It is my freeswitch log. I have followed this guide before: >>>>>>> >>>>>>> https://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP >>>>>>> >>>>>>> I have enabled this and make a call: >>>>>>> >>>>>>> sofia global siptrace on >>>>>>> sofia loglevel all 9 >>>>>>> sofia tracelevel alert >>>>>>> console loglevel debug >>>>>>> fsctl debug_level 10 >>>>>>> >>>>>>> This is what you will get in my pastebin >>>>>>> >>>>>>> >>>>>>> 2014-12-17 9:27 GMT-05:00 Brian West : >>>>>>> >>>>>>>> sofia global siptrace on >>>>>>>> >>>>>>>> from fs_cli >>>>>>>> >>>>>>>> On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau < >>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>> >>>>>>>>> Sorry for this noob question but how can I see sip traffic? Is >>>>>>>>> there a specific command to show this? Is it what we find in >>>>>>>>> freeswitch.log? If so, I attached my log file in my first post. >>>>>>>>> >>>>>>>>> Thanks for you help >>>>>>>>> >>>>>>>>> 2014-12-16 15:44 GMT-05:00 Brian West : >>>>>>>>> >>>>>>>>>> have you looked at the signalling? What does the sip traffic >>>>>>>>>> show? Please pastebin that. >>>>>>>>>> >>>>>>>>>> On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau < >>>>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>>>> >>>>>>>>>>> Same problem... >>>>>>>>>>> >>>>>>>>>>> 2014-12-16 13:55 GMT-05:00 Brian West : >>>>>>>>>>> >>>>>>>>>>>> Guessing you don't have UPNP or NAT-PMP on your network, there >>>>>>>>>>>> for that won't work, >>>>>>>>>>>> >>>>>>>>>>>> ext-sip-ip=autonat:x.x.x.x >>>>>>>>>>>> ext-rtp-ip=autonat:x.x.x.x >>>>>>>>>>>> >>>>>>>>>>>> Set local-network-ac to rfc1918.auto >>>>>>>>>>>> >>>>>>>>>>>> On Tue, Dec 16, 2014 at 12:15 PM, Support Technique < >>>>>>>>>>>> support at targointernet.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> 2014-12-16 12:25 GMT-05:00 Brian West : >>>>>>>>>>>>> >>>>>>>>>>>>>> On the system behind nat what do you have ext-rtp-ip, >>>>>>>>>>>>>> ext-sip-ip and local-network-acl set to? >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau < >>>>>>>>>>>>>> frederick at targointernet.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> We have an issue with one freeswitch server behind nat. We >>>>>>>>>>>>>>> have a setup like this: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> -One master Freeswitch server >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> -One freeswitch server connected to the master (Public IP) >>>>>>>>>>>>>>> - Server A >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> -One freeswitch server connected to the master (behind >>>>>>>>>>>>>>> nat) - Server B >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> If server A call server B, nothing happens. There is no >>>>>>>>>>>>>>> sound. After 30 sec, it times out. We have done a tcpdump. >From server A to >>>>>>>>>>>>>>> master packets are ok. From Master to server B, we have seen that there is >>>>>>>>>>>>>>> no source and no destination ports for sip invite. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> If we use our cellphone and we call server B, there is no >>>>>>>>>>>>>>> problem. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> I have attached the failed call pcap file and freeswitch's >>>>>>>>>>>>>>> log file so you can take a look at them. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Master = Freeswitch v1.4.13 >>>>>>>>>>>>>>> Server A = Freeswitch v.1.4.13 >>>>>>>>>>>>>>> Server B = Freeswitch v.1.4.14 (Updated to latest release >>>>>>>>>>>>>>> since we have issues with this server) >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Thanks in advance. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> PS: The failed call is from 514-448-0773. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>> 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 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> 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 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> 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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Fr?d?rick Pruneau >>>> Administrateur r?seau | Network administrator >>>> Targo Communications >>>> Ste-Clotilde : (450) 826-0031 >>>> Montr?al : *(514) 448-0773 <%28514%29%20448-0773> * >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> >> >> -- >> >> Fr?d?rick Pruneau >> Administrateur r?seau | Network administrator >> Targo Communications >> Ste-Clotilde : (450) 826-0031 >> Montr?al : *(514) 448-0773 <%28514%29%20448-0773> * >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://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 >> > > > _________________________________________________________________________ > 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 > -- Fr?d?rick Pruneau Administrateur r?seau | Network administrator Targo Communications Ste-Clotilde : (450) 826-0031 Montr?al : *(514) 448-0773 * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/357d93d4/attachment-0001.html From aademattia at comcast.net Thu Jan 15 16:27:00 2015 From: aademattia at comcast.net (Andrew) Date: Thu, 15 Jan 2015 08:27:00 -0500 Subject: [Freeswitch-users] how can i keep A-LEG do not hangup In-Reply-To: References: Message-ID: <001201d030c6$f5b5e1e0$e121a5a0$@comcast.net> This may be able to help you but I set session.SetAutoHangup(false); This will stop the dial plan from ending. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ax lyb Sent: Thursday, January 15, 2015 2:52 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] how can i keep A-LEG do not hangup All: recently i write a small program based freeswitch, in this program i only do things like callcenter mod, here is what i do: 1. originate a call (" originate sofia/gateway/gw01/xxxx my-out-call XML default"), in the default.xml, i already config it as follow: 65 66 67 72 73 2. when call-out-destination customer pick up the call, i control the flow to playback a wav file, then , bridge the call to a free agent while the wav file play over in CHANNEL_EXECUTE_COMPLETE, these steps run normally ok. 3. the question is : if i have two agent in one moment, and this time i have 3 or > 3 out-call to bridge, i'll leave the other call do nothing,for i can not get more free agents to service; some special scene is i have two agents (A1,A2) service 3 call (C1,C2,C3), A1 service C1, A2 service C2,then C3 is only A-LEG when play over a wav file,about 100 seconds later, C3 is hangup(it's not user hangup manually), how can i do make the C3 call not hangup until i get a free agent to service it? any suggest is appreciate, ax.lyb.lei at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/114cadec/attachment.html From adahary at gmail.com Thu Jan 15 16:33:16 2015 From: adahary at gmail.com (Assaf Dahary) Date: Thu, 15 Jan 2015 15:33:16 +0200 Subject: [Freeswitch-users] mobile to mobile over 3g/4g Message-ID: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> Hi, I'm currently using bypass_media to make two mobile call each directly (RTP bypass FS) over the same wifi network. Once I detect that caller and callee are sharing the same WiFi network then I switch to bypass_media in dialplan (works well in big complex buildings). I would like also to use the same bypass_media mode when mobiles connect over 3g/4g data mobile network. I fully understand that both mobiles should be connected to the same mobile operator and should share a common subnet routing (without NAT) for them to 'see' each other private IP addr. Is it possible for two mobiles on the same mobile 3g/4g network to ping each other using their private IP addr? Should mobile operators allow this kind of direct peer connections (I couldn't ping internally with my Orange mobile operator)? Did anyone successfully make a direct bypass_media call between two sip mobiles over 3g/4g network? Regards assaf -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/9c3eca6e/attachment.html From adam.ben.ayoun1 at gmail.com Thu Jan 15 16:27:15 2015 From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun) Date: Thu, 15 Jan 2015 15:27:15 +0200 Subject: [Freeswitch-users] SIP over Websocket VS SIP over TCP In-Reply-To: References: Message-ID: In our case, all clients are calling to the conference on FreeSwitch, hence, no registration is required at the moment, but it definitely opens up other options for future. On 10 January 2015 at 20:38, Adam Ben-Ayoun wrote: > Great to hear that. Thanks again. > > On 10 January 2015 at 20:35, Carlos Ruiz D?az > wrote: > >> WebRTC doesn't specify a signalling protocol. This means that you can use >> SIP over any transport you want to carry the webRTC enabled SDP. >> >> FS will receive the SDP, detect that has a RTP/SAVPF profile and start >> handling it accordingly. >> >> Take for example Jitsi or IMSDroid, they both support webRTC and do SIP >> over UDP/TCP/TLS. >> >> Regards, >> Carlos >> On Jan 10, 2015 12:08 PM, "Adam Ben-Ayoun" >> wrote: >> >>> Thanks Anthony. I assume that means I can use SIP over TCP/TLS for >>> signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work >>> when communicating with FS (when stuff like fingerprint, etc)? >>> >>> On 10 January 2015 at 19:49, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> The WebRTC media engine is driven completely by the SDP, the transport >>>> will not make any difference. >>>> >>>> >>>> On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun < >>>> adam.ben.ayoun1 at gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> We are developing a mobile client that will use the WebRTC media stack >>>>> and Freeswitch as an MCU (only for conference calls). My question is, since >>>>> we build a native app, can we use SIP over TCP for signalling? In other >>>>> words, if Freeswitch receives the WebRTC kind of SDP, will it be able to >>>>> communicate in the same way as if we were using the SIP over Websocket (the >>>>> other Freeswitch option)? Any corner cases/considerations with this? Our >>>>> goal is to avoid implementing SIP over Websocket on the client as much as >>>>> possible. >>>>> >>>>> Thanks, >>>>> Adam >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>> >>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>> http://twitter.com/FreeSWITCH >>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> _________________________________________________________________________ >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/8d5ff6ab/attachment-0001.html From luis.daniel.lucio at gmail.com Thu Jan 15 17:16:34 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Thu, 15 Jan 2015 09:16:34 -0500 Subject: [Freeswitch-users] mobile to mobile over 3g/4g In-Reply-To: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> References: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> Message-ID: 2015-01-15 8:33 GMT-05:00 Assaf Dahary : > Is it possible for two mobiles on the same mobile 3g/4g network to ping each > other using their private IP addr? This is not a voip related question. You should try that on your mobile company. >Should mobile operators allow this kind of direct peer connections (I couldn't ping internally with my Orange mobile operator)? Again, that not voip, this is networking related. Each carrier can do with their network as they want. >Did anyone successfully make a direct bypass_media call between two sip mobiles over 3g/4g network? In ROGERS (Canada), you can because they dont do FW and they assign public IP to mobiles. Again, it depends on the network rules of your operator, nothing to do with the voip. You may consider not using by pass for IPs different than your LAN profile. 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 From juanito1982 at gmail.com Thu Jan 15 19:10:31 2015 From: juanito1982 at gmail.com (=?UTF-8?Q?Juan_Antonio_Iba=C3=B1ez_Santorum?=) Date: Thu, 15 Jan 2015 17:10:31 +0100 Subject: [Freeswitch-users] Playing with ACL and authenticate users In-Reply-To: References: <143FDEBB-14E2-4A52-A884-6DFD4D950311@gmail.com> <07BF4904977CC645B485E970424193AD130923A604@localhost> Message-ID: Hello, I would like to know if there were any news in this subject. Is possible to mix both IP auth users and user/password users in the same profile without the need of using respond application? Regards 2013-06-04 18:55 GMT+02:00 Michael Collins : > > > > On Tue, Jun 4, 2013 at 3:32 AM, wrote: > >> In short!? >> >> Yes... >> >> You can't use ip-auth and password-auth in the same profile >> >> regards >> > > Pro Tip: you *can* have both on the same profile. Brian West said that > you'd need to make use of the 'respond' dialplan app for those who are not > IP auth'd. Use respond with data='401' and that should force the phone to > do the normal digest auth dialog. > -MC > > >> >> ___________________________ >> >> And now I'm lost. Please, can anybody point me where the issue can be? >> Do I need to have different sofia profiles, one for IP authentication and >> one for username/password auth? >> >> Regards, >> Alfonso. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.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 >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/ec21d848/attachment.html From steveayre at gmail.com Thu Jan 15 19:29:41 2015 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 15 Jan 2015 16:29:41 +0000 Subject: [Freeswitch-users] mobile to mobile over 3g/4g In-Reply-To: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> References: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> Message-ID: I would expect that while it might work on some operators you won't be able to rely on it working for all mobile networks. If your users aren't under your control and therefore aren't on networks you've had an opportunity to test this on then you'll probably see many calls with no audio. On 15 January 2015 at 13:33, Assaf Dahary wrote: > Hi, > > > > I'm currently using bypass_media to make two mobile call each directly > (RTP bypass FS) over the same wifi network. > > Once I detect that caller and callee are sharing the same WiFi network > then I switch to bypass_media in dialplan (works well in big complex > buildings). > > > > I would like also to use the same bypass_media mode when mobiles connect > over 3g/4g data mobile network. > > I fully understand that both mobiles should be connected to the same > mobile operator and should share a common subnet routing (without NAT) for > them to 'see' each other private IP addr. > > > > Is it possible for two mobiles on the same mobile 3g/4g network to ping > each other using their private IP addr? > > > > Should mobile operators allow this kind of direct peer connections (I > couldn't ping internally with my Orange mobile operator)? > > > > Did anyone successfully make a direct bypass_media call between two sip > mobiles over 3g/4g network? > > > > Regards > > > > assaf > > > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/dafd2a6a/attachment.html From adahary at gmail.com Thu Jan 15 19:47:22 2015 From: adahary at gmail.com (Assaf Dahary) Date: Thu, 15 Jan 2015 18:47:22 +0200 Subject: [Freeswitch-users] mobile to mobile over 3g/4g In-Reply-To: References: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> Message-ID: <779edd43-435f-4d7a-afdf-ced2118a6277@email.bluemailapp.com> It seams that some mobile operators allow internal direct connectivity and some just block it. Direct connectivity will increase call quality and drop call latency. I should start mapping mobile operators by trying first bypass media on internal calls and if fail then to cancel it and call via FS . On Jan 15, 2015, 18:32, at 18:32, Steven Ayre wrote: >I would expect that while it might work on some operators you won't be >able >to rely on it working for all mobile networks. > >If your users aren't under your control and therefore aren't on >networks >you've had an opportunity to test this on then you'll probably see many >calls with no audio. > > > >On 15 January 2015 at 13:33, Assaf Dahary wrote: > >> Hi, >> >> >> >> I'm currently using bypass_media to make two mobile call each >directly >> (RTP bypass FS) over the same wifi network. >> >> Once I detect that caller and callee are sharing the same WiFi >network >> then I switch to bypass_media in dialplan (works well in big complex >> buildings). >> >> >> >> I would like also to use the same bypass_media mode when mobiles >connect >> over 3g/4g data mobile network. >> >> I fully understand that both mobiles should be connected to the same >> mobile operator and should share a common subnet routing (without >NAT) for >> them to 'see' each other private IP addr. >> >> >> >> Is it possible for two mobiles on the same mobile 3g/4g network to >ping >> each other using their private IP addr? >> >> >> >> Should mobile operators allow this kind of direct peer connections (I >> couldn't ping internally with my Orange mobile operator)? >> >> >> >> Did anyone successfully make a direct bypass_media call between two >sip >> mobiles over 3g/4g network? >> >> >> >> Regards >> >> >> >> assaf >> >> >> >> >> >> >_________________________________________________________________________ >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/695caeee/attachment-0001.html From brian at freeswitch.org Thu Jan 15 19:52:25 2015 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2015 10:52:25 -0600 Subject: [Freeswitch-users] how can i keep A-LEG do not hangup In-Reply-To: <001201d030c6$f5b5e1e0$e121a5a0$@comcast.net> References: <001201d030c6$f5b5e1e0$e121a5a0$@comcast.net> Message-ID: Thats not relevant in this case, The autoHangup is when used in say lua... This case you will probably want this combo: https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket#mod_event_socket-linger https://wiki.freeswitch.org/wiki/Variable_park_after_bridge On Thu, Jan 15, 2015 at 7:27 AM, Andrew wrote: > This may be able to help you but I set session.SetAutoHangup(false); > > This will stop the dial plan from ending. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *ax lyb > *Sent:* Thursday, January 15, 2015 2:52 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] how can i keep A-LEG do not hangup > > > > All: > > recently i write a small program based freeswitch, in this program i > only do things like callcenter mod, > > here is what i do: > > > > 1. originate a call (? originate sofia/gateway/gw01/xxxx my-out-call > XML default?), in the default.xml, > > i already config it as follow: > > > > > > 65 "^my-call-out$"> > > 66 > > 67 > > 72 > > 73 > > > > 2. when call-out-destination customer pick up the call, i control the > flow to playback a wav file, then , > > bridge the call to a free agent while the wav file play over in > CHANNEL_EXECUTE_COMPLETE, > > these steps run normally ok. > > > > 3. the question is : if i have two agent in one moment, and this time i > have 3 or > 3 out-call to bridge, > > i?ll leave the other call do nothing,for i can not get more free > agents to service; some special scene > > is i have two agents (A1,A2) service 3 call (C1,C2,C3), A1 service > C1, A2 service C2,then C3 is only > > A-LEG when play over a wav file,about 100 seconds later, C3 is > hangup(it?s not user hangup manually), > > how can i do make the C3 call not hangup until i get a free agent to > service it? > > > > > > any suggest is appreciate, > > > > > > > > ax.lyb.lei at gmail.com > > > > _________________________________________________________________________ > 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/20150115/cca1f53e/attachment.html From david.villasmil at gmail.com Thu Jan 15 20:04:31 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Thu, 15 Jan 2015 18:04:31 +0100 Subject: [Freeswitch-users] Playing with ACL and authenticate users In-Reply-To: References: <143FDEBB-14E2-4A52-A884-6DFD4D950311@gmail.com> <07BF4904977CC645B485E970424193AD130923A604@localhost> Message-ID: I think if you have acl auth set, and if fails, it will fallback to digest auth... not sure though. Try it. On Jan 15, 2015 5:12 PM, "Juan Antonio Iba?ez Santorum" < juanito1982 at gmail.com> wrote: > Hello, > > I would like to know if there were any news in this subject. Is > possible to mix both IP auth users and user/password users in the same > profile without the need of using respond application? > > Regards > > 2013-06-04 18:55 GMT+02:00 Michael Collins : > >> >> >> >> On Tue, Jun 4, 2013 at 3:32 AM, wrote: >> >>> In short!? >>> >>> Yes... >>> >>> You can't use ip-auth and password-auth in the same profile >>> >>> regards >>> >> >> Pro Tip: you *can* have both on the same profile. Brian West said that >> you'd need to make use of the 'respond' dialplan app for those who are not >> IP auth'd. Use respond with data='401' and that should force the phone to >> do the normal digest auth dialog. >> -MC >> >> >>> >>> ___________________________ >>> >>> And now I'm lost. Please, can anybody point me where the issue can be? >>> Do I need to have different sofia profiles, one for IP authentication >>> and one for username/password auth? >>> >>> Regards, >>> Alfonso. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.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 >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.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 >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/495c74d6/attachment-0001.html From GeorgePhelps at gfphelps.com Thu Jan 15 20:08:41 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Thu, 15 Jan 2015 12:08:41 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <147601d02f32$37a9f5a0$a6 fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> <007901d02ff5$67075090$3515f1b0$@gfphelps.com> Message-ID: <041401d030e5$ea3b6730$beb23590$@gfphelps.com> Michael Collins, I had previously tried the ?Enterprise Originate? and syntax, but it did not make any difference at the time. I retested today with that syntax, and I am still seeing the same problem. If that is the recommended configuration for my situation, I will kept that syntax in my dialplan. I also tested with a difference VoIP service provider. Better results in that the INVITE timer now runs for 28 seconds, as opposed to just 10 seconds with the previous VoIP service provider. During the 28 seconds ? with both extensions ringing ? I was able to answer one extension. However, the call disconnected just as soon as I answered it, and then immediately started ringing again. This continued until the 28 seconds ran out. I will look into this later, and gather additional logs. I also retested, with the new VoIP service provider ? and dialing just one extension works fine. Dialing just one extension also worked with the previous VoIP service provider, even with the 10 second INVITE timer, as long as I answered within the 10 seconds window. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 10:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Glad you have the book. On page 19 it covers the use of enterprise originate. I think possibly you need to use the method as discussed on page 21. Try something like this: Could just be that early media is not being ignored on both user dialout attempts. -MC On Wed, Jan 14, 2015 at 4:27 AM, George F. Phelps wrote: Michael Collins, I already have the book. Thanks! Here?s my dialplan: New log file uploaded to: http://pastebin.com/gnEpPzk9 To me, the most significant event in the log file is the SIP CANCEL message ? starting at line #321: tport.c:3023 tport_deliver() tport_deliver(0x95daa0): msg 0xad8fb0 (437 bytes) from udp/169.XX.XX.XX:5080/sip next=(nil) nta.c:2880 agent_recv_request() nta: received CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (CSeq 1) nta.c:3026 agent_recv_request() nta: CANCEL (1) is going to INVITE (1) I don?t think it?s related, but I am also curious about log file line #285: sres.c:2987 sres_query_report_error() sres(q=0x98b050): reporting error NAME_ERR for SRV _sip._udp.sip.switch2voip.us Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 2:41 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? We covered this nicely in chapter 1 of the FreeSWITCH Cookbook I'm sorry that I'm late to the party so I am missing some information. Can you pastebin not only the call log but also the dialplan code for the example in question? One other tip: it appears that the log that you are pasting is coming directly from the FreeSWITCH console. By default the console does not have debug level output enabled. Try entering the command "console loglevel debug" and you'll see way more log lines, mostly yellow text. Those lines will most likely contain the clues needed to unravel this mystery. Thanks, MC On Tue, Jan 13, 2015 at 2:42 PM, George F. Phelps wrote: New logfile uploaded to: http://pastebin.com/CFFvVarS The log contains default Freeswitch console log messages, plus a SIP trace of a failed call. BTW, both extensions were ringing ? prior to the CANCEL message (see context below). In the log I see the INVITE from my VoIP service provider: recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: ------------------------------------------------------------------------ INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: ------------------------------------------------------------------------ SIP/2.0 100 Trying (Then, subsequent INVITE messages to my two extensions. But other no messages to/from my VoIP service provider.) And then, a spontaneous CANCEL from my VoIP service provider, approximately 10 seconds after the initial INVITE message. Due to a SIP ?Timer B? timeout? Seems way too short. recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: ------------------------------------------------------------------------ CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (Freeswitch cleanup of SIP sessions to my extensions?) Bote Man--> I have two local extensions. Individually, the extensions can make and receive both internal and external calls. It?s only the simultaneous ringing for external, inbound calls that is not working at the moment. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Tuesday, January 13, 2015 2:33 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I suggest you configure and register 3 total local phones to your FS installation, configure 2 of them as the target of your simultaneous ring group, and call them with the 3rd phone. Until you can get that working, calling through a carrier is adding another layer of complexity to the problem and confusing the issue. Out of the box FreeSWITCH does not utter voice codes, they must be coming from your carrier. Also, the debug-level logs very likely tell you exactly what is happening, even though they can be staggering to decipher as a newcomer to FS. Learning how to read them pays off in so many ways, though. I find the color-coded logs on the console or viewed via FS_cli to be helpful in these instances. Bote From: George F. Phelps Sent: Tuesday, 13 January, 2015 08:10 Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I tried? ?but that did not resolve the problem. 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: Tuesday, January 13, 2015 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" wrote: David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: Thanks, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/0c3681d4/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/20150115/0c3681d4/attachment-0001.bin From ahabiba at gmail.com Thu Jan 15 20:26:23 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Thu, 15 Jan 2015 20:26:23 +0300 Subject: [Freeswitch-users] Security Issue In-Reply-To: References: Message-ID: <280CF4B1-555B-4DCA-B9E9-774177940DCA@gmail.com> Thank you really Michael,David and Brian, I did a simple change to the external sip profile which resolved the issue from my point of view. what I did is I add the below line to the external sip profile, which inform it to valid any request from external system against ACL list. From: Michael Collins > To: FreeSWITCH Users Help > Date: January 15, 2015 at 7:19:07 AM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue On Wed, Jan 14, 2015 at 9:40 AM, Ahmed Habiba > wrote: Thank you really David, Here is my point, the sip-trace in the first mail shows that, the call comes to public context mainly through port 5080, and however the originator IP was not defined in my ACL list Freeswitch continue to process the call for some reason. Just an FYI, the external profile does not have auth-calls param set to true, so FS simply tries to route the call in the public context without sending back an auth challenge. Since the public context is pretty paranoid it's not exactly easy to dial out. Also, just because FS tries to route the call does not mean that FS considers the call to be "authenticated." If you want all traffic coming in to your server to be authenticated then either send it all to the internal profile (i.e. port 5060) or add auth-calls to your external profile. The bigger question you may want to ask is: why are these random IP even getting to your server? Do you allow public access to your system? If so, why? If not, then you need a firewall (iptables or whatnot) to block those SIP messages from ever getting to your FreeSWITCH. You may also be interested in something like fail2ban and voipbl.org . -MC even if it come to 5060, I was expecting some request for digest authentication, which is not shown in the log. From: David Villasmil Govea > To: FreeSWITCH Users Help > Date: January 14, 2015 at 8:30:35 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Security Issue Authorization is done if you configure your sip profile to do it. By default 5060 (internal) requires authentication, 5080 (external) doesn't but it does use the ACL to allow or not calls. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/863df20d/attachment.html From juanito1982 at gmail.com Thu Jan 15 20:35:30 2015 From: juanito1982 at gmail.com (=?UTF-8?Q?Juan_Antonio_Iba=C3=B1ez_Santorum?=) Date: Thu, 15 Jan 2015 18:35:30 +0100 Subject: [Freeswitch-users] Playing with ACL and authenticate users In-Reply-To: References: <143FDEBB-14E2-4A52-A884-6DFD4D950311@gmail.com> <07BF4904977CC645B485E970424193AD130923A604@localhost> Message-ID: I tried to set auth-acl parameter to the user definition but FS still answer with 407 Auth required Regards 2015-01-15 18:04 GMT+01:00 David Villasmil Govea : > I think if you have acl auth set, and if fails, it will fallback to digest > auth... not sure though. Try it. > On Jan 15, 2015 5:12 PM, "Juan Antonio Iba?ez Santorum" < > juanito1982 at gmail.com> wrote: > >> Hello, >> >> I would like to know if there were any news in this subject. Is >> possible to mix both IP auth users and user/password users in the same >> profile without the need of using respond application? >> >> Regards >> >> 2013-06-04 18:55 GMT+02:00 Michael Collins : >> >>> >>> >>> >>> On Tue, Jun 4, 2013 at 3:32 AM, wrote: >>> >>>> In short!? >>>> >>>> Yes... >>>> >>>> You can't use ip-auth and password-auth in the same profile >>>> >>>> regards >>>> >>> >>> Pro Tip: you *can* have both on the same profile. Brian West said that >>> you'd need to make use of the 'respond' dialplan app for those who are not >>> IP auth'd. Use respond with data='401' and that should force the phone to >>> do the normal digest auth dialog. >>> -MC >>> >>> >>>> >>>> ___________________________ >>>> >>>> And now I'm lost. Please, can anybody point me where the issue can be? >>>> Do I need to have different sofia profiles, one for IP authentication >>>> and one for username/password auth? >>>> >>>> Regards, >>>> Alfonso. >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.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 >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.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 >>>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.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 >> > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/8b88a7cb/attachment-0001.html From regis.freeswitch.org at tornad.net Thu Jan 15 21:53:18 2015 From: regis.freeswitch.org at tornad.net (Regis M) Date: Thu, 15 Jan 2015 19:53:18 +0100 Subject: [Freeswitch-users] Playing with ACL and authenticate users In-Reply-To: References: <143FDEBB-14E2-4A52-A884-6DFD4D950311@gmail.com> <07BF4904977CC645B485E970424193AD130923A604@localhost> Message-ID: +1 to have a clean and simple both auth method sample config... don't have time to test yet :) Regards, 2015-01-15 18:35 GMT+01:00 Juan Antonio Iba?ez Santorum < juanito1982 at gmail.com>: > I tried to set auth-acl parameter to the user definition but FS still > answer with 407 Auth required > > Regards > > 2015-01-15 18:04 GMT+01:00 David Villasmil Govea < > david.villasmil at gmail.com>: > >> I think if you have acl auth set, and if fails, it will fallback to >> digest auth... not sure though. Try it. >> On Jan 15, 2015 5:12 PM, "Juan Antonio Iba?ez Santorum" < >> juanito1982 at gmail.com> wrote: >> >>> Hello, >>> >>> I would like to know if there were any news in this subject. Is >>> possible to mix both IP auth users and user/password users in the same >>> profile without the need of using respond application? >>> >>> Regards >>> >>> 2013-06-04 18:55 GMT+02:00 Michael Collins : >>> >>>> >>>> >>>> >>>> On Tue, Jun 4, 2013 at 3:32 AM, wrote: >>>> >>>>> In short!? >>>>> >>>>> Yes... >>>>> >>>>> You can't use ip-auth and password-auth in the same profile >>>>> >>>>> regards >>>>> >>>> >>>> Pro Tip: you *can* have both on the same profile. Brian West said that >>>> you'd need to make use of the 'respond' dialplan app for those who are not >>>> IP auth'd. Use respond with data='401' and that should force the phone to >>>> do the normal digest auth dialog. >>>> -MC >>>> >>>> >>>>> >>>>> ___________________________ >>>>> >>>>> And now I'm lost. Please, can anybody point me where the issue can be? >>>>> Do I need to have different sofia profiles, one for IP authentication >>>>> and one for username/password auth? >>>>> >>>>> Regards, >>>>> Alfonso. >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.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 >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.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 >>> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/833a9d4b/attachment.html From alipey at gmail.com Thu Jan 15 22:05:08 2015 From: alipey at gmail.com (Ali Pey) Date: Thu, 15 Jan 2015 14:05:08 -0500 Subject: [Freeswitch-users] leg_timeout and group_confirm_cancel_timeout Message-ID: Hello, If I am using group confirm and have a number of destinations to ring and have group_confirm_cancel_timeout set, what happens if my first destination answers the phone but does not press any key to confirm the call. Will the call ever go to the second destination? In another word, when using group confirm, a call passes through three phases: 1- Call is ringing 2- Call is answered, waiting to be confirmed 3- Call is confirmed and bridged Is there any timeout setting for Phase two? Would bridge continue to the next destination if phase 2 fails? Thanks and regards, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/6cc3c8bf/attachment.html From rdsmallwood928 at gmail.com Thu Jan 15 23:36:53 2015 From: rdsmallwood928 at gmail.com (Robert Smallwood) Date: Thu, 15 Jan 2015 13:36:53 -0700 Subject: [Freeswitch-users] Sound delays when using WebRTC Message-ID: Hi All, I'm currently using a browser (chrome) to make calls via webrtc to freeswitch (1.5.15b+git~20141219T193610Z~35cb0ad286~64bit). However, I am experiencing delays when hearing our initial prompting message. I'd like to hook onto some event that lets me know that all of session.streamFile() will be heard and not cut off. We have a dial plan, and in that dial plan (js file) we do this sequence of events... (psuedo code) if(session.ready()) session.answer() ...some reading of headers... session.streamFile("Our welcome prompt", func return true) On the client side I can watch the logs and notice that I hear media as soon as this log comes through: | jssip.rtcsession.rtcmediahandler | ICE connection state changed to "connected" However, this can be a second or two behind the streamFile call in the dial-plan. What I would like to be able to do is hook into the ICE state change in the dial-plan file and wait until I get it to start streaming the file. I have tried playing around with some of the methods found here: https://wiki.freeswitch.org/wiki/Session, namely session.mediaReady, session.waitForMedia, and checking the session.state to see if any of them change when I get the ice state message client side to no avail. Is there a way to accomplish this (Listening for ICE state changes in a dial plan)? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/c6f94d8f/attachment.html From anthony.minessale at gmail.com Thu Jan 15 23:57:41 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 15 Jan 2015 14:57:41 -0600 Subject: [Freeswitch-users] Sound delays when using WebRTC In-Reply-To: References: Message-ID: First you should try updating to latest master. Basically if you are doing webrtc development you need to constantly update to keep up with the moving target. If you still notice its clipping a bit you can set the global variable answer_delay=2500 in vars.xml to pad the beginning of the call. There is not hooks for dialplan to know about ice, its abstract. however in newer versions the act of answering the call will cause the core to read constantly until at least the ice is negotiated. This is more of a side-effect of not caring about normal telephony expectations on the part of the webrtc implementors than anything else. Its an inherent side-effect of ice and dtls to cause call setup errors. On Thu, Jan 15, 2015 at 2:36 PM, Robert Smallwood wrote: > Hi All, > > I'm currently using a browser (chrome) to make calls via webrtc to > freeswitch (1.5.15b+git~20141219T193610Z~35cb0ad286~64bit). However, I am > experiencing delays when hearing our initial prompting message. I'd like > to hook onto some event that lets me know that all of session.streamFile() > will be heard and not cut off. > > We have a dial plan, and in that dial plan (js file) we do this sequence > of events... (psuedo code) > > if(session.ready()) > session.answer() > ...some reading of headers... > session.streamFile("Our welcome prompt", func return true) > > On the client side I can watch the logs and notice that I hear media as > soon as this log comes through: > > | jssip.rtcsession.rtcmediahandler | ICE connection state changed to > "connected" > > However, this can be a second or two behind the streamFile call in the > dial-plan. > > What I would like to be able to do is hook into the ICE state change in > the dial-plan file and wait until I get it to start streaming the file. > > I have tried playing around with some of the methods found here: > https://wiki.freeswitch.org/wiki/Session, namely session.mediaReady, > session.waitForMedia, and checking the session.state to see if any of them > change when I get the ice state message client side to no avail. > > Is there a way to accomplish this (Listening for ICE state changes in a > dial plan)? > > _________________________________________________________________________ > 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/d71beb5d/attachment-0001.html From nneul at mst.edu Fri Jan 16 00:05:36 2015 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 15 Jan 2015 15:05:36 -0600 Subject: [Freeswitch-users] any handset recommendations that operate like polycom+cisco? Message-ID: <54B82BA0.20201@mst.edu> Looking for any handset/desk phone recommendations that have the "every line key can be it's own SIP account, even on expansion modules" behavior. Yealink units for example only allow a set number of lines, and the rest don't operate as line keys, so even though you can have a 30 + button phone, it's not usable for a receptionist or secretary that may handle calls for 15 different people with their own direct lines. With the polycom and cisco (sccp) phones, you can define N different SIP accounts. Are there any other phones out there that operate this way that anyone can recommend? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From david.villasmil at gmail.com Fri Jan 16 00:08:29 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Thu, 15 Jan 2015 22:08:29 +0100 Subject: [Freeswitch-users] any handset recommendations that operate like polycom+cisco? In-Reply-To: <54B82BA0.20201@mst.edu> References: <54B82BA0.20201@mst.edu> Message-ID: Linksys/cisco On Jan 15, 2015 10:06 PM, "Nathan Neulinger" wrote: > Looking for any handset/desk phone recommendations that have the "every > line key can be it's own SIP account, even on > expansion modules" behavior. > > Yealink units for example only allow a set number of lines, and the rest > don't operate as line keys, so even though you > can have a 30 + button phone, it's not usable for a receptionist or > secretary that may handle calls for 15 different > people with their own direct lines. > > With the polycom and cisco (sccp) phones, you can define N different SIP > accounts. > > Are there any other phones out there that operate this way that anyone can > recommend? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/be034582/attachment.html From ignaciodurli at cinchcast.com Fri Jan 16 01:42:26 2015 From: ignaciodurli at cinchcast.com (Ignacio Durli) Date: Thu, 15 Jan 2015 22:42:26 +0000 Subject: [Freeswitch-users] Make conference play command async based on when prompt finishes Message-ID: Hello, Our application interacts with freeswitch both using api commands and event socket library for receiving events in an IVR application. One of our requirements is to continue with IVR flow once a prompt has finished. Prompts are being played using the "conference play [|]". We found that FS response comes immediately after the play starts. Although we found that an event arrives when the prompt ends, we don't find convenient to have that resolved on the code that is receiving the events and would like to have something that returns once the prompt is finished. Is there any way to susbscribe an api call in a way that returns once a specific event happens? What's the difference between using or not the "async" param in the "conference play" api call? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/1e6130fc/attachment.html From notify.sina at gmail.com Fri Jan 16 01:48:23 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Thu, 15 Jan 2015 22:48:23 +0000 Subject: [Freeswitch-users] Help Needed Debugging Lua Script References: <0000014ac4a0ec5e-81630ae2-7f78-4b3a-8659-6ba76a3c9b96-000000@email.amazonses.com> Message-ID: Hi List, I think I have finally come up with something that works, the script captures the details correctly, but I wonder if there is a better way to write it? I am trying to get it to call the dialer back and play a message, which it does correctly. Please what can I do to make it to retry the dialler's number in case the call doesn't go through the first (say) two times, just in case the dialler cut it off mistakenly, or the telco plays a repeated message after freeswitch hung up, preventing the callback from coming through? (I've seen this happen a few times with a local telco). Thanks! number_to_call = argv[1]; caller_id = argv[2]; api = freeswitch.API(); dialString = "{origination_caller_id_name="..caller_id..",origination_caller_id_number="..caller_id.."}sofia/gateway/sipgw/"..number_to_call..""; session1 = freeswitch.Session(dialString); session1:sleep(5000); if (session1:ready()) then session1:sleep(35000); api:execute("bgapi originate", "session1"); session1:sleep(3000); session1:streamFile('/tmp/stop_calling_me_stalker.wav'); session1:hangup("NORMAL_CLEARING"); end On Wed Jan 07 2015 at 2:45:04 PM Avi Marcus wrote: > Two things: > 1) You aren't grabbing the arg, but the channel variable.. try this in > your script: > caller_id_number = argv[1] > number_to_call = argv[2] > > 2) I don't think you're managing your hangup/callback originate properly. > I don't think you want to use bgapi... or maybe you just need a > destination. It's "originate > sofia/A > endpoint" -- you need to specify where it goes to, the lua script can't > "receive" the call. You can have it received by e.g: &lua(pickup.lua) > > api = freeswitch.API() > api:execute("originate", DialString.." &lua(pickup.lua)"); > > Also: > Maybe you want to use it as a hangup hook. Instead of: > > Do: > > > > -Avi > > On Wed, Jan 7, 2015 at 1:52 AM, Sina Owolabi > wrote: > >> Hi List, >> >> FreeSWITCH newbie here again. >> I am trying to cobble togther a lua callback script, my first attempt >> was successful, but I am trying to make it slightly more elegant. >> I don't see any errors when I try to run this but the callback isnt >> happening. >> This is my very second attempt trying to write in lua, so I would be >> very grateful for any help. >> >> The user is expected to dial in, have the call hangup and FreeSWITCH call >> back. >> >> I'm passing a modified $effective_caller_id_number and >> $destination_number to the lua script: >> >> >> > expression="^1(\d{10})$"require-nested="false"> >> > data="effective_caller_id_number=+234${1}"/> >> > data="effective_caller_id_name=+234${1}"/> >> >> >> > data="destination_number=+12312345${1}${2}" /> >> >> >> >> >> >> >> The script itself: >> >> api = freeswitch.API(); >> call_string = "bagpi originate >> >> {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" >> >> freeswitch.msleep(5000); >> if (session:ready()) then >> caller_id_number = session:getVariable("destination_number"); >> caller_id_name = session:getVariable("destination_number"); >> number_to_call = >> session:getVariable("effective_caller_id_number"); >> >> api:executeString(call_string); >> freeswitch.msleep(2000); >> session:streamFile("/tmp/get_off_my_lawn.wav"); >> session:hangup("NORMAL_CLEARING"); >> end >> >> _________________________________________________________________________ >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/5fd1f56b/attachment-0001.html From GeorgePhelps at gfphelps.com Fri Jan 16 02:51:16 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Thu, 15 Jan 2015 18:51:16 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <041401d030e5$ea3b6730$beb23590$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <14 7601d02f32$37a9f5a0$a 6 fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> <007901d02ff5$67075090$3515f1b0$@gfphelps.com> <041401d030e5$ea3b6730$beb23590$@gfphelps.com> Message-ID: <04ce01d0311e$272fac40$758f04c0$@gfphelps.com> New failed call log at: http://pastebin.com/cErTyGht Scenario? Two registered extensions, 1000 and 1001. Inbound call, to simultaneously ring both extensions. Both extensions start ringing. I answer on extension 1001. The call immediately drops, and then begins ringing again in a couple of seconds. >From the log? Line #754: extension 1001 OKs its INVITE. Line #835: debug messages indicates extension 1001 answered the call. Line #854: FreeSWITCH CANCELs the call to extension 1000. Line #887: FreeSWITCH terminates the inbound call with a ?480 Temporarily Unavailable? response. Line #930: FreeSWITCH sends a BYE to extension 1001. Line #1117: a new (regenerated), inbound call request. Without simultaneous ring enabled to two extensions, i.e., ringing only extension 1001 ? the call is handled with no problems. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Thursday, January 15, 2015 12:09 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Michael Collins, I had previously tried the ?Enterprise Originate? and syntax, but it did not make any difference at the time. I retested today with that syntax, and I am still seeing the same problem. If that is the recommended configuration for my situation, I will kept that syntax in my dialplan. I also tested with a difference VoIP service provider. Better results in that the INVITE timer now runs for 28 seconds, as opposed to just 10 seconds with the previous VoIP service provider. During the 28 seconds ? with both extensions ringing ? I was able to answer one extension. However, the call disconnected just as soon as I answered it, and then immediately started ringing again. This continued until the 28 seconds ran out. I will look into this later, and gather additional logs. I also retested, with the new VoIP service provider ? and dialing just one extension works fine. Dialing just one extension also worked with the previous VoIP service provider, even with the 10 second INVITE timer, as long as I answered within the 10 seconds window. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 10:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Glad you have the book. On page 19 it covers the use of enterprise originate. I think possibly you need to use the method as discussed on page 21. Try something like this: Could just be that early media is not being ignored on both user dialout attempts. -MC On Wed, Jan 14, 2015 at 4:27 AM, George F. Phelps wrote: Michael Collins, I already have the book. Thanks! Here?s my dialplan: New log file uploaded to: http://pastebin.com/gnEpPzk9 To me, the most significant event in the log file is the SIP CANCEL message ? starting at line #321: tport.c:3023 tport_deliver() tport_deliver(0x95daa0): msg 0xad8fb0 (437 bytes) from udp/169.XX.XX.XX:5080/sip next=(nil) nta.c:2880 agent_recv_request() nta: received CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (CSeq 1) nta.c:3026 agent_recv_request() nta: CANCEL (1) is going to INVITE (1) I don?t think it?s related, but I am also curious about log file line #285: sres.c:2987 sres_query_report_error() sres(q=0x98b050): reporting error NAME_ERR for SRV _sip._udp.sip.switch2voip.us Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 2:41 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? We covered this nicely in chapter 1 of the FreeSWITCH Cookbook I'm sorry that I'm late to the party so I am missing some information. Can you pastebin not only the call log but also the dialplan code for the example in question? One other tip: it appears that the log that you are pasting is coming directly from the FreeSWITCH console. By default the console does not have debug level output enabled. Try entering the command "console loglevel debug" and you'll see way more log lines, mostly yellow text. Those lines will most likely contain the clues needed to unravel this mystery. Thanks, MC On Tue, Jan 13, 2015 at 2:42 PM, George F. Phelps wrote: New logfile uploaded to: http://pastebin.com/CFFvVarS The log contains default Freeswitch console log messages, plus a SIP trace of a failed call. BTW, both extensions were ringing ? prior to the CANCEL message (see context below). In the log I see the INVITE from my VoIP service provider: recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: ------------------------------------------------------------------------ INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: ------------------------------------------------------------------------ SIP/2.0 100 Trying (Then, subsequent INVITE messages to my two extensions. But other no messages to/from my VoIP service provider.) And then, a spontaneous CANCEL from my VoIP service provider, approximately 10 seconds after the initial INVITE message. Due to a SIP ?Timer B? timeout? Seems way too short. recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: ------------------------------------------------------------------------ CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (Freeswitch cleanup of SIP sessions to my extensions?) Bote Man--> I have two local extensions. Individually, the extensions can make and receive both internal and external calls. It?s only the simultaneous ringing for external, inbound calls that is not working at the moment. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Tuesday, January 13, 2015 2:33 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I suggest you configure and register 3 total local phones to your FS installation, configure 2 of them as the target of your simultaneous ring group, and call them with the 3rd phone. Until you can get that working, calling through a carrier is adding another layer of complexity to the problem and confusing the issue. Out of the box FreeSWITCH does not utter voice codes, they must be coming from your carrier. Also, the debug-level logs very likely tell you exactly what is happening, even though they can be staggering to decipher as a newcomer to FS. Learning how to read them pays off in so many ways, though. I find the color-coded logs on the console or viewed via FS_cli to be helpful in these instances. Bote From: George F. Phelps Sent: Tuesday, 13 January, 2015 08:10 Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I tried? ?but that did not resolve the problem. 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: Tuesday, January 13, 2015 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" wrote: David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: Thanks, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/6e8b87c6/attachment-0001.html From luis.daniel.lucio at gmail.com Fri Jan 16 03:13:15 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Thu, 15 Jan 2015 19:13:15 -0500 Subject: [Freeswitch-users] Help Needed Debugging Lua Script In-Reply-To: References: <0000014ac4a0ec5e-81630ae2-7f78-4b3a-8659-6ba76a3c9b96-000000@email.amazonses.com> Message-ID: Don't loose your time, contact me offline On Jan 15, 2015 5:49 PM, "Sina Owolabi" wrote: > Hi List, > > I think I have finally come up with something that works, the script > captures the details correctly, but I wonder if there is a better way to > write it? I am trying to get it to call the dialer back and play a message, > which it does correctly. > Please what can I do to make it to retry the dialler's number in case the > call doesn't go through the first (say) two times, just in case the dialler > cut it off mistakenly, or the telco plays a repeated message after > freeswitch hung up, preventing the callback from coming through? (I've seen > this happen a few times with a local telco). Thanks! > > number_to_call = argv[1]; > caller_id = argv[2]; > api = freeswitch.API(); > dialString = > "{origination_caller_id_name="..caller_id..",origination_caller_id_number="..caller_id.."}sofia/gateway/sipgw/"..number_to_call..""; > session1 = freeswitch.Session(dialString); > session1:sleep(5000); > > if (session1:ready()) then > session1:sleep(35000); > api:execute("bgapi originate", "session1"); > session1:sleep(3000); > session1:streamFile('/tmp/stop_calling_me_stalker.wav'); > session1:hangup("NORMAL_CLEARING"); > end > > > On Wed Jan 07 2015 at 2:45:04 PM Avi Marcus wrote: > >> Two things: >> 1) You aren't grabbing the arg, but the channel variable.. try this in >> your script: >> caller_id_number = argv[1] >> number_to_call = argv[2] >> >> 2) I don't think you're managing your hangup/callback originate properly. >> I don't think you want to use bgapi... or maybe you just need a >> destination. It's "originate >> sofia/A >> endpoint" -- you need to specify where it goes to, the lua script can't >> "receive" the call. You can have it received by e.g: &lua(pickup.lua) >> >> api = freeswitch.API() >> api:execute("originate", DialString.." &lua(pickup.lua)"); >> >> Also: >> Maybe you want to use it as a hangup hook. Instead of: >> >> Do: >> >> >> >> -Avi >> >> On Wed, Jan 7, 2015 at 1:52 AM, Sina Owolabi >> wrote: >> >>> Hi List, >>> >>> FreeSWITCH newbie here again. >>> I am trying to cobble togther a lua callback script, my first attempt >>> was successful, but I am trying to make it slightly more elegant. >>> I don't see any errors when I try to run this but the callback isnt >>> happening. >>> This is my very second attempt trying to write in lua, so I would be >>> very grateful for any help. >>> >>> The user is expected to dial in, have the call hangup and FreeSWITCH >>> call back. >>> >>> I'm passing a modified $effective_caller_id_number and >>> $destination_number to the lua script: >>> >>> >>> >> expression="^1(\d{10})$"require-nested="false"> >>> >> data="effective_caller_id_number=+234${1}"/> >>> >> data="effective_caller_id_name=+234${1}"/> >>> >>> >>> >> data="destination_number=+12312345${1}${2}" /> >>> >>> >>> >>> >>> >>> >>> The script itself: >>> >>> api = freeswitch.API(); >>> call_string = "bagpi originate >>> >>> {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" >>> >>> freeswitch.msleep(5000); >>> if (session:ready()) then >>> caller_id_number = session:getVariable("destination_number"); >>> caller_id_name = session:getVariable("destination_number"); >>> number_to_call = >>> session:getVariable("effective_caller_id_number"); >>> >>> api:executeString(call_string); >>> freeswitch.msleep(2000); >>> session:streamFile("/tmp/get_off_my_lawn.wav"); >>> session:hangup("NORMAL_CLEARING"); >>> end >>> >>> _________________________________________________________________________ >>> 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 > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/0cdc4423/attachment.html From luis.daniel.lucio at gmail.com Fri Jan 16 03:18:10 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Thu, 15 Jan 2015 19:18:10 -0500 Subject: [Freeswitch-users] how can i keep A-LEG do not hangup In-Reply-To: References: <001201d030c6$f5b5e1e0$e121a5a0$@comcast.net> Message-ID: Read about queues On Jan 15, 2015 11:53 AM, "Brian West" wrote: > Thats not relevant in this case, The autoHangup is when used in say lua... > > This case you will probably want this combo: > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket#mod_event_socket-linger > https://wiki.freeswitch.org/wiki/Variable_park_after_bridge > > > > > On Thu, Jan 15, 2015 at 7:27 AM, Andrew wrote: > >> This may be able to help you but I set session.SetAutoHangup(false); >> >> This will stop the dial plan from ending. >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *ax lyb >> *Sent:* Thursday, January 15, 2015 2:52 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] how can i keep A-LEG do not hangup >> >> >> >> All: >> >> recently i write a small program based freeswitch, in this program i >> only do things like callcenter mod, >> >> here is what i do: >> >> >> >> 1. originate a call (" originate sofia/gateway/gw01/xxxx my-out-call >> XML default"), in the default.xml, >> >> i already config it as follow: >> >> >> >> >> >> 65 > "^my-call-out$"> >> >> 66 >> >> 67 >> >> 72 >> >> 73 >> >> >> >> 2. when call-out-destination customer pick up the call, i control >> the flow to playback a wav file, then , >> >> bridge the call to a free agent while the wav file play over in >> CHANNEL_EXECUTE_COMPLETE, >> >> these steps run normally ok. >> >> >> >> 3. the question is : if i have two agent in one moment, and this time >> i have 3 or > 3 out-call to bridge, >> >> i'll leave the other call do nothing,for i can not get more free >> agents to service; some special scene >> >> is i have two agents (A1,A2) service 3 call (C1,C2,C3), A1 service >> C1, A2 service C2,then C3 is only >> >> A-LEG when play over a wav file,about 100 seconds later, C3 is >> hangup(it's not user hangup manually), >> >> how can i do make the C3 call not hangup until i get a free agent to >> service it? >> >> >> >> >> >> any suggest is appreciate, >> >> >> >> >> >> >> >> ax.lyb.lei at gmail.com >> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150115/d5a2cf82/attachment-0001.html From ax.lyb.lei at gmail.com Fri Jan 16 05:19:06 2015 From: ax.lyb.lei at gmail.com (ax lyb) Date: Fri, 16 Jan 2015 10:19:06 +0800 Subject: [Freeswitch-users] how can i keep A-LEG do not hangup (Luis Daniel Lucio Quiroz) In-Reply-To: References: Message-ID: hi, brian,Andrew i known a little in lua,i do my customer callcenter in the following style: 1. create a inbound connect to see what channel hangup (BaseFreeswitchClient is a socket-connection wrap) int connectFS() { //"auth ClueCon\n\n"; const char* AUTH_ME="ClueCon"; m_pFsClient = new BaseFreeswitchClient(reinterpret_cast(this)); if(m_pFsClient->ConnectFS(AUTH_ME) < 0 ) { delete m_pFsClient; m_pFsClient = NULL; return -1; } // const char* strFilter = "event plain CHANNEL_CREATE \ // CHANNEL_ANSWER \ // CHANNEL_BRIDGE \ // CHANNEL_HANGUP \ // CHANNEL_HANGUP_COMPLETE\n\n"; const char* strFilter = "event plain CHANNEL_HANGUP_COMPLETE\n\n"; if( m_pFsClient->filterEvent(strFilter) < 0 ) { delete m_pFsClient ; m_pFsClient = NULL; return -1; } return 0; } 2. create a server for outbound connect, the outbound connect for every call-session to FILTER CHANNEL_ANSWER,CHANNEL_HANGUP_COMPLETE,CHANNEL_EXECUTE_COMPLETE case ESL_EVENT_CHANNEL_DATA: { const char* strEvtName = evt.getHeader("Event-Name"); char szFilter[1024]; const char* strFilter = "event plain CHANNEL_ANSWER CHANNEL_BRIDGE CHANNEL_EXECUTE_COMPLETE DTMF\n\n"; snprintf(szFilter, sizeof(szFilter),"%sfilter Unique-ID %s\n\n",strFilter, evt.getHeader("Caller-Unique-ID")); int nWriteSize = strnlen(szFilter,sizeof(szFilter)); LOG(LS_ERROR)<onWrite(szFilter, nWriteSize); } 3. set "session.SetAutoHangup(false);?, which channel_variable should i set? thanks > ? 2015?1?16??08:18?freeswitch-users-request at lists.freeswitch.org ??? > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: Help Needed Debugging Lua Script (Luis Daniel Lucio Quiroz) > 2. Re: how can i keep A-LEG do not hangup (Luis Daniel Lucio Quiroz) > > ???: Luis Daniel Lucio Quiroz > ???: FreeSWITCH Users Help > ??: 2015?1?16? GMT+808:13:15 > ??????: FreeSWITCH Users Help > ??: ??? [Freeswitch-users] Help Needed Debugging Lua Script > > > Don't loose your time, contact me offline > > On Jan 15, 2015 5:49 PM, "Sina Owolabi" > wrote: > Hi List, > > I think I have finally come up with something that works, the script captures the details correctly, but I wonder if there is a better way to write it? I am trying to get it to call the dialer back and play a message, which it does correctly. > Please what can I do to make it to retry the dialler's number in case the call doesn't go through the first (say) two times, just in case the dialler cut it off mistakenly, or the telco plays a repeated message after freeswitch hung up, preventing the callback from coming through? (I've seen this happen a few times with a local telco). Thanks! > > number_to_call = argv[1]; > caller_id = argv[2]; > api = freeswitch.API(); > dialString = "{origination_caller_id_name="..caller_id..",origination_caller_id_number="..caller_id.."}sofia/gateway/sipgw/"..number_to_call..""; > session1 = freeswitch.Session(dialString); > session1:sleep(5000); > > if (session1:ready()) then > session1:sleep(35000); > api:execute("bgapi originate", "session1"); > session1:sleep(3000); > session1:streamFile('/tmp/stop_calling_me_stalker.wav'); > session1:hangup("NORMAL_CLEARING"); > end > > > On Wed Jan 07 2015 at 2:45:04 PM Avi Marcus > wrote: > Two things: > 1) You aren't grabbing the arg, but the channel variable.. try this in your script: > caller_id_number = argv[1] > number_to_call = argv[2] > > 2) I don't think you're managing your hangup/callback originate properly. > I don't think you want to use bgapi... or maybe you just need a destination. It's "originate sofia/A endpoint" -- you need to specify where it goes to, the lua script can't "receive" the call. You can have it received by e.g: &lua(pickup.lua) > > api = freeswitch.API() > api:execute("originate", DialString.." &lua(pickup.lua)"); > > Also: > Maybe you want to use it as a hangup hook. Instead of: > > Do: > > > > -Avi > > On Wed, Jan 7, 2015 at 1:52 AM, Sina Owolabi > wrote: > Hi List, > > FreeSWITCH newbie here again. > I am trying to cobble togther a lua callback script, my first attempt > was successful, but I am trying to make it slightly more elegant. > I don't see any errors when I try to run this but the callback isnt happening. > This is my very second attempt trying to write in lua, so I would be > very grateful for any help. > > The user is expected to dial in, have the call hangup and FreeSWITCH call back. > > I'm passing a modified $effective_caller_id_number and > $destination_number to the lua script: > > > expression="^1(\d{10})$"require-nested="false"> > > > > > > > > > > > > The script itself: > > api = freeswitch.API(); > call_string = "bagpi originate > {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" > > freeswitch.msleep(5000); > if (session:ready()) then > caller_id_number = session:getVariable("destination_number"); > caller_id_name = session:getVariable("destination_number"); > number_to_call = session:getVariable("effective_caller_id_number"); > > api:executeString(call_string); > freeswitch.msleep(2000); > session:streamFile("/tmp/get_off_my_lawn.wav"); > session:hangup("NORMAL_CLEARING"); > end > > _________________________________________________________________________ > 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 > _________________________________________________________________________ > 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 > > > > ???: Luis Daniel Lucio Quiroz > ???: FreeSWITCH Users Help > ??: 2015?1?16? GMT+808:18:10 > ??????: FreeSWITCH Users Help > ??: ??? [Freeswitch-users] how can i keep A-LEG do not hangup > > > Read about queues > > On Jan 15, 2015 11:53 AM, "Brian West" > wrote: > Thats not relevant in this case, The autoHangup is when used in say lua... > > This case you will probably want this combo: > > https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket#mod_event_socket-linger > https://wiki.freeswitch.org/wiki/Variable_park_after_bridge > > > > > On Thu, Jan 15, 2015 at 7:27 AM, Andrew > wrote: > This may be able to help you but I set session.SetAutoHangup(false); > > This will stop the dial plan from ending. > > ? <> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of ax lyb > Sent: Thursday, January 15, 2015 2:52 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] how can i keep A-LEG do not hangup > > > > All: > > recently i write a small program based freeswitch, in this program i only do things like callcenter mod, > > here is what i do: > > > > 1. originate a call (? originate sofia/gateway/gw01/xxxx my-out-call XML default?), in the default.xml, > > i already config it as follow: > > > > > > 65 > > 66 > > 67 > > 72 > > 73 > > > > 2. when call-out-destination customer pick up the call, i control the flow to playback a wav file, then , > > bridge the call to a free agent while the wav file play over in CHANNEL_EXECUTE_COMPLETE, > > these steps run normally ok. > > > > 3. the question is : if i have two agent in one moment, and this time i have 3 or > 3 out-call to bridge, > > i?ll leave the other call do nothing,for i can not get more free agents to service; some special scene > > is i have two agents (A1,A2) service 3 call (C1,C2,C3), A1 service C1, A2 service C2,then C3 is only > > A-LEG when play over a wav file,about 100 seconds later, C3 is hangup(it?s not user hangup manually), > > how can i do make the C3 call not hangup until i get a free agent to service it? > > > > > > any suggest is appreciate, > > > > > > > > ax.lyb.lei at gmail.com > > > > _________________________________________________________________________ > 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 > > > _______________________________________________ > 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 ax lyb ax.lyb.lei at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/95972414/attachment-0001.html From yehavi.bourvine at gmail.com Fri Jan 16 07:56:15 2015 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 16 Jan 2015 06:56:15 +0200 Subject: [Freeswitch-users] any handset recommendations that operate like polycom+cisco? In-Reply-To: <54B82BA0.20201@mst.edu> References: <54B82BA0.20201@mst.edu> Message-ID: SNOM has up to 12 SIPaccounts (as far as I remember). How many phoones do you need? If more than just a few, try asking Yealink for adding more SIP accounts. They are quite responsive for such requests. __Yehavi: 2015-01-15 23:05 GMT+02:00 Nathan Neulinger : > Looking for any handset/desk phone recommendations that have the "every > line key can be it's own SIP account, even on > expansion modules" behavior. > > Yealink units for example only allow a set number of lines, and the rest > don't operate as line keys, so even though you > can have a 30 + button phone, it's not usable for a receptionist or > secretary that may handle calls for 15 different > people with their own direct lines. > > With the polycom and cisco (sccp) phones, you can define N different SIP > accounts. > > Are there any other phones out there that operate this way that anyone can > recommend? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/80dffd1d/attachment.html From raphael.lechner at gmail.com Fri Jan 16 13:09:46 2015 From: raphael.lechner at gmail.com (Raphael Lechner) Date: Fri, 16 Jan 2015 11:09:46 +0100 Subject: [Freeswitch-users] voicemail silence_detection is always detecting silence Message-ID: Hi, I configured an extension that first call for some seconds a phone and if nobody is picking up, the caller is hearing a playback and can press 1 for leaving a voicemail and 2 to get redirected to a mobile phone. The Problem is that in my test environment with a SIP Provider everything works fine after tuning the voicemail.conf.xml to The called python script: def handler(session, args): voicemail = args.split(' ')[0] dtmf_pressed = args.split(' ')[1] forward_number = args.split(' ')[2] callerid = session.getVariable("caller_id_number") callername = session.getVariable("caller_id_name").lstrip() if dtmf_pressed == '1': send_sms('377XXXXXXX?,?New Voicemail from %s %s' % (callername, callerid)) session.execute("export", "skip_greeting=true") session.execute("export", "skip_instructions=true") session.execute("answer") session.execute("voicemail", "default 192.168.17.252 10?) #session.execute("bridge", "loopback/app=voicemail:default %s %s" % (conf['network']['ip'],voicemail)) elif dtmf_pressed == '2': consoleLog( "info", "Call is forwarded to %s\n" % forward_number) session.transfer(forward_number, "XML", "default") else: consoleLog( "info", "DTMF received is %s and not 1 or 2.Hangup Call\n" % (dtmf_pressed)) session.hangup() Thank you, Raphael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/e74d71d3/attachment.html From areski at gmail.com Fri Jan 16 14:39:19 2015 From: areski at gmail.com (Areski) Date: Fri, 16 Jan 2015 12:39:19 +0100 Subject: [Freeswitch-users] robo caller In-Reply-To: References: Message-ID: Yes, it's via this mod_amd module. On Thu, Jan 15, 2015 at 4:19 AM, Michael Collins wrote: > Areski, > > Does Newfies support mod_com_amd? Just curious how you handle machines. > -MC > > On Wed, Jan 14, 2015 at 8:44 AM, Areski wrote: > >> Newfies-Dialer (http://www.newfies-dialer.org/) might help and obviously >> it's built on top of FreeSWITCH. >> We built a flexible module for appointment reminders: >> http://docs.newfies-dialer.org/en/latest/user-guide-doc/appointment.html >> >> If you want to code this yourself, we use ESL ( >> https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library) >> to originate the calls and Lua to build the IVR part ( >> https://freeswitch.org/confluence/display/FREESWITCH/mod_lua). >> >> On Wed, Jan 14, 2015 at 5:37 PM, Moishe Grunstein >> wrote: >> >>> You can also have a look at http://www.newfies-dialer.org/ >>> >>> >>> >>> Thanks, >>> >>> >>> >>> Moishe Grunstein >>> >>> Tornado Computer Systems, Inc. >>> >>> 212.400.7650 888.IPPBX.US >>> *Service Request Email: support at nysolutions.com >>> * >>> >>> [image: cid:image001.jpg at 01C72F94.9EE45D60] >>> >>> >>> Computer Networking * Managed Services * IP Video Surveillance * Network >>> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >>> Security * Site Surveys * CMS >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David >>> Villasmil Govea >>> *Sent:* Wednesday, January 14, 2015 11:31 AM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] robo caller >>> >>> >>> >>> That's fairly simple to implement. Look into at mod_amd, orignate and >>> ivr. >>> >>> On Jan 14, 2015 5:28 PM, "Sean Devoy" wrote: >>> >>> Does anyone have a sample RoboCaller script? Perhaps I am using the >>> wrong name and that is why I can?t find one. I have a doctor?s office that >>> wants to automate the reminder calls about appointments to their patients. >>> >>> >>> >>> I am curious how people handle answering machine detection as well? >>> >>> >>> >>> Thanks, >>> >>> Sean >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> Kind regards, >> /Areski >> >> ---- >> Arezqui Belaid, >> Founder at Star2Billing (www.star2billing.com) >> >> Tel: +34650784355 >> Twitter: http://twitter.com/areskib >> LinkedIn: http://www.linkedin.com/in/areski >> >> _________________________________________________________________________ >> 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 > -- Kind regards, /Areski ---- Arezqui Belaid, Founder at Star2Billing (www.star2billing.com) Tel: +34650784355 Twitter: http://twitter.com/areskib LinkedIn: http://www.linkedin.com/in/areski -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/2ef85348/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/2ef85348/attachment-0001.jpg From raphael.lechner at gmail.com Fri Jan 16 15:24:12 2015 From: raphael.lechner at gmail.com (Raphael Lechner) Date: Fri, 16 Jan 2015 13:24:12 +0100 Subject: [Freeswitch-users] voicemail silence_detection is always detecting silence In-Reply-To: References: Message-ID: I missed to write the used FreeSWITCH version. I tried with 1.4.14 and 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit) Thank you > On 16 Jan 2015, at 11:09, Raphael Lechner wrote: > > Hi, > > I configured an extension that first call for some seconds a phone and if nobody is picking up, the caller is hearing a playback and can press 1 for leaving a voicemail and 2 to get redirected to a mobile phone. > The Problem is that in my test environment with a SIP Provider everything works fine after tuning the voicemail.conf.xml to > > > After that I can record a message and that works as expected. > Is there a way do disable the silence_detection or any hint what I can change? > > I tried changing the silence-threshold to 1,50 and silence-hits to 300,30000 but nothing has changed > > Debug Log > https://pastebin.freeswitch.org/23851 > > The called python script: > def handler(session, args): > voicemail = args.split(' ')[0] > dtmf_pressed = args.split(' ')[1] > forward_number = args.split(' ')[2] > callerid = session.getVariable("caller_id_number") > callername = session.getVariable("caller_id_name").lstrip() > > if dtmf_pressed == '1': > send_sms('377XXXXXXX?,?New Voicemail from %s %s' % (callername, callerid)) > session.execute("export", "skip_greeting=true") > session.execute("export", "skip_instructions=true") > session.execute("answer") > session.execute("voicemail", "default 192.168.17.252 10?) > #session.execute("bridge", "loopback/app=voicemail:default %s %s" % (conf['network']['ip'],voicemail)) > elif dtmf_pressed == '2': > consoleLog( "info", "Call is forwarded to %s\n" % forward_number) > session.transfer(forward_number, "XML", "default") > else: > consoleLog( "info", "DTMF received is %s and not 1 or 2.Hangup Call\n" % (dtmf_pressed)) > session.hangup() > > Thank you, > Raphael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/0448d091/attachment.html From ax.lyb.lei at gmail.com Fri Jan 16 15:33:29 2015 From: ax.lyb.lei at gmail.com (ax lyb) Date: Fri, 16 Jan 2015 20:33:29 +0800 Subject: [Freeswitch-users] how can i keep A-LEG do not hangup, Vol 103, Issue 199 In-Reply-To: References: Message-ID: ALL: JUST TRY SOME ideas then i can hold the A-LEG call wait until a free agent to service it, i do this to give a tone_stream after play a wav file, the implement piece of code just as follow: int CallOutProcessBase::processExecuteComplete(const FSBaseMsg &evt, CallTaskObj *pTask,IESLOperation* pCon) { const char* strOriginateUUid = evt.getHeader("variable_origination_uuid"); const char* strApp = evt.getHeader("Application"); const char* strInputKey = evt.getHeader(V_USER_INPUT); //const char* strFlag = evt.getHeader("variable_ax_proc_flag"); if(! strcmp( strApp,"playback")) { // const char* strParam = "1 1 2 2000 1 /usr/local/freeswitch/sounds/input_1.wav /usr/local/freeswitch/sounds/input_1.wav user_input_value ^(1)$" LOG(LS_ERROR)<<"channel playback Complete:"<find(strOriginateUUid); if( pSession && pSession->state() < E_CST_WAIT_AGENT ) { pCon->execute("playback", "tone_stream://%(2000,4000,440,480);loops=-1", strOriginateUUid,false); pTask->onCallStateChanged(strOriginateUUid, E_CST_WAIT_AGENT , strInputKey); LOG(LS_INFO) << "EXEC PLAYBACK STREAM:"<phoneNo() <<","<uuids( ); return 0; } LOG(LS_ERROR)<<"we need more agent:"<phoneNo()<<","<uuids(); return 0; } hope it?s useful to someone same as me! thanks for every one give me instructors > ? 2015?1?16??10:19?freeswitch-users-request at lists.freeswitch.org ??? > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: how can i keep A-LEG do not hangup (Luis Daniel Lucio > Quiroz) (ax lyb) > > ???: ax lyb > ???: freeswitch-users at lists.freeswitch.org > ??: 2015?1?16? GMT+810:19:06 > ??????: FreeSWITCH Users Help > ??: ??? [Freeswitch-users] how can i keep A-LEG do not hangup (Luis Daniel Lucio Quiroz) > > > hi, brian,Andrew > > i known a little in lua,i do my customer callcenter in the following style: > > 1. create a inbound connect to see what channel hangup > (BaseFreeswitchClient is a socket-connection wrap) > > int connectFS() > { > //"auth ClueCon\n\n"; > const char* AUTH_ME="ClueCon"; > m_pFsClient = new BaseFreeswitchClient(reinterpret_cast(this)); > if(m_pFsClient->ConnectFS(AUTH_ME) < 0 ) > { > delete m_pFsClient; > m_pFsClient = NULL; > return -1; > } > > // const char* strFilter = "event plain CHANNEL_CREATE \ > // CHANNEL_ANSWER \ > // CHANNEL_BRIDGE \ > // CHANNEL_HANGUP \ > // CHANNEL_HANGUP_COMPLETE\n\n"; > const char* strFilter = "event plain CHANNEL_HANGUP_COMPLETE\n\n"; > > if( m_pFsClient->filterEvent(strFilter) < 0 ) > { > delete m_pFsClient ; > m_pFsClient = NULL; > return -1; > } > return 0; > } > > 2. create a server for outbound connect, the outbound connect for every call-session to FILTER > CHANNEL_ANSWER,CHANNEL_HANGUP_COMPLETE,CHANNEL_EXECUTE_COMPLETE > > case ESL_EVENT_CHANNEL_DATA: > { > const char* strEvtName = evt.getHeader("Event-Name"); > > char szFilter[1024]; > const char* strFilter = "event plain CHANNEL_ANSWER CHANNEL_BRIDGE CHANNEL_EXECUTE_COMPLETE DTMF\n\n"; > snprintf(szFilter, sizeof(szFilter),"%sfilter Unique-ID %s\n\n",strFilter, > evt.getHeader("Caller-Unique-ID")); > int nWriteSize = strnlen(szFilter,sizeof(szFilter)); > LOG(LS_ERROR)< <<",nWriteSize="< p->onWrite(szFilter, nWriteSize); > } > > 3. set "session.SetAutoHangup(false);?, which channel_variable should i set? > > > thanks > > >> ? 2015?1?16??08:18?freeswitch-users-request at lists.freeswitch.org ??? >> >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> Today's Topics: >> >> 1. Re: Help Needed Debugging Lua Script (Luis Daniel Lucio Quiroz) >> 2. Re: how can i keep A-LEG do not hangup (Luis Daniel Lucio Quiroz) >> >> ???: Luis Daniel Lucio Quiroz >> ???: FreeSWITCH Users Help >> ??: 2015?1?16? GMT+808:13:15 >> ??????: FreeSWITCH Users Help >> ??: ??? [Freeswitch-users] Help Needed Debugging Lua Script >> >> >> Don't loose your time, contact me offline >> >> On Jan 15, 2015 5:49 PM, "Sina Owolabi" > wrote: >> Hi List, >> >> I think I have finally come up with something that works, the script captures the details correctly, but I wonder if there is a better way to write it? I am trying to get it to call the dialer back and play a message, which it does correctly. >> Please what can I do to make it to retry the dialler's number in case the call doesn't go through the first (say) two times, just in case the dialler cut it off mistakenly, or the telco plays a repeated message after freeswitch hung up, preventing the callback from coming through? (I've seen this happen a few times with a local telco). Thanks! >> >> number_to_call = argv[1]; >> caller_id = argv[2]; >> api = freeswitch.API(); >> dialString = "{origination_caller_id_name="..caller_id..",origination_caller_id_number="..caller_id.."}sofia/gateway/sipgw/"..number_to_call..""; >> session1 = freeswitch.Session(dialString); >> session1:sleep(5000); >> >> if (session1:ready()) then >> session1:sleep(35000); >> api:execute("bgapi originate", "session1"); >> session1:sleep(3000); >> session1:streamFile('/tmp/stop_calling_me_stalker.wav'); >> session1:hangup("NORMAL_CLEARING"); >> end >> >> >> On Wed Jan 07 2015 at 2:45:04 PM Avi Marcus > wrote: >> Two things: >> 1) You aren't grabbing the arg, but the channel variable.. try this in your script: >> caller_id_number = argv[1] >> number_to_call = argv[2] >> >> 2) I don't think you're managing your hangup/callback originate properly. >> I don't think you want to use bgapi... or maybe you just need a destination. It's "originate sofia/A endpoint" -- you need to specify where it goes to, the lua script can't "receive" the call. You can have it received by e.g: &lua(pickup.lua) >> >> api = freeswitch.API() >> api:execute("originate", DialString.." &lua(pickup.lua)"); >> >> Also: >> Maybe you want to use it as a hangup hook. Instead of: >> >> Do: >> >> >> >> -Avi >> >> On Wed, Jan 7, 2015 at 1:52 AM, Sina Owolabi > wrote: >> Hi List, >> >> FreeSWITCH newbie here again. >> I am trying to cobble togther a lua callback script, my first attempt >> was successful, but I am trying to make it slightly more elegant. >> I don't see any errors when I try to run this but the callback isnt happening. >> This is my very second attempt trying to write in lua, so I would be >> very grateful for any help. >> >> The user is expected to dial in, have the call hangup and FreeSWITCH call back. >> >> I'm passing a modified $effective_caller_id_number and >> $destination_number to the lua script: >> >> >> > expression="^1(\d{10})$"require-nested="false"> >> >> >> >> >> >> >> >> >> >> >> >> The script itself: >> >> api = freeswitch.API(); >> call_string = "bagpi originate >> {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" >> >> freeswitch.msleep(5000); >> if (session:ready()) then >> caller_id_number = session:getVariable("destination_number"); >> caller_id_name = session:getVariable("destination_number"); >> number_to_call = session:getVariable("effective_caller_id_number"); >> >> api:executeString(call_string); >> freeswitch.msleep(2000); >> session:streamFile("/tmp/get_off_my_lawn.wav"); >> session:hangup("NORMAL_CLEARING"); >> end >> >> _________________________________________________________________________ >> 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 >> _________________________________________________________________________ >> 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 >> >> >> >> ???: Luis Daniel Lucio Quiroz >> ???: FreeSWITCH Users Help >> ??: 2015?1?16? GMT+808:18:10 >> ??????: FreeSWITCH Users Help >> ??: ??? [Freeswitch-users] how can i keep A-LEG do not hangup >> >> >> Read about queues >> >> On Jan 15, 2015 11:53 AM, "Brian West" > wrote: >> Thats not relevant in this case, The autoHangup is when used in say lua... >> >> This case you will probably want this combo: >> >> https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket#mod_event_socket-linger >> https://wiki.freeswitch.org/wiki/Variable_park_after_bridge >> >> >> >> >> On Thu, Jan 15, 2015 at 7:27 AM, Andrew > wrote: >> This may be able to help you but I set session.SetAutoHangup(false); >> >> This will stop the dial plan from ending. >> >> ? <> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of ax lyb >> Sent: Thursday, January 15, 2015 2:52 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] how can i keep A-LEG do not hangup >> >> >> >> All: >> >> recently i write a small program based freeswitch, in this program i only do things like callcenter mod, >> >> here is what i do: >> >> >> >> 1. originate a call (? originate sofia/gateway/gw01/xxxx my-out-call XML default?), in the default.xml, >> >> i already config it as follow: >> >> >> >> >> >> 65 >> >> 66 >> >> 67 >> >> 72 >> >> 73 >> >> >> >> 2. when call-out-destination customer pick up the call, i control the flow to playback a wav file, then , >> >> bridge the call to a free agent while the wav file play over in CHANNEL_EXECUTE_COMPLETE, >> >> these steps run normally ok. >> >> >> >> 3. the question is : if i have two agent in one moment, and this time i have 3 or > 3 out-call to bridge, >> >> i?ll leave the other call do nothing,for i can not get more free agents to service; some special scene >> >> is i have two agents (A1,A2) service 3 call (C1,C2,C3), A1 service C1, A2 service C2,then C3 is only >> >> A-LEG when play over a wav file,about 100 seconds later, C3 is hangup(it?s not user hangup manually), >> >> how can i do make the C3 call not hangup until i get a free agent to service it? >> >> >> >> >> >> any suggest is appreciate, >> >> >> >> >> >> >> >> ax.lyb.lei at gmail.com >> >> >> >> _________________________________________________________________________ >> 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 >> >> >> _______________________________________________ >> 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 > > ax lyb > ax.lyb.lei at gmail.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 ax lyb ax.lyb.lei at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/79a2931a/attachment-0001.html From GeorgePhelps at gfphelps.com Fri Jan 16 15:42:16 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Fri, 16 Jan 2015 07:42:16 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <04ce01d0311e$272fac40$758f04c0$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <14 7601d02f32$37a9f5a0$a 6 fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> <007901d02ff5$67075090$3515f1b0$@gfphelps.com> <041401d030e5$ea3b6730$beb23590$@gfphelps.com> <04ce01d0311e$272fac40$758f04c0$@gfphelps.com> Message-ID: <053301d03189$dc9b9a10$95d2ce30$@gfphelps.com> Michael Collins, Any feedback on the debug log that I uploaded yesterday? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Thursday, January 15, 2015 6:51 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? New failed call log at: http://pastebin.com/cErTyGht Scenario? Two registered extensions, 1000 and 1001. Inbound call, to simultaneously ring both extensions. Both extensions start ringing. I answer on extension 1001. The call immediately drops, and then begins ringing again in a couple of seconds. >From the log? Line #754: extension 1001 OKs its INVITE. Line #835: debug messages indicates extension 1001 answered the call. Line #854: FreeSWITCH CANCELs the call to extension 1000. Line #887: FreeSWITCH terminates the inbound call with a ?480 Temporarily Unavailable? response. Line #930: FreeSWITCH sends a BYE to extension 1001. Line #1117: a new (regenerated), inbound call request. Without simultaneous ring enabled to two extensions, i.e., ringing only extension 1001 ? the call is handled with no problems. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Thursday, January 15, 2015 12:09 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Michael Collins, I had previously tried the ?Enterprise Originate? and syntax, but it did not make any difference at the time. I retested today with that syntax, and I am still seeing the same problem. If that is the recommended configuration for my situation, I will kept that syntax in my dialplan. I also tested with a difference VoIP service provider. Better results in that the INVITE timer now runs for 28 seconds, as opposed to just 10 seconds with the previous VoIP service provider. During the 28 seconds ? with both extensions ringing ? I was able to answer one extension. However, the call disconnected just as soon as I answered it, and then immediately started ringing again. This continued until the 28 seconds ran out. I will look into this later, and gather additional logs. I also retested, with the new VoIP service provider ? and dialing just one extension works fine. Dialing just one extension also worked with the previous VoIP service provider, even with the 10 second INVITE timer, as long as I answered within the 10 seconds window. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 10:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Glad you have the book. On page 19 it covers the use of enterprise originate. I think possibly you need to use the method as discussed on page 21. Try something like this: Could just be that early media is not being ignored on both user dialout attempts. -MC On Wed, Jan 14, 2015 at 4:27 AM, George F. Phelps wrote: Michael Collins, I already have the book. Thanks! Here?s my dialplan: New log file uploaded to: http://pastebin.com/gnEpPzk9 To me, the most significant event in the log file is the SIP CANCEL message ? starting at line #321: tport.c:3023 tport_deliver() tport_deliver(0x95daa0): msg 0xad8fb0 (437 bytes) from udp/169.XX.XX.XX:5080/sip next=(nil) nta.c:2880 agent_recv_request() nta: received CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (CSeq 1) nta.c:3026 agent_recv_request() nta: CANCEL (1) is going to INVITE (1) I don?t think it?s related, but I am also curious about log file line #285: sres.c:2987 sres_query_report_error() sres(q=0x98b050): reporting error NAME_ERR for SRV _sip._udp.sip.switch2voip.us Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 2:41 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? We covered this nicely in chapter 1 of the FreeSWITCH Cookbook I'm sorry that I'm late to the party so I am missing some information. Can you pastebin not only the call log but also the dialplan code for the example in question? One other tip: it appears that the log that you are pasting is coming directly from the FreeSWITCH console. By default the console does not have debug level output enabled. Try entering the command "console loglevel debug" and you'll see way more log lines, mostly yellow text. Those lines will most likely contain the clues needed to unravel this mystery. Thanks, MC On Tue, Jan 13, 2015 at 2:42 PM, George F. Phelps wrote: New logfile uploaded to: http://pastebin.com/CFFvVarS The log contains default Freeswitch console log messages, plus a SIP trace of a failed call. BTW, both extensions were ringing ? prior to the CANCEL message (see context below). In the log I see the INVITE from my VoIP service provider: recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: ------------------------------------------------------------------------ INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: ------------------------------------------------------------------------ SIP/2.0 100 Trying (Then, subsequent INVITE messages to my two extensions. But other no messages to/from my VoIP service provider.) And then, a spontaneous CANCEL from my VoIP service provider, approximately 10 seconds after the initial INVITE message. Due to a SIP ?Timer B? timeout? Seems way too short. recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: ------------------------------------------------------------------------ CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (Freeswitch cleanup of SIP sessions to my extensions?) Bote Man--> I have two local extensions. Individually, the extensions can make and receive both internal and external calls. It?s only the simultaneous ringing for external, inbound calls that is not working at the moment. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Tuesday, January 13, 2015 2:33 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I suggest you configure and register 3 total local phones to your FS installation, configure 2 of them as the target of your simultaneous ring group, and call them with the 3rd phone. Until you can get that working, calling through a carrier is adding another layer of complexity to the problem and confusing the issue. Out of the box FreeSWITCH does not utter voice codes, they must be coming from your carrier. Also, the debug-level logs very likely tell you exactly what is happening, even though they can be staggering to decipher as a newcomer to FS. Learning how to read them pays off in so many ways, though. I find the color-coded logs on the console or viewed via FS_cli to be helpful in these instances. Bote From: George F. Phelps Sent: Tuesday, 13 January, 2015 08:10 Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I tried? ?but that did not resolve the problem. 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: Tuesday, January 13, 2015 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" wrote: David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: Thanks, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/8d74ff5a/attachment-0001.html From rentmycoder at gmail.com Fri Jan 16 16:11:30 2015 From: rentmycoder at gmail.com (rentmycoder rentmycoder) Date: Fri, 16 Jan 2015 14:11:30 +0100 Subject: [Freeswitch-users] freeswitch reject calling without registration Message-ID: Hi all, How is it possible to to reject calling without registration? In asterisk there is no option for this... Thanks, John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/04cff0f3/attachment.html From bordmi at rarus.ru Fri Jan 16 16:49:23 2015 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Fri, 16 Jan 2015 17:49:23 +0400 Subject: [Freeswitch-users] working sipp scenario for testing kamailio+FS with authorization Message-ID: Have anyone subj? I`m look to RFC and see one, but when i`m looking to sip trace I see another... And it works on SIP-clients, but not on my scenario (in attach) ;tag=[pid][call_number] To: [field3] Call-ID: [call_id] CSeq: [cseq] INVITE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=[field0] 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> ;tag=[pid][call_number] To: [field3] [peer_tag_param] Call-ID: [call_id] CSeq: [cseq] ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: [len] v=0 o=[field0] 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> ;tag=[pid][call_number] To: [service] Call-ID: [call_id] CSeq: [cseq] INVITE Contact: sip:sipp@[local_ip]:[local_port] [field2] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> ;tag=[pid][call_number] To: [field3] [peer_tag_param] Call-ID: [call_id] CSeq: [cseq] ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/57e8db82/attachment.html From telishisheer at gmail.com Fri Jan 16 17:02:51 2015 From: telishisheer at gmail.com (Shisheer Teli) Date: Fri, 16 Jan 2015 19:32:51 +0530 Subject: [Freeswitch-users] Video call disconnect after 30 seconds in freeswitch In-Reply-To: References: <24EA0E57C97E4003A10704499F21E931@gmail.com> Message-ID: My all users are on public ip. so i don't need NAT. i checked in all SIP clients (x-lte, zoiper,qutecom) but still video disconnect after 30 seconds. Regards, Shisheer T On Mon, Jan 12, 2015 at 1:04 PM, Shisheer Teli wrote: > i still not found any solution for this ... > > On Sat, Jan 10, 2015 at 12:25 PM, Shisheer Teli > wrote: > >> jira has been made. >> >> FS-7143 >> >> https://freeswitch.org/jira/browse/FS-7143 >> >> Regrads, >> Shisheer T >> >> >> >> On Fri, Jan 9, 2015 at 7:30 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> iirc, were asked to make a jira with attached logs. >>> >>> >>> On Friday, January 9, 2015, Shisheer Teli >>> wrote: >>> >>>> I did following changes but still video call disconnect after 30 >>>> seconds, audio call working properly. >>>> >>>> Attched log file >>>> vars.xml >>>> >>>> >>>> >>>> >>>> host:domain.example.com is another possible value; however this will >>>> not toggle the autonat flags. If you are behind NAT, with dynamic DNS (and >>>> stun doesn't work) you should write a script that determines your public IP >>>> address, makes the change and calls reloadxml. This also holds true for the >>>> external profile. No special processing happens to determine the IP address >>>> before the variable gets passed to the external profile. >>>> >>>> host:domain.example.com may be used in places "where you have two >>>> interfaces in a box and one is public facing and one isn't, so one never >>>> has to tell the lies." >>>> >>>> - source bwk on irc. >>>> >>>> internal.xml >>>> >>>> >>>> >>>> external.xml >>>> >>>> >>>> >>>> >>>> >>>> On Fri, Jan 9, 2015 at 4:54 AM, Seven Du wrote: >>>> >>>>> perhaps report a jira with debug level log and >>>>> >>>>> sofia global siptrace on >>>>> >>>>> >>>>> -- >>>>> Seven Du >>>>> http://about.me/dujinfang >>>>> http://www.dujinfang.com >>>>> http://www.freeswitch.org.cn >>>>> >>>>> Sent with Sparrow >>>>> >>>>> On Thursday, January 8, 2015 at 6:52 PM, Shisheer Teli wrote: >>>>> >>>>> i am using FreeSWITCH Version >>>>> 1.5.15b+git~20150108T034022Z~1ed290e930~64bit (git 1ed290e 2015-01-08 >>>>> 03:40:22Z 64bit) >>>>> >>>>> and still video call disconnect after 30 seconds >>>>> >>>>> Regards, >>>>> Shisheer T >>>>> >>>>> On Wed, Jan 7, 2015 at 9:52 PM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>> Try latest master or 1.4.15 >>>>> >>>>> >>>>> On Wed, Jan 7, 2015 at 9:34 AM, Shisheer Teli >>>>> wrote: >>>>> >>>>> Hi Team, >>>>> >>>>> I don't know what happen , but when I start video call it disconnected >>>>> after every 30 seconds. >>>>> >>>>> e.g. >>>>> x-lite to x-lite call : video call disconnect after 30 seconds >>>>> >>>>> X-lite to Zoiper : video call continue, but no video sending. >>>>> >>>>> >>>>> Regards >>>>> Shisheer T >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>>> >>>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>>> http://twitter.com/FreeSWITCH >>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>>> * >>>>> >>>>> ClueCon Weekly Development Call >>>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> Shisheer Teli >>>>> Phone: +91-022 2278 2519 / 2121 >>>>> shisheer at tifr.res.in >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> Shisheer Teli >>>> Phone: +91-022 2278 2519 / 2121 >>>> shisheer at tifr.res.in >>>> >>> >>> >>> -- >>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>> >>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>> http://twitter.com/FreeSWITCH >>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>> >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> Regards, >> Shisheer Teli >> Phone: +91-022 2278 2519 / 2121 >> shisheer at tifr.res.in >> > > -- Regards, Shisheer Teli Phone: +91-022 2278 2519 / 2121 shisheer at tifr.res.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/a3161623/attachment-0001.html From notify.sina at gmail.com Fri Jan 16 17:36:36 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Fri, 16 Jan 2015 14:36:36 +0000 Subject: [Freeswitch-users] Help Needed Debugging Lua Script References: <0000014ac4a0ec5e-81630ae2-7f78-4b3a-8659-6ba76a3c9b96-000000@email.amazonses.com> Message-ID: But how will I learn? On Fri, 16 Jan 2015 at 01:17, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > Don't loose your time, contact me offline > On Jan 15, 2015 5:49 PM, "Sina Owolabi" wrote: > >> Hi List, >> >> I think I have finally come up with something that works, the script >> captures the details correctly, but I wonder if there is a better way to >> write it? I am trying to get it to call the dialer back and play a message, >> which it does correctly. >> Please what can I do to make it to retry the dialler's number in case the >> call doesn't go through the first (say) two times, just in case the dialler >> cut it off mistakenly, or the telco plays a repeated message after >> freeswitch hung up, preventing the callback from coming through? (I've seen >> this happen a few times with a local telco). Thanks! >> >> number_to_call = argv[1]; >> caller_id = argv[2]; >> api = freeswitch.API(); >> dialString = >> "{origination_caller_id_name="..caller_id..",origination_caller_id_number="..caller_id.."}sofia/gateway/sipgw/"..number_to_call..""; >> session1 = freeswitch.Session(dialString); >> session1:sleep(5000); >> >> if (session1:ready()) then >> session1:sleep(35000); >> api:execute("bgapi originate", "session1"); >> session1:sleep(3000); >> session1:streamFile('/tmp/stop_calling_me_stalker.wav'); >> session1:hangup("NORMAL_CLEARING"); >> end >> >> >> On Wed Jan 07 2015 at 2:45:04 PM Avi Marcus wrote: >> >>> Two things: >>> 1) You aren't grabbing the arg, but the channel variable.. try this in >>> your script: >>> caller_id_number = argv[1] >>> number_to_call = argv[2] >>> >>> 2) I don't think you're managing your hangup/callback originate properly. >>> I don't think you want to use bgapi... or maybe you just need a >>> destination. It's "originate >>> sofia/A >>> endpoint" -- you need to specify where it goes to, the lua script can't >>> "receive" the call. You can have it received by e.g: &lua(pickup.lua) >>> >>> api = freeswitch.API() >>> api:execute("originate", DialString.." &lua(pickup.lua)"); >>> >>> Also: >>> Maybe you want to use it as a hangup hook. Instead of: >>> >>> Do: >>> >>> >>> >>> -Avi >>> >>> On Wed, Jan 7, 2015 at 1:52 AM, Sina Owolabi >>> wrote: >>> >>>> Hi List, >>>> >>>> FreeSWITCH newbie here again. >>>> I am trying to cobble togther a lua callback script, my first attempt >>>> was successful, but I am trying to make it slightly more elegant. >>>> I don't see any errors when I try to run this but the callback isnt >>>> happening. >>>> This is my very second attempt trying to write in lua, so I would be >>>> very grateful for any help. >>>> >>>> The user is expected to dial in, have the call hangup and FreeSWITCH >>>> call back. >>>> >>>> I'm passing a modified $effective_caller_id_number and >>>> $destination_number to the lua script: >>>> >>>> >>>> >>> expression="^1(\d{10})$"require-nested="false"> >>>> >>> data="effective_caller_id_number=+234${1}"/> >>>> >>> data="effective_caller_id_name=+234${1}"/> >>>> >>>> >>>> >>> data="destination_number=+12312345${1}${2}" /> >>>> >>>> >>>> >>>> >>>> >>>> >>>> The script itself: >>>> >>>> api = freeswitch.API(); >>>> call_string = "bagpi originate >>>> >>>> {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" >>>> >>>> freeswitch.msleep(5000); >>>> if (session:ready()) then >>>> caller_id_number = session:getVariable("destination_number"); >>>> caller_id_name = session:getVariable("destination_number"); >>>> number_to_call = >>>> session:getVariable("effective_caller_id_number"); >>>> >>>> api:executeString(call_string); >>>> freeswitch.msleep(2000); >>>> session:streamFile("/tmp/get_off_my_lawn.wav"); >>>> session:hangup("NORMAL_CLEARING"); >>>> end >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >> >> >> _________________________________________________________________________ >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/6587fefa/attachment.html From david.villasmil at gmail.com Fri Jan 16 17:40:30 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Fri, 16 Jan 2015 15:40:30 +0100 Subject: [Freeswitch-users] Help Needed Debugging Lua Script In-Reply-To: References: <0000014ac4a0ec5e-81630ae2-7f78-4b3a-8659-6ba76a3c9b96-000000@email.amazonses.com> Message-ID: Correct attitude! On Jan 16, 2015 3:37 PM, "Sina Owolabi" wrote: > But how will I learn? > On Fri, 16 Jan 2015 at 01:17, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > >> Don't loose your time, contact me offline >> On Jan 15, 2015 5:49 PM, "Sina Owolabi" wrote: >> >>> Hi List, >>> >>> I think I have finally come up with something that works, the script >>> captures the details correctly, but I wonder if there is a better way to >>> write it? I am trying to get it to call the dialer back and play a message, >>> which it does correctly. >>> Please what can I do to make it to retry the dialler's number in case >>> the call doesn't go through the first (say) two times, just in case the >>> dialler cut it off mistakenly, or the telco plays a repeated message after >>> freeswitch hung up, preventing the callback from coming through? (I've seen >>> this happen a few times with a local telco). Thanks! >>> >>> number_to_call = argv[1]; >>> caller_id = argv[2]; >>> api = freeswitch.API(); >>> dialString = >>> "{origination_caller_id_name="..caller_id..",origination_caller_id_number="..caller_id.."}sofia/gateway/sipgw/"..number_to_call..""; >>> session1 = freeswitch.Session(dialString); >>> session1:sleep(5000); >>> >>> if (session1:ready()) then >>> session1:sleep(35000); >>> api:execute("bgapi originate", "session1"); >>> session1:sleep(3000); >>> session1:streamFile('/tmp/stop_calling_me_stalker.wav'); >>> session1:hangup("NORMAL_CLEARING"); >>> end >>> >>> >>> On Wed Jan 07 2015 at 2:45:04 PM Avi Marcus wrote: >>> >>>> Two things: >>>> 1) You aren't grabbing the arg, but the channel variable.. try this in >>>> your script: >>>> caller_id_number = argv[1] >>>> number_to_call = argv[2] >>>> >>>> 2) I don't think you're managing your hangup/callback originate >>>> properly. >>>> I don't think you want to use bgapi... or maybe you just need a >>>> destination. It's "originate >>>> sofia/A >>>> endpoint" -- you need to specify where it goes to, the lua script can't >>>> "receive" the call. You can have it received by e.g: &lua(pickup.lua) >>>> >>>> api = freeswitch.API() >>>> api:execute("originate", DialString.." &lua(pickup.lua)"); >>>> >>>> Also: >>>> Maybe you want to use it as a hangup hook. Instead of: >>>> >>>> Do: >>>> >>>> >>>> >>>> -Avi >>>> >>>> On Wed, Jan 7, 2015 at 1:52 AM, Sina Owolabi >>>> wrote: >>>> >>>>> Hi List, >>>>> >>>>> FreeSWITCH newbie here again. >>>>> I am trying to cobble togther a lua callback script, my first attempt >>>>> was successful, but I am trying to make it slightly more elegant. >>>>> I don't see any errors when I try to run this but the callback isnt >>>>> happening. >>>>> This is my very second attempt trying to write in lua, so I would be >>>>> very grateful for any help. >>>>> >>>>> The user is expected to dial in, have the call hangup and FreeSWITCH >>>>> call back. >>>>> >>>>> I'm passing a modified $effective_caller_id_number and >>>>> $destination_number to the lua script: >>>>> >>>>> >>>>> >>>> expression="^1(\d{10})$"require-nested="false"> >>>>> >>>> data="effective_caller_id_number=+234${1}"/> >>>>> >>>> data="effective_caller_id_name=+234${1}"/> >>>>> >>>>> >>>>> >>>> data="destination_number=+12312345${1}${2}" /> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> The script itself: >>>>> >>>>> api = freeswitch.API(); >>>>> call_string = "bagpi originate >>>>> >>>>> {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" >>>>> >>>>> freeswitch.msleep(5000); >>>>> if (session:ready()) then >>>>> caller_id_number = session:getVariable("destination_number"); >>>>> caller_id_name = session:getVariable("destination_number"); >>>>> number_to_call = >>>>> session:getVariable("effective_caller_id_number"); >>>>> >>>>> api:executeString(call_string); >>>>> freeswitch.msleep(2000); >>>>> session:streamFile("/tmp/get_off_my_lawn.wav"); >>>>> session:hangup("NORMAL_CLEARING"); >>>>> end >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>> >>> >>> _________________________________________________________________________ >>> 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 > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/c3b990d7/attachment-0001.html From notify.sina at gmail.com Fri Jan 16 17:47:24 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Fri, 16 Jan 2015 14:47:24 +0000 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? Message-ID: Hi List! I'm checking out GUI billing and CDR records applications for FreeSWITCH. Most I've tried (bluebox, fusion, vbilling ) seem to either want to delete my existing configuration , or install their own Freeswitch , don't like CentOS, and totally ignore my own which I'm quite partial to. I wonder if there are any suggestions of CDR applications that will just fit in to what's already there? I'm running the latest master on a CentOS 64bit server. Thanks in advance ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/39a6cad7/attachment.html From krice at freeswitch.org Fri Jan 16 18:04:44 2015 From: krice at freeswitch.org (Ken Rice) Date: Fri, 16 Jan 2015 15:04:44 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <54b9288c8f3aa_d618b5333061655@ip-10-16-128-237.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/52ed5597/attachment.html From krice at freeswitch.org Fri Jan 16 18:11:54 2015 From: krice at freeswitch.org (Ken Rice) Date: Fri, 16 Jan 2015 09:11:54 -0600 Subject: [Freeswitch-users] freeswitch reject calling without registration In-Reply-To: Message-ID: There?s not really an options for this altho you could create one by checking ot see if a user is registered in the sofia database registrations table. However if you are doing this for security, that?s a fairly poor choice, but spec outbound calling has nothing to do with registration, registration is simply to tell the server where to send calls for a dynamically addressed end point. Using registration as a filter first to be allowed to make calls is a pretty low bar to get thru as the credentials are the same either way. On 1/16/15, 7:11 AM, "rentmycoder rentmycoder" wrote: > Hi all, > > How is it possible to to reject calling without registration? > In asterisk there is no option for this... > > Thanks, > John > > > _________________________________________________________________________ > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/47bf3b28/attachment.html From vipkilla at gmail.com Fri Jan 16 18:18:44 2015 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 16 Jan 2015 10:18:44 -0500 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? In-Reply-To: References: Message-ID: There are many CDR modues... Check confluence: https://freeswitch.org/confluence/dosearchsite.action?queryString=cdr+modules https://freeswitch.org/confluence/display/FREESWITCH/CDR On Fri, Jan 16, 2015 at 9:47 AM, Sina Owolabi wrote: > Hi List! > > I'm checking out GUI billing and CDR records applications for FreeSWITCH. > Most I've tried (bluebox, fusion, vbilling ) seem to either want to delete > my existing configuration , or install their own Freeswitch , don't like > CentOS, and totally ignore my own which I'm quite partial to. > > I wonder if there are any suggestions of CDR applications that will just > fit in to what's already there? I'm running the latest master on a CentOS > 64bit server. > > Thanks in advance ! > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/f2d9bcee/attachment.html From abalashov at evaristesys.com Fri Jan 16 18:39:05 2015 From: abalashov at evaristesys.com (Alex Balashov) Date: Fri, 16 Jan 2015 10:39:05 -0500 Subject: [Freeswitch-users] Polycom busy lamp with IVR? Message-ID: <54B93099.5070000@evaristesys.com> Hi, Sorry if it's a mundane question: I've got two Polycom phones sharing a registration appearance and SLA / busy lamp works fine with them. What I would like is to create a second line appearance whose busy lamp lights up when a call comes into our main IVR. Is this possible? How would I do it? Are there presence hints that can be arbitrarily directed from the dialplan? I wasn't able to discern this from the Polycom presence or general Polycom documentation. Thank you! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ From max at nysolutions.com Fri Jan 16 19:00:16 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 16 Jan 2015 16:00:16 +0000 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? In-Reply-To: References: Message-ID: FusionPBX does cdr?s however not billing, if you are looking for a billing solution see https://wiki.freeswitch.org/wiki/Billing Anything should work on Cenos, however Debian is currently the recommended Freeswitch platform. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sina Owolabi Sent: Friday, January 16, 2015 9:47 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? Hi List! I'm checking out GUI billing and CDR records applications for FreeSWITCH. Most I've tried (bluebox, fusion, vbilling ) seem to either want to delete my existing configuration , or install their own Freeswitch , don't like CentOS, and totally ignore my own which I'm quite partial to. I wonder if there are any suggestions of CDR applications that will just fit in to what's already there? I'm running the latest master on a CentOS 64bit server. Thanks in advance ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/d64a2a02/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/d64a2a02/attachment-0001.jpg From danb.lists at gmail.com Fri Jan 16 19:10:42 2015 From: danb.lists at gmail.com (DanB) Date: Fri, 16 Jan 2015 17:10:42 +0100 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? In-Reply-To: References: Message-ID: <54B93802.2020906@gmail.com> Hi Sina, CGRateS is a billing engine and I think it has what you are looking for. It does not touch/influence your FreeSWITCH configuration files. It can serve you as simple CDR server or complete charging solution with some pretty fancy prepaid/postpaid scenarios. Once calculated, CDRs will be accessible to be exported or retrieved via JSON-RPC calls. DanB From notify.sina at gmail.com Fri Jan 16 20:52:44 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Fri, 16 Jan 2015 17:52:44 +0000 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? References: <54B93802.2020906@gmail.com> Message-ID: Thanks everyone. Maybe I should switch to Debian after all. On Fri Jan 16 2015 at 5:14:44 PM DanB wrote: > Hi Sina, > > CGRateS is a billing engine and I think it has what you are looking for. > It does not touch/influence your FreeSWITCH configuration files. It can > serve you as simple CDR server or complete charging solution with some > pretty fancy prepaid/postpaid scenarios. Once calculated, CDRs will be > accessible to be exported or retrieved via JSON-RPC calls. > > DanB > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/1683df5e/attachment.html From zoell at zoell.us Fri Jan 16 20:58:48 2015 From: zoell at zoell.us (=?UTF-8?B?Wm9sdMOhbiBTemFiw7M=?=) Date: Fri, 16 Jan 2015 17:58:48 +0000 Subject: [Freeswitch-users] Asterisk's nat=yes feature Message-ID: Hi, I have a scenario where my softphone is behind nat and freeswitch is not behind nat. With asterisk I could make the audio work with nat=yes in the user's configuration. What is the settings for this on freeswitch? I have tried everything from the following pages: http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA http://wiki.freeswitch.org/wiki/NAT http://wiki.freeswitch.org/wiki/NAT_Traversal http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios (btw these are linking back to each other) Is it not that just easy as net=yes in asterisk? Many thanks, Zoltan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/8ab94b9c/attachment.html From auge at virtues.net Fri Jan 16 21:12:34 2015 From: auge at virtues.net (Thomas Auge) Date: Fri, 16 Jan 2015 15:12:34 -0300 Subject: [Freeswitch-users] mod_opus bug? Message-ID: <54B95492.2090309@virtues.net> Hello list, I noticed that in an opus call where the a-leg sends usedtx=0, it is properly forwarded to the b-leg, which in this case also correctly responds with usedtx=0. However, the local SDP for the a-leg then does not have usedtx in the fmtp at all, and it seems to be enabled. My C skills are limited, but could this be a bug: static opus_codec_settings_t default_codec_settings = { .... /*.usedtx */ 1, .... }; By default, usedtx is enabled, correct? However, in static char *gen_fmtp(opus_codec_settings_t *settings, switch_memory_pool_t *pool) there's this: if (settings->usedtx) { snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "usedtx=1; "); } Am I interpreting this correctly that if usedtx is zero, it is omitted in the fmtp line, but then enabled, because that's the default setting? Problem is that Chrome can't handle DTX and generates comfort noise every time it kicks in. Cheers! Thomas P.S.: If my interpretation is correct, it'll also affect useinbandfec. From anthony.minessale at gmail.com Fri Jan 16 21:16:35 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 16 Jan 2015 12:16:35 -0600 Subject: [Freeswitch-users] mod_opus bug? In-Reply-To: <54B95492.2090309@virtues.net> References: <54B95492.2090309@virtues.net> Message-ID: Its also a bug to send bugs to the mailing list instead of http://jira.freeswitch.org On Fri, Jan 16, 2015 at 12:12 PM, Thomas Auge wrote: > Hello list, > > I noticed that in an opus call where the a-leg sends usedtx=0, it is > properly forwarded to the b-leg, which in this case > also correctly responds with usedtx=0. However, the local SDP for the > a-leg then does not have usedtx in the fmtp at > all, and it seems to be enabled. > > My C skills are limited, but could this be a bug: > > static opus_codec_settings_t default_codec_settings = { > .... > /*.usedtx */ 1, > .... > }; > > By default, usedtx is enabled, correct? > > However, in > > static char *gen_fmtp(opus_codec_settings_t *settings, > switch_memory_pool_t *pool) > > there's this: > > if (settings->usedtx) { > snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "usedtx=1; "); > } > > > Am I interpreting this correctly that if usedtx is zero, it is omitted in > the fmtp line, but then enabled, because > that's the default setting? > > Problem is that Chrome can't handle DTX and generates comfort noise every > time it kicks in. > > Cheers! > > Thomas > > P.S.: If my interpretation is correct, it'll also affect useinbandfec. > > _________________________________________________________________________ > 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/a0756322/attachment.html From auge at virtues.net Fri Jan 16 21:26:04 2015 From: auge at virtues.net (Thomas Auge) Date: Fri, 16 Jan 2015 15:26:04 -0300 Subject: [Freeswitch-users] mod_opus bug? In-Reply-To: References: <54B95492.2090309@virtues.net> Message-ID: <54B957BC.4080405@virtues.net> I figured I'd ask for an opinion before filing a bogus bug. :P On 16.01.2015 15:16, Anthony Minessale wrote: > Its also a bug to send bugs to the mailing list instead of http://jira.freeswitch.org > > > On Fri, Jan 16, 2015 at 12:12 PM, Thomas Auge > wrote: > > Hello list, > > I noticed that in an opus call where the a-leg sends usedtx=0, it is properly forwarded to the b-leg, which in this case > also correctly responds with usedtx=0. However, the local SDP for the a-leg then does not have usedtx in the fmtp at > all, and it seems to be enabled. > > My C skills are limited, but could this be a bug: > > static opus_codec_settings_t default_codec_settings = { > .... > /*.usedtx */ 1, > .... > }; > > By default, usedtx is enabled, correct? > > However, in > > static char *gen_fmtp(opus_codec_settings_t *settings, switch_memory_pool_t *pool) > > there's this: > > if (settings->usedtx) { > snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "usedtx=1; "); > } > > > Am I interpreting this correctly that if usedtx is zero, it is omitted in the fmtp line, but then enabled, because > that's the default setting? > > Problem is that Chrome can't handle DTX and generates comfort noise every time it kicks in. > > Cheers! > > Thomas > > P.S.: If my interpretation is correct, it'll also affect useinbandfec. > > _________________________________________________________________________ > 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 > > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? _http://freeswitch.org/g+_ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > > _________________________________________________________________________ > 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 > From anthony.minessale at gmail.com Fri Jan 16 21:28:02 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 16 Jan 2015 12:28:02 -0600 Subject: [Freeswitch-users] mod_opus bug? In-Reply-To: <54B957BC.4080405@virtues.net> References: <54B95492.2090309@virtues.net> <54B957BC.4080405@virtues.net> Message-ID: Its better to ask by filing the jira its one click to close it and if there is a real issue we need logs etc attached and something to reference if we push a fix. On Fri, Jan 16, 2015 at 12:26 PM, Thomas Auge wrote: > I figured I'd ask for an opinion before filing a bogus bug. :P > > > On 16.01.2015 15:16, Anthony Minessale wrote: > > Its also a bug to send bugs to the mailing list instead of > http://jira.freeswitch.org > > > > > > On Fri, Jan 16, 2015 at 12:12 PM, Thomas Auge auge at virtues.net>> wrote: > > > > Hello list, > > > > I noticed that in an opus call where the a-leg sends usedtx=0, it is > properly forwarded to the b-leg, which in this case > > also correctly responds with usedtx=0. However, the local SDP for > the a-leg then does not have usedtx in the fmtp at > > all, and it seems to be enabled. > > > > My C skills are limited, but could this be a bug: > > > > static opus_codec_settings_t default_codec_settings = { > > .... > > /*.usedtx */ 1, > > .... > > }; > > > > By default, usedtx is enabled, correct? > > > > However, in > > > > static char *gen_fmtp(opus_codec_settings_t *settings, > switch_memory_pool_t *pool) > > > > there's this: > > > > if (settings->usedtx) { > > snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), > "usedtx=1; "); > > } > > > > > > Am I interpreting this correctly that if usedtx is zero, it is > omitted in the fmtp line, but then enabled, because > > that's the default setting? > > > > Problem is that Chrome can't handle DTX and generates comfort noise > every time it kicks in. > > > > Cheers! > > > > Thomas > > > > P.S.: If my interpretation is correct, it'll also affect > useinbandfec. > > > > > _________________________________________________________________________ > > 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 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 > > > > > > > > > > -- > > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > > ? irc.freenode.net #freeswitch ? _ > http://freeswitch.org/g+_ > > > > ClueCon Weekly Development Call > > ? sip:888 at conference.freeswitch.org sip%3A888 at conference.freeswitch.org> ? +19193869900 > > > > > > > > _________________________________________________________________________ > > 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/7b3a3faa/attachment-0001.html From anthony.minessale at gmail.com Fri Jan 16 21:39:44 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 16 Jan 2015 12:39:44 -0600 Subject: [Freeswitch-users] mod_opus bug? In-Reply-To: References: <54B95492.2090309@virtues.net> <54B957BC.4080405@virtues.net> Message-ID: Looking at your issue however I don't see dtx being enabled when I lab up what you described. Try this in vars.xml to disable CNG completely. On Fri, Jan 16, 2015 at 12:28 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Its better to ask by filing the jira its one click to close it and if > there is a real issue we need logs etc attached and something to reference > if we push a fix. > > > > On Fri, Jan 16, 2015 at 12:26 PM, Thomas Auge wrote: > >> I figured I'd ask for an opinion before filing a bogus bug. :P >> >> >> On 16.01.2015 15:16, Anthony Minessale wrote: >> > Its also a bug to send bugs to the mailing list instead of >> http://jira.freeswitch.org >> > >> > >> > On Fri, Jan 16, 2015 at 12:12 PM, Thomas Auge > > wrote: >> > >> > Hello list, >> > >> > I noticed that in an opus call where the a-leg sends usedtx=0, it >> is properly forwarded to the b-leg, which in this case >> > also correctly responds with usedtx=0. However, the local SDP for >> the a-leg then does not have usedtx in the fmtp at >> > all, and it seems to be enabled. >> > >> > My C skills are limited, but could this be a bug: >> > >> > static opus_codec_settings_t default_codec_settings = { >> > .... >> > /*.usedtx */ 1, >> > .... >> > }; >> > >> > By default, usedtx is enabled, correct? >> > >> > However, in >> > >> > static char *gen_fmtp(opus_codec_settings_t *settings, >> switch_memory_pool_t *pool) >> > >> > there's this: >> > >> > if (settings->usedtx) { >> > snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), >> "usedtx=1; "); >> > } >> > >> > >> > Am I interpreting this correctly that if usedtx is zero, it is >> omitted in the fmtp line, but then enabled, because >> > that's the default setting? >> > >> > Problem is that Chrome can't handle DTX and generates comfort noise >> every time it kicks in. >> > >> > Cheers! >> > >> > Thomas >> > >> > P.S.: If my interpretation is correct, it'll also affect >> useinbandfec. >> > >> > >> _________________________________________________________________________ >> > 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 > 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 >> > >> > >> > >> > >> > -- >> > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >> > >> > ? http://freeswitch.org/ ? http://cluecon.com/ ? >> http://twitter.com/FreeSWITCH >> > ? irc.freenode.net #freeswitch ? _ >> http://freeswitch.org/g+_ >> > >> > ClueCon Weekly Development Call >> > ? sip:888 at conference.freeswitch.org > sip%3A888 at conference.freeswitch.org> ? +19193869900 >> > >> > >> > >> > >> _________________________________________________________________________ >> > 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 >> > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/affe029c/attachment.html From notify.sina at gmail.com Fri Jan 16 21:42:51 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Fri, 16 Jan 2015 18:42:51 +0000 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? References: <54B93802.2020906@gmail.com> Message-ID: Quick question though.. Which is best, git clone freeswitch or to use the Debian repo? On Fri Jan 16 2015 at 6:52:44 PM Sina Owolabi wrote: > Thanks everyone. Maybe I should switch to Debian after all. > > On Fri Jan 16 2015 at 5:14:44 PM DanB wrote: > >> Hi Sina, >> >> CGRateS is a billing engine and I think it has what you are looking for. >> It does not touch/influence your FreeSWITCH configuration files. It can >> serve you as simple CDR server or complete charging solution with some >> pretty fancy prepaid/postpaid scenarios. Once calculated, CDRs will be >> accessible to be exported or retrieved via JSON-RPC calls. >> >> DanB >> >> _________________________________________________________________________ >> 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 >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/b446e100/attachment.html From vipkilla at gmail.com Fri Jan 16 21:57:04 2015 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 16 Jan 2015 13:57:04 -0500 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? In-Reply-To: References: <54B93802.2020906@gmail.com> Message-ID: I prefer git clone master branch. On Fri, Jan 16, 2015 at 1:42 PM, Sina Owolabi wrote: > Quick question though.. Which is best, git clone freeswitch or to use the > Debian repo? > > > On Fri Jan 16 2015 at 6:52:44 PM Sina Owolabi > wrote: > >> Thanks everyone. Maybe I should switch to Debian after all. >> >> On Fri Jan 16 2015 at 5:14:44 PM DanB wrote: >> >>> Hi Sina, >>> >>> CGRateS is a billing engine and I think it has what you are looking for. >>> It does not touch/influence your FreeSWITCH configuration files. It can >>> serve you as simple CDR server or complete charging solution with some >>> pretty fancy prepaid/postpaid scenarios. Once calculated, CDRs will be >>> accessible to be exported or retrieved via JSON-RPC calls. >>> >>> DanB >>> >>> ____________________________________________________________ >>> _____________ >>> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/e15bd19f/attachment-0001.html From vipkilla at gmail.com Fri Jan 16 21:57:58 2015 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 16 Jan 2015 13:57:58 -0500 Subject: [Freeswitch-users] Asterisk's nat=yes feature In-Reply-To: References: Message-ID: Are you having NAT issues? FS is pretty good at auto-fixing NAT On Fri, Jan 16, 2015 at 12:58 PM, Zolt?n Szab? wrote: > Hi, > > I have a scenario where my softphone is behind nat and freeswitch is not > behind nat. With asterisk I could make the audio work with nat=yes in the > user's configuration. > > What is the settings for this on freeswitch? > > I have tried everything from the following pages: > http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA > http://wiki.freeswitch.org/wiki/NAT > http://wiki.freeswitch.org/wiki/NAT_Traversal > http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios > > (btw these are linking back to each other) > > Is it not that just easy as net=yes in asterisk? > > Many thanks, > Zoltan > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/5a437359/attachment.html From auge at virtues.net Fri Jan 16 21:59:45 2015 From: auge at virtues.net (Thomas Auge) Date: Fri, 16 Jan 2015 15:59:45 -0300 Subject: [Freeswitch-users] mod_opus bug? In-Reply-To: References: <54B95492.2090309@virtues.net> <54B957BC.4080405@virtues.net> Message-ID: <54B95FA1.2050106@virtues.net> Stupid question: Where do you "see" that? I couldn't find anything in the logs. I don't *know* if it was enabled. We just had some embarrassing sound glitches on air today which sounded exactly like what happened when DTX was enabled, and then I noticed that usedtx=0 wasn't there in the a-leg local SDP, it's enabled by default in mod_opus, and figured (hoped) that might be it. I didn't know comfort noise could be disabled on the freeswitch side. I wonder how Chrome reacts to that. The Chrome people did discuss the merit of CN, but opted to keep it. :/ Will try that, thanks! :-) On 16.01.2015 15:39, Anthony Minessale wrote: > Looking at your issue however I don't see dtx being enabled when I lab up what you described. > > Try this in vars.xml to disable CNG completely. > > > > > > > On Fri, Jan 16, 2015 at 12:28 PM, Anthony Minessale > > wrote: > > Its better to ask by filing the jira its one click to close it and if there is a real issue we need logs etc > attached and something to reference if we push a fix. > > > > On Fri, Jan 16, 2015 at 12:26 PM, Thomas Auge > wrote: > > I figured I'd ask for an opinion before filing a bogus bug. :P > > > On 16.01.2015 15:16, Anthony Minessale wrote: > > Its also a bug to send bugs to the mailing list instead ofhttp://jira.freeswitch.org > > > > > > On Fri, Jan 16, 2015 at 12:12 PM, Thomas Auge >> wrote: > > > > Hello list, > > > > I noticed that in an opus call where the a-leg sends usedtx=0, it is properly forwarded to the b-leg, which in this case > > also correctly responds with usedtx=0. However, the local SDP for the a-leg then does not have usedtx in the fmtp at > > all, and it seems to be enabled. > > > > My C skills are limited, but could this be a bug: > > > > static opus_codec_settings_t default_codec_settings = { > > .... > > /*.usedtx */ 1, > > .... > > }; > > > > By default, usedtx is enabled, correct? > > > > However, in > > > > static char *gen_fmtp(opus_codec_settings_t *settings, switch_memory_pool_t *pool) > > > > there's this: > > > > if (settings->usedtx) { > > snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "usedtx=1; "); > > } > > > > > > Am I interpreting this correctly that if usedtx is zero, it is omitted in the fmtp line, but then enabled, because > > that's the default setting? > > > > Problem is that Chrome can't handle DTX and generates comfort noise every time it kicks in. > > > > Cheers! > > > > Thomas > > > > P.S.: If my interpretation is correct, it'll also affect useinbandfec. > > > > _________________________________________________________________________ > > 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 > > > > > > > > > > -- > > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > > > ?http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > > ? irc.freenode.net #freeswitch ? _http://freeswitch.org/g+_ > > > > ClueCon Weekly Development Call > > ? sip:888 at conference.freeswitch.org > > ? +19193869900 > > > > > > > > > _________________________________________________________________________ > > 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 > > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? _http://freeswitch.org/g+_ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? _http://freeswitch.org/g+_ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > > _________________________________________________________________________ > 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 > From auge at virtues.net Fri Jan 16 22:11:32 2015 From: auge at virtues.net (Thomas Auge) Date: Fri, 16 Jan 2015 16:11:32 -0300 Subject: [Freeswitch-users] mod_opus bug? In-Reply-To: References: <54B95492.2090309@virtues.net> <54B957BC.4080405@virtues.net> Message-ID: <54B96264.1020305@virtues.net> It's not a bug in Freeswitch, it's the Comrex losing the usedtx=0, not Freeswitch. I forgot that I had changed that myself before and with the latest mod_opus update I switched back to the vanilla version. My bad, sorry! It would be nice to have more config options for opus to work around such things. :) On 16.01.2015 15:39, Anthony Minessale wrote: > Looking at your issue however I don't see dtx being enabled when I lab up what you described. > > Try this in vars.xml to disable CNG completely. > > > > > > > On Fri, Jan 16, 2015 at 12:28 PM, Anthony Minessale > > wrote: > > Its better to ask by filing the jira its one click to close it and if there is a real issue we need logs etc > attached and something to reference if we push a fix. > > > > On Fri, Jan 16, 2015 at 12:26 PM, Thomas Auge > wrote: > > I figured I'd ask for an opinion before filing a bogus bug. :P > > > On 16.01.2015 15:16, Anthony Minessale wrote: > > Its also a bug to send bugs to the mailing list instead ofhttp://jira.freeswitch.org > > > > > > On Fri, Jan 16, 2015 at 12:12 PM, Thomas Auge >> wrote: > > > > Hello list, > > > > I noticed that in an opus call where the a-leg sends usedtx=0, it is properly forwarded to the b-leg, which in this case > > also correctly responds with usedtx=0. However, the local SDP for the a-leg then does not have usedtx in the fmtp at > > all, and it seems to be enabled. > > > > My C skills are limited, but could this be a bug: > > > > static opus_codec_settings_t default_codec_settings = { > > .... > > /*.usedtx */ 1, > > .... > > }; > > > > By default, usedtx is enabled, correct? > > > > However, in > > > > static char *gen_fmtp(opus_codec_settings_t *settings, switch_memory_pool_t *pool) > > > > there's this: > > > > if (settings->usedtx) { > > snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "usedtx=1; "); > > } > > > > > > Am I interpreting this correctly that if usedtx is zero, it is omitted in the fmtp line, but then enabled, because > > that's the default setting? > > > > Problem is that Chrome can't handle DTX and generates comfort noise every time it kicks in. > > > > Cheers! > > > > Thomas > > > > P.S.: If my interpretation is correct, it'll also affect useinbandfec. > > > > _________________________________________________________________________ > > 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 > > > > > > > > > > -- > > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > > > ?http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > > ? irc.freenode.net #freeswitch ? _http://freeswitch.org/g+_ > > > > ClueCon Weekly Development Call > > ? sip:888 at conference.freeswitch.org > > ? +19193869900 > > > > > > > > > _________________________________________________________________________ > > 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 > > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? _http://freeswitch.org/g+_ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? _http://freeswitch.org/g+_ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > > _________________________________________________________________________ > 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 > From nbhatti at gmail.com Fri Jan 16 22:19:35 2015 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Fri, 16 Jan 2015 22:19:35 +0300 Subject: [Freeswitch-users] User tagging against IP ACL Message-ID: When a call is authenticated via an ACL, is there a way we can flag the user who?s being authenticated? This is needed to billing/identification purpose. Since the user does not exists in the directory and only allowed by an IP ACL, how can we know which user the IP belongs to? ? Thanks, Muhammad Naseer Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/a8679e12/attachment-0001.html From david.villasmil at gmail.com Fri Jan 16 22:22:34 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Fri, 16 Jan 2015 20:22:34 +0100 Subject: [Freeswitch-users] Asterisk's nat=yes feature In-Reply-To: References: Message-ID: I've actually *never* had a nat problem with FS.. it's pretty awesome at nat... On Jan 16, 2015 7:58 PM, "Vik Killa" wrote: > Are you having NAT issues? > FS is pretty good at auto-fixing NAT > > On Fri, Jan 16, 2015 at 12:58 PM, Zolt?n Szab? wrote: > >> Hi, >> >> I have a scenario where my softphone is behind nat and freeswitch is not >> behind nat. With asterisk I could make the audio work with nat=yes in the >> user's configuration. >> >> What is the settings for this on freeswitch? >> >> I have tried everything from the following pages: >> http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA >> http://wiki.freeswitch.org/wiki/NAT >> http://wiki.freeswitch.org/wiki/NAT_Traversal >> http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios >> >> (btw these are linking back to each other) >> >> Is it not that just easy as net=yes in asterisk? >> >> Many thanks, >> Zoltan >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/68495bec/attachment.html From krice at freeswitch.org Fri Jan 16 22:24:24 2015 From: krice at freeswitch.org (Ken Rice) Date: Fri, 16 Jan 2015 13:24:24 -0600 Subject: [Freeswitch-users] Asterisk's nat=yes feature In-Reply-To: Message-ID: Theres not a NAT=yes setting FS ?just works? in this area.. I?m guessing you missed the big popup box about that particular wiki being deprecated and containing a lot of stale information... Try checking out https://freeswitch.org/confluence/display/FREESWITCH/Auto+Nat On 1/16/15, 11:58 AM, "Zolt?n Szab?" wrote: > Hi, > > I have a scenario where my softphone is behind nat and freeswitch is not > behind nat. With asterisk I could make the audio work with nat=yes in the > user's configuration. > > What is the settings for this on freeswitch? > > I have tried everything from the following pages: > http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA? > http://wiki.freeswitch.org/wiki/NAT? > http://wiki.freeswitch.org/wiki/NAT_Traversal? > http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios? > > (btw these are linking back to each other) > > Is it not that just easy as net=yes in asterisk? > > Many thanks, > Zoltan > > > _________________________________________________________________________ > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/64b8a844/attachment.html From brian at freeswitch.org Fri Jan 16 22:25:07 2015 From: brian at freeswitch.org (Brian West) Date: Fri, 16 Jan 2015 13:25:07 -0600 Subject: [Freeswitch-users] User tagging against IP ACL In-Reply-To: References: Message-ID: see directory/default/brian.xml This builds the 'domains' acl from the cidr= tag on the user, so you create your users, it will call set_user for you applying any variables just as if they used digest auth. On Fri, Jan 16, 2015 at 1:19 PM, Muhammad Naseer Bhatti wrote: > > When a call is authenticated via an ACL, is there a way we can flag the > user who?s being authenticated? This is needed to billing/identification > purpose. Since the user does not exists in the directory and only allowed > by an IP ACL, how can we know which user the IP belongs to? > > ? > Thanks, > Muhammad Naseer Bhatti > > _________________________________________________________________________ > 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/20150116/6082b75c/attachment.html From anthony.minessale at gmail.com Fri Jan 16 22:26:26 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 16 Jan 2015 13:26:26 -0600 Subject: [Freeswitch-users] mod_opus bug? In-Reply-To: <54B96264.1020305@virtues.net> References: <54B95492.2090309@virtues.net> <54B957BC.4080405@virtues.net> <54B96264.1020305@virtues.net> Message-ID: I labbed it up based on your description and added logging to where it turns on dtx and it never turned it on. I always turn off the CNG because it creates delay in the chrome side. There was also a buffer overflow discovered in opus that caused such glitches so if you are not running the latest you can try that as the overflow could also lead to crashes and dropped calls. On Fri, Jan 16, 2015 at 1:11 PM, Thomas Auge wrote: > It's not a bug in Freeswitch, it's the Comrex losing the usedtx=0, not > Freeswitch. I forgot that I had changed that > myself before and with the latest mod_opus update I switched back to the > vanilla version. > > My bad, sorry! > > It would be nice to have more config options for opus to work around such > things. :) > > > On 16.01.2015 15:39, Anthony Minessale wrote: > > Looking at your issue however I don't see dtx being enabled when I lab > up what you described. > > > > Try this in vars.xml to disable CNG completely. > > > > > > > > > > > > > > On Fri, Jan 16, 2015 at 12:28 PM, Anthony Minessale < > anthony.minessale at gmail.com > > > wrote: > > > > Its better to ask by filing the jira its one click to close it and > if there is a real issue we need logs etc > > attached and something to reference if we push a fix. > > > > > > > > On Fri, Jan 16, 2015 at 12:26 PM, Thomas Auge > wrote: > > > > I figured I'd ask for an opinion before filing a bogus bug. :P > > > > > > On 16.01.2015 15:16, Anthony Minessale wrote: > > > Its also a bug to send bugs to the mailing list instead > ofhttp://jira.freeswitch.org > > > > > > > > > On Fri, Jan 16, 2015 at 12:12 PM, Thomas Auge < > auge at virtues.net >> wrote: > > > > > > Hello list, > > > > > > I noticed that in an opus call where the a-leg sends > usedtx=0, it is properly forwarded to the b-leg, which in this case > > > also correctly responds with usedtx=0. However, the local > SDP for the a-leg then does not have usedtx in the fmtp at > > > all, and it seems to be enabled. > > > > > > My C skills are limited, but could this be a bug: > > > > > > static opus_codec_settings_t default_codec_settings = { > > > .... > > > /*.usedtx */ 1, > > > .... > > > }; > > > > > > By default, usedtx is enabled, correct? > > > > > > However, in > > > > > > static char *gen_fmtp(opus_codec_settings_t *settings, > switch_memory_pool_t *pool) > > > > > > there's this: > > > > > > if (settings->usedtx) { > > > snprintf(buf + strlen(buf), sizeof(buf) - > strlen(buf), "usedtx=1; "); > > > } > > > > > > > > > Am I interpreting this correctly that if usedtx is zero, > it is omitted in the fmtp line, but then enabled, because > > > that's the default setting? > > > > > > Problem is that Chrome can't handle DTX and generates > comfort noise every time it kicks in. > > > > > > Cheers! > > > > > > Thomas > > > > > > P.S.: If my interpretation is correct, it'll also affect > useinbandfec. > > > > > > > _________________________________________________________________________ > > > 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 FreeSWITCH-users at lists.freeswitch.org> > > 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 > > > > > > > > > > > > > > > -- > > > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > > > > > ?http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > > > ? irc.freenode.net < > http://irc.freenode.net> #freeswitch ? _http://freeswitch.org/g+_ > > > > > > ClueCon Weekly Development Call > > > ? sip:888 at conference.freeswitch.org sip%3A888 at conference.freeswitch.org> > > sip%253A888 at conference.freeswitch.org>> ? +19193869900 > > > > > > > > > > > > > > > _________________________________________________________________________ > > > 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 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 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 > > > > > > > > > > -- > > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > > ? irc.freenode.net #freeswitch ? _ > http://freeswitch.org/g+_ > > > > ClueCon Weekly Development Call > > ? sip:888 at conference.freeswitch.org sip%3A888 at conference.freeswitch.org> ? +19193869900 > > > > > > > > > > -- > > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > > ? irc.freenode.net #freeswitch ? _ > http://freeswitch.org/g+_ > > > > ClueCon Weekly Development Call > > ? sip:888 at conference.freeswitch.org sip%3A888 at conference.freeswitch.org> ? +19193869900 > > > > > > > > _________________________________________________________________________ > > 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/2875a6a3/attachment-0001.html From abalashov at evaristesys.com Fri Jan 16 22:59:38 2015 From: abalashov at evaristesys.com (Alex Balashov) Date: Fri, 16 Jan 2015 14:59:38 -0500 Subject: [Freeswitch-users] Polycom busy lamp with IVR? In-Reply-To: <54B93099.5070000@evaristesys.com> References: <54B93099.5070000@evaristesys.com> Message-ID: <54B96DAA.2040909@evaristesys.com> I'm thinking this can probably be done by calling API functions, for instance via ESL, that have the effect of lighting up a phone's busy lamp. I'm just not sure how to do it, and the documentation on available BLF-influencing presence events is sparse. On 01/16/2015 10:39 AM, Alex Balashov wrote: > Hi, > > Sorry if it's a mundane question: > > I've got two Polycom phones sharing a registration appearance and SLA / > busy lamp works fine with them. > > What I would like is to create a second line appearance whose busy lamp > lights up when a call comes into our main IVR. Is this possible? How > would I do it? Are there presence hints that can be arbitrarily directed > from the dialplan? I wasn't able to discern this from the Polycom > presence or general Polycom documentation. > > Thank you! > > -- Alex > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ From abalashov at evaristesys.com Fri Jan 16 23:17:46 2015 From: abalashov at evaristesys.com (Alex Balashov) Date: Fri, 16 Jan 2015 15:17:46 -0500 Subject: [Freeswitch-users] Polycom busy lamp with IVR? In-Reply-To: <54B96DAA.2040909@evaristesys.com> References: <54B93099.5070000@evaristesys.com> <54B96DAA.2040909@evaristesys.com> Message-ID: <54B971EA.2020104@evaristesys.com> I've tried sending this event: https://wiki.freeswitch.org/wiki/PRESENCE_IN_event_example e.g. sendevent PRESENCE_IN proto: sip from: monitoredLine at pbx.corp.evaristesys.com login: monitoredLine at pbx.corp.evaristesys.com event_type: presence alt_event_type: dialog Presence-Call-Direction: inbound answer-state: confirmed ... where my handset has SUBSCRIBE'd to monitoredLine at pbx.corp.evaristesys.com. I am getting NOTIFYs, but without any stanzas inside the event package wrapper, i.e. it's the reset stanza. Content-Type: application/dialog-info+xml Content-Length: 169 Am I missing something about what I ought to set answer-state to? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ From abalashov at evaristesys.com Fri Jan 16 23:36:07 2015 From: abalashov at evaristesys.com (Alex Balashov) Date: Fri, 16 Jan 2015 15:36:07 -0500 Subject: [Freeswitch-users] SOLVED: Polycom busy lamp with IVR? In-Reply-To: <54B971EA.2020104@evaristesys.com> References: <54B93099.5070000@evaristesys.com> <54B96DAA.2040909@evaristesys.com> <54B971EA.2020104@evaristesys.com> Message-ID: <54B97637.3050606@evaristesys.com> I went trawling through some old mailing list threads and discovered that the solution is to use a channel bridge through loopback: This will link the presence entity 6789540670 at pbx.corp.evaristesys.com to the state of the bridge. A static BLF directory key can then be provisioned on the Polycom and voila! -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ From david.villasmil at gmail.com Sat Jan 17 00:26:00 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Fri, 16 Jan 2015 22:26:00 +0100 Subject: [Freeswitch-users] thousands of zombie calls Message-ID: Hello guys, I have a FS which when i execute on the cli: callcenter_config queue list members agents_queue I got THOUSANDS of "Abandoned" calls like this: agents_queue|single_box|e2685c96-6f85-4024-bd9a-5b2ac5b413d1||123456|Outbound Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned agents_queue|single_box|7bc18797-2913-476b-90ed-3f6905b99eda||123456|Outbound Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned agents_queue|single_box|96826416-0a90-4012-a68f-083534247424||123456|Outbound Call|0|1421429675|0|0|1421429676|0|0|||Abandoned This bot is used to generate calls to a test server via originate. But at this time the application which send the calls out is not even running! Furthermore, I stopped and started FS and the calls come back! WTF?? Where could all these calls come from?? Thanks for your help David -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/65f2dbb6/attachment.html From david.villasmil at gmail.com Sat Jan 17 00:41:02 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Fri, 16 Jan 2015 22:41:02 +0100 Subject: [Freeswitch-users] thousands of zombie calls In-Reply-To: References: Message-ID: What's even weirder, if I try to uuid_kill one of those channels, fs responds "no such channel".. so there's really no call there! Anyone? On Fri, Jan 16, 2015 at 10:26 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Hello guys, > > I have a FS which when i execute on the cli: > > callcenter_config queue list members agents_queue > > I got THOUSANDS of "Abandoned" calls like this: > > agents_queue|single_box|e2685c96-6f85-4024-bd9a-5b2ac5b413d1||123456|Outbound > Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned > agents_queue|single_box|7bc18797-2913-476b-90ed-3f6905b99eda||123456|Outbound > Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned > agents_queue|single_box|96826416-0a90-4012-a68f-083534247424||123456|Outbound > Call|0|1421429675|0|0|1421429676|0|0|||Abandoned > > This bot is used to generate calls to a test server via originate. But at > this time the application which send the calls out is not even running! > > Furthermore, I stopped and started FS and the calls come back! WTF?? > Where could all these calls come from?? > > Thanks for your help > > David > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/a9eba35a/attachment.html From karl-theo_hofer at inteli-sim.com Sat Jan 17 00:51:33 2015 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Fri, 16 Jan 2015 22:51:33 +0100 Subject: [Freeswitch-users] how to break a synchron execute in ESL Message-ID: <54B987E5.10408@inteli-sim.com> Hi there we have a short voicemail set up and we like to break the synchronise be a following asynchron execute, so the the voicemail-owner is not forced to listen to the complete menu. how can we do this? bacs XAKK $con->setEventLock("1"); $con->execute("playback", "$sound_base/Welcome to your Message Centre.wav", "$leg_uuid"); $con->execute("playback", "$sound_base/You_have_no_message.wav", "$leg_uuid"); $con->execute("playback", "$sound_base/To listen to your personal welcome greeting press 1 to save your personal greeting press 2 if you want to replace your current personal greeting press 3 if you like to delete.wav", "$leg_uuid"); $con->setEventLock("0"); $con->execute("break", "playback", $leg_uuid); -- With best regards Karl Theo Hofer M: +46 7030 22178 E: karl-theo_hofer at inteli-sim.com -------------- next part -------------- An HTML attachment was scrubbed... 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Name: photo.jpg Type: image/jpeg Size: 1380 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/b39b2547/attachment.jpg From karl-theo_hofer at inteli-sim.com Sat Jan 17 00:55:10 2015 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Fri, 16 Jan 2015 22:55:10 +0100 Subject: [Freeswitch-users] Aalternative to PlayAndGetDigits Message-ID: <54B988BE.8020407@inteli-sim.com> HI There does someone of you have a alternative to the play and get digit function problem is that all dtmf events are released. which kills our scripts. -- With best regards Karl Theo Hofer M: +46 7030 22178 E: karl-theo_hofer at inteli-sim.com From adam.ben.ayoun1 at gmail.com Fri Jan 16 23:47:54 2015 From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun) Date: Fri, 16 Jan 2015 22:47:54 +0200 Subject: [Freeswitch-users] SIP over Websocket VS SIP over TCP In-Reply-To: References: Message-ID: Update: We integrated pjsip with WebRTC on iOS, we are using pjsip for the signalling part and passing the SDP back and forth to and from WebRTC. Our current issue is that after initializing the call we can't hear any audio (should hear moh at this point), everything looks OK on Freeswitch until we start seeing alot of those errors: 2015-01-16 20:25:31.111260 [ERR] mod_opus.c:418 Decoder Error: corrupted stream fs:4080 plc:0! 2015-01-16 20:25:31.111260 [ERR] switch_core_io.c:697 Codec RAW Signed Linear (16 bit) decoder error! [9] Any idea what might be wrong here? Thanks, Adam On 15 January 2015 at 15:27, Adam Ben-Ayoun wrote: > In our case, all clients are calling to the conference on FreeSwitch, > hence, no registration is required at the moment, but it definitely opens > up other options for future. > > On 10 January 2015 at 20:38, Adam Ben-Ayoun > wrote: > >> Great to hear that. Thanks again. >> >> On 10 January 2015 at 20:35, Carlos Ruiz D?az >> wrote: >> >>> WebRTC doesn't specify a signalling protocol. This means that you can >>> use SIP over any transport you want to carry the webRTC enabled SDP. >>> >>> FS will receive the SDP, detect that has a RTP/SAVPF profile and start >>> handling it accordingly. >>> >>> Take for example Jitsi or IMSDroid, they both support webRTC and do SIP >>> over UDP/TCP/TLS. >>> >>> Regards, >>> Carlos >>> On Jan 10, 2015 12:08 PM, "Adam Ben-Ayoun" >>> wrote: >>> >>>> Thanks Anthony. I assume that means I can use SIP over TCP/TLS for >>>> signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work >>>> when communicating with FS (when stuff like fingerprint, etc)? >>>> >>>> On 10 January 2015 at 19:49, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> The WebRTC media engine is driven completely by the SDP, the transport >>>>> will not make any difference. >>>>> >>>>> >>>>> On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun < >>>>> adam.ben.ayoun1 at gmail.com> wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> We are developing a mobile client that will use the WebRTC media >>>>>> stack and Freeswitch as an MCU (only for conference calls). My question is, >>>>>> since we build a native app, can we use SIP over TCP for signalling? In >>>>>> other words, if Freeswitch receives the WebRTC kind of SDP, will it be able >>>>>> to communicate in the same way as if we were using the SIP over Websocket >>>>>> (the other Freeswitch option)? Any corner cases/considerations with this? >>>>>> Our goal is to avoid implementing SIP over Websocket on the client as much >>>>>> as possible. >>>>>> >>>>>> Thanks, >>>>>> Adam >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>>> >>>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>>> http://twitter.com/FreeSWITCH >>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>>> * >>>>> >>>>> ClueCon Weekly Development Call >>>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>> >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/e6362844/attachment.html From brian at freeswitch.org Sat Jan 17 00:58:53 2015 From: brian at freeswitch.org (Brian West) Date: Fri, 16 Jan 2015 15:58:53 -0600 Subject: [Freeswitch-users] Aalternative to PlayAndGetDigits In-Reply-To: <54B988BE.8020407@inteli-sim.com> References: <54B988BE.8020407@inteli-sim.com> Message-ID: How about you slow down, relax, stop sending two emails with opposite issues within a 3 minute window. Step back, What are you trying to do exactly? Not currently but overall what are your goals? On Fri, Jan 16, 2015 at 3:55 PM, kthofer wrote: > HI There > > does someone of you have a alternative to the play and get digit function > problem is that all dtmf events are released. > > which kills our scripts. > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > _________________________________________________________________________ > 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/20150116/698de263/attachment.html From mike at jerris.com Sat Jan 17 01:00:33 2015 From: mike at jerris.com (Michael Jerris) Date: Fri, 16 Jan 2015 17:00:33 -0500 Subject: [Freeswitch-users] Aalternative to PlayAndGetDigits In-Reply-To: <54B988BE.8020407@inteli-sim.com> References: <54B988BE.8020407@inteli-sim.com> Message-ID: <667C37A2-CFE3-4321-8DE6-186BC33A71D6@jerris.com> I don't understand the statement about the dtmf events getting released. > On Jan 16, 2015, at 4:55 PM, kthofer wrote: > > HI There > > does someone of you have a alternative to the play and get digit function > problem is that all dtmf events are released. > > which kills our scripts. > From karl-theo_hofer at inteli-sim.com Sat Jan 17 01:24:55 2015 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Fri, 16 Jan 2015 23:24:55 +0100 Subject: [Freeswitch-users] Aalternative to PlayAndGetDigits In-Reply-To: <667C37A2-CFE3-4321-8DE6-186BC33A71D6@jerris.com> References: <54B988BE.8020407@inteli-sim.com> <667C37A2-CFE3-4321-8DE6-186BC33A71D6@jerris.com> Message-ID: <54B98FB7.9010103@inteli-sim.com> Hi There guess i lost some text somewhere, sorry. If you are in a event for instance channel answer there you do play and get digits. Play and get digits blocks and will not release the channel answer event but listen for further DTMF signals. when play and get digits is finished, the channel answer event will get released and all dtmf events which were blocked in play and get digits will be received. can we get play and get digits to not generate events for the key presses that play and get digits already handled. With best regards Karl Theo Hofer Michael Jerris skrev den 2015-01-16 23:00: > I don't understand the statement about the dtmf events getting released. > >> On Jan 16, 2015, at 4:55 PM, kthofer wrote: >> >> HI There >> >> does someone of you have a alternative to the play and get digit function >> problem is that all dtmf events are released. >> >> which kills our scripts. >> > > _________________________________________________________________________ > 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 From mike at jerris.com Sat Jan 17 01:44:16 2015 From: mike at jerris.com (Michael Jerris) Date: Fri, 16 Jan 2015 17:44:16 -0500 Subject: [Freeswitch-users] Aalternative to PlayAndGetDigits In-Reply-To: <54B98FB7.9010103@inteli-sim.com> References: <54B988BE.8020407@inteli-sim.com> <667C37A2-CFE3-4321-8DE6-186BC33A71D6@jerris.com> <54B98FB7.9010103@inteli-sim.com> Message-ID: <9C9A935F-1127-4376-BE65-52321DDFDE39@jerris.com> Can you post a small example of exactly what you are doing. Most likely you don't want to be running that inside an event handler and blocking there. > On Jan 16, 2015, at 5:24 PM, kthofer wrote: > > Hi There > guess i lost some text somewhere, sorry. > > If you are in a event for instance channel answer > there you do play and get digits. > > > Play and get digits blocks and will not release the channel answer event > but listen for further DTMF signals. > > when play and get digits is finished, the channel answer event will get > released > and all dtmf events which were blocked in play and get digits > will be received. > > can we get play and get digits > to not generate events for the key presses that play and get digits > already handled. > > > > With best regards > > Karl Theo Hofer > > Michael Jerris skrev den 2015-01-16 23:00: >> I don't understand the statement about the dtmf events getting released. >> >>> On Jan 16, 2015, at 4:55 PM, kthofer wrote: >>> >>> HI There >>> >>> does someone of you have a alternative to the play and get digit function >>> problem is that all dtmf events are released. >>> >>> which kills our scripts. >>> From karl-theo_hofer at inteli-sim.com Sat Jan 17 01:57:48 2015 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Fri, 16 Jan 2015 23:57:48 +0100 Subject: [Freeswitch-users] Aalternative to PlayAndGetDigits In-Reply-To: <9C9A935F-1127-4376-BE65-52321DDFDE39@jerris.com> References: <54B988BE.8020407@inteli-sim.com> <667C37A2-CFE3-4321-8DE6-186BC33A71D6@jerris.com> <54B98FB7.9010103@inteli-sim.com> <9C9A935F-1127-4376-BE65-52321DDFDE39@jerris.com> Message-ID: <54B9976C.8030102@inteli-sim.com> Hi Michael Most likely you don't want to be running that inside an event handler and blocking there. thats exactly we like to avoid!! How can we avoid it we used play and get digits to start and stop voice mail announcements but stopped it and did start working with dtmf events and $con->execute("playback"... but this has issues because we can not break synchronety in ESL so we are thinking of going back to play and get digits or do you have some other alternative? With best regards Karl Theo Hofer Michael Jerris skrev den 2015-01-16 23:44: > Can you post a small example of exactly what you are doing. Most likely you don't want to be running that inside an event handler and blocking there. > >> On Jan 16, 2015, at 5:24 PM, kthofer wrote: >> >> Hi There >> guess i lost some text somewhere, sorry. >> >> If you are in a event for instance channel answer >> there you do play and get digits. >> >> >> Play and get digits blocks and will not release the channel answer event >> but listen for further DTMF signals. >> >> when play and get digits is finished, the channel answer event will get >> released >> and all dtmf events which were blocked in play and get digits >> will be received. >> >> can we get play and get digits >> to not generate events for the key presses that play and get digits >> already handled. >> >> >> >> With best regards >> >> Karl Theo Hofer >> >> Michael Jerris skrev den 2015-01-16 23:00: >>> I don't understand the statement about the dtmf events getting released. >>> >>>> On Jan 16, 2015, at 4:55 PM, kthofer wrote: >>>> >>>> HI There >>>> >>>> does someone of you have a alternative to the play and get digit function >>>> problem is that all dtmf events are released. >>>> >>>> which kills our scripts. >>>> > > _________________________________________________________________________ > 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 From italorossib at gmail.com Sat Jan 17 05:20:26 2015 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Fri, 16 Jan 2015 23:20:26 -0300 Subject: [Freeswitch-users] thousands of zombie calls In-Reply-To: References: Message-ID: Set abandoned-resume-allowed to false Em 16/01/2015 18:42, "David Villasmil Govea" escreveu: > What's even weirder, if I try to uuid_kill one of those channels, fs > responds "no such channel".. so there's really no call there! > > Anyone? > > On Fri, Jan 16, 2015 at 10:26 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> Hello guys, >> >> I have a FS which when i execute on the cli: >> >> callcenter_config queue list members agents_queue >> >> I got THOUSANDS of "Abandoned" calls like this: >> >> agents_queue|single_box|e2685c96-6f85-4024-bd9a-5b2ac5b413d1||123456|Outbound >> Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned >> agents_queue|single_box|7bc18797-2913-476b-90ed-3f6905b99eda||123456|Outbound >> Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned >> agents_queue|single_box|96826416-0a90-4012-a68f-083534247424||123456|Outbound >> Call|0|1421429675|0|0|1421429676|0|0|||Abandoned >> >> This bot is used to generate calls to a test server via originate. But at >> this time the application which send the calls out is not even running! >> >> Furthermore, I stopped and started FS and the calls come back! WTF?? >> Where could all these calls come from?? >> >> Thanks for your help >> >> David >> >> >> -- >> DVG >> >> -- >> Imagination is more important than knowledge >> Albert Einstein >> > > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/0f26143b/attachment.html From luis.daniel.lucio at gmail.com Sat Jan 17 06:13:12 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Fri, 16 Jan 2015 22:13:12 -0500 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? In-Reply-To: References: <54B93802.2020906@gmail.com> Message-ID: FusionPBX does billing, check https://okay.com.mx/en/entrepreneurs/billing-for-fusionpbx-with-freeswitch.html On Jan 16, 2015 1:57 PM, "Vik Killa" wrote: > I prefer git clone master branch. > > > On Fri, Jan 16, 2015 at 1:42 PM, Sina Owolabi > wrote: > >> Quick question though.. Which is best, git clone freeswitch or to use the >> Debian repo? >> >> >> On Fri Jan 16 2015 at 6:52:44 PM Sina Owolabi >> wrote: >> >>> Thanks everyone. Maybe I should switch to Debian after all. >>> >>> On Fri Jan 16 2015 at 5:14:44 PM DanB wrote: >>> >>>> Hi Sina, >>>> >>>> CGRateS is a billing engine and I think it has what you are looking for. >>>> It does not touch/influence your FreeSWITCH configuration files. It can >>>> serve you as simple CDR server or complete charging solution with some >>>> pretty fancy prepaid/postpaid scenarios. Once calculated, CDRs will be >>>> accessible to be exported or retrieved via JSON-RPC calls. >>>> >>>> DanB >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> 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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/1a617760/attachment.html From luis.daniel.lucio at gmail.com Sat Jan 17 06:51:28 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Fri, 16 Jan 2015 22:51:28 -0500 Subject: [Freeswitch-users] robo caller In-Reply-To: References: Message-ID: Install FusionPBX and look into 'broadcast calls' On Jan 16, 2015 6:40 AM, "Areski" wrote: > Yes, it's via this mod_amd module. > > On Thu, Jan 15, 2015 at 4:19 AM, Michael Collins > wrote: > >> Areski, >> >> Does Newfies support mod_com_amd? Just curious how you handle machines. >> -MC >> >> On Wed, Jan 14, 2015 at 8:44 AM, Areski wrote: >> >>> Newfies-Dialer (http://www.newfies-dialer.org/) might help and >>> obviously it's built on top of FreeSWITCH. >>> We built a flexible module for appointment reminders: >>> http://docs.newfies-dialer.org/en/latest/user-guide-doc/appointment.html >>> >>> If you want to code this yourself, we use ESL ( >>> https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library) >>> to originate the calls and Lua to build the IVR part ( >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_lua). >>> >>> On Wed, Jan 14, 2015 at 5:37 PM, Moishe Grunstein >>> wrote: >>> >>>> You can also have a look at http://www.newfies-dialer.org/ >>>> >>>> >>>> >>>> Thanks, >>>> >>>> >>>> >>>> Moishe Grunstein >>>> >>>> Tornado Computer Systems, Inc. >>>> >>>> 212.400.7650 888.IPPBX.US >>>> *Service Request Email: support at nysolutions.com >>>> * >>>> >>>> [image: cid:image001.jpg at 01C72F94.9EE45D60] >>>> >>>> >>>> Computer Networking * Managed Services * IP Video Surveillance * >>>> Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * >>>> Network Security * Site Surveys * CMS >>>> >>>> >>>> >>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David >>>> Villasmil Govea >>>> *Sent:* Wednesday, January 14, 2015 11:31 AM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* Re: [Freeswitch-users] robo caller >>>> >>>> >>>> >>>> That's fairly simple to implement. Look into at mod_amd, orignate and >>>> ivr. >>>> >>>> On Jan 14, 2015 5:28 PM, "Sean Devoy" wrote: >>>> >>>> Does anyone have a sample RoboCaller script? Perhaps I am using the >>>> wrong name and that is why I can't find one. I have a doctor's office that >>>> wants to automate the reminder calls about appointments to their patients. >>>> >>>> >>>> >>>> I am curious how people handle answering machine detection as well? >>>> >>>> >>>> >>>> Thanks, >>>> >>>> Sean >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>> >>> >>> >>> -- >>> Kind regards, >>> /Areski >>> >>> ---- >>> Arezqui Belaid, >>> Founder at Star2Billing (www.star2billing.com) >>> >>> Tel: +34650784355 >>> Twitter: http://twitter.com/areskib >>> LinkedIn: http://www.linkedin.com/in/areski >>> >>> _________________________________________________________________________ >>> 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 >> > > > > -- > Kind regards, > /Areski > > ---- > Arezqui Belaid, > Founder at Star2Billing (www.star2billing.com) > > Tel: +34650784355 > Twitter: http://twitter.com/areskib > LinkedIn: http://www.linkedin.com/in/areski > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/0c2b77a6/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150116/0c2b77a6/attachment-0001.jpg From notify.sina at gmail.com Sat Jan 17 11:36:03 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Sat, 17 Jan 2015 08:36:03 +0000 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? References: Message-ID: Thanks Luis, Dan, Vik, everyone! I'm cloning the master. I'm more interested in CDR than billing actually, though I now think one leads to the other. On Fri Jan 16 2015 at 5:03:33 PM Moishe Grunstein wrote: > FusionPBX does cdr?s however not billing, if you are looking for a > billing solution see https://wiki.freeswitch.org/wiki/Billing > > > > Anything should work on Cenos, however Debian is currently the recommended > Freeswitch platform. > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: **support at nysolutions.com* > > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sina Owolabi > *Sent:* Friday, January 16, 2015 9:47 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Any CDR tool for Freeswitch that won't > remove/overwrite existing config? > > > > Hi List! > > I'm checking out GUI billing and CDR records applications for FreeSWITCH. > Most I've tried (bluebox, fusion, vbilling ) seem to either want to delete > my existing configuration , or install their own Freeswitch , don't like > CentOS, and totally ignore my own which I'm quite partial to. > > I wonder if there are any suggestions of CDR applications that will just > fit in to what's already there? I'm running the latest master on a CentOS > 64bit server. > > Thanks in advance ! > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/52ab8640/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/52ab8640/attachment.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/52ab8640/attachment-0001.jpg From babak.freeswitch at gmail.com Sat Jan 17 15:54:03 2015 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Sat, 17 Jan 2015 16:24:03 +0330 Subject: [Freeswitch-users] Startup script is not respecting real time priority Message-ID: Hi I'm using startup script supplied in repo for centos 6.5 and setting the -rp like this: PROG_NAME=freeswitch PID_FILE=/usr/local/freeswitch/run/freeswitch.pid FS_USER=freeswitch FS_FILE=/usr/local/freeswitch/bin/freeswitch FS_HOME=/usr/local/freeswitch/run LOCK_FILE=/var/lock/subsys/freeswitch *FREESWITCH_ARGS="-nc -rp"* RETVAL=0 but freeswitch is always running with default priority as seen in top: 16263 freeswit 39 19 875m 30m 7416 S 1.0 13.0 0:17.44 freeswitch and in "ps aux|grep freeswitch" I see: 500 16263 0.8 12.9 897004 31120 ? SNl 15:38 0:18 /usr/local/freeswitch/bin/freeswitch -nc -rp so it seems freeswitch is starting with proper args but priority is not set. If I run freeswitch as root in bash it is working will: runing ./freeswitch -nc -rp results in: 16550 root -2 -10 865m 28m 6988 S 6.3 12.2 0:00.52 freeswitch Should I change anything else in the script? or add any special permission to user "freeswitch"? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/2a09f508/attachment.html From david.villasmil at gmail.com Sat Jan 17 15:57:35 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Sat, 17 Jan 2015 13:57:35 +0100 Subject: [Freeswitch-users] thousands of zombie calls In-Reply-To: References: Message-ID: Thanks Italo, I'll try that and let you know. On Jan 17, 2015 3:21 AM, "?talo Rossi" wrote: > Set abandoned-resume-allowed to false > Em 16/01/2015 18:42, "David Villasmil Govea" > escreveu: > >> What's even weirder, if I try to uuid_kill one of those channels, fs >> responds "no such channel".. so there's really no call there! >> >> Anyone? >> >> On Fri, Jan 16, 2015 at 10:26 PM, David Villasmil Govea < >> david.villasmil at gmail.com> wrote: >> >>> Hello guys, >>> >>> I have a FS which when i execute on the cli: >>> >>> callcenter_config queue list members agents_queue >>> >>> I got THOUSANDS of "Abandoned" calls like this: >>> >>> agents_queue|single_box|e2685c96-6f85-4024-bd9a-5b2ac5b413d1||123456|Outbound >>> Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned >>> agents_queue|single_box|7bc18797-2913-476b-90ed-3f6905b99eda||123456|Outbound >>> Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned >>> agents_queue|single_box|96826416-0a90-4012-a68f-083534247424||123456|Outbound >>> Call|0|1421429675|0|0|1421429676|0|0|||Abandoned >>> >>> This bot is used to generate calls to a test server via originate. But >>> at this time the application which send the calls out is not even running! >>> >>> Furthermore, I stopped and started FS and the calls come back! WTF?? >>> Where could all these calls come from?? >>> >>> Thanks for your help >>> >>> David >>> >>> >>> -- >>> DVG >>> >>> -- >>> Imagination is more important than knowledge >>> Albert Einstein >>> >> >> >> >> -- >> DVG >> >> -- >> Imagination is more important than knowledge >> Albert Einstein >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/0708beea/attachment-0001.html From godson.g at gmail.com Sat Jan 17 16:52:39 2015 From: godson.g at gmail.com (Godson Gera) Date: Sat, 17 Jan 2015 19:22:39 +0530 Subject: [Freeswitch-users] how to break a synchron execute in ESL In-Reply-To: <54B987E5.10408@inteli-sim.com> References: <54B987E5.10408@inteli-sim.com> Message-ID: You can set platback_terminators channel var to a specific digit and announce that digit to caller if he wants to break the prompt. On Sat, Jan 17, 2015 at 3:21 AM, kthofer wrote: > Hi there > > we have a short voicemail set up > and we like to break the synchronise be a following asynchron execute, > so the the voicemail-owner is not forced to listen to the complete menu. > > how can we do this? > > > [image: bacs XAKK] > > $con->setEventLock("1"); > $con->execute("playback", "$sound_base/Welcome to your Message > Centre.wav", "$leg_uuid"); > $con->execute("playback", "$sound_base/You_have_no_message.wav", > "$leg_uuid"); > $con->execute("playback", "$sound_base/To listen to your personal welcome > greeting press 1 to save your personal greeting press 2 if you want to > replace your current personal greeting press 3 if you like to delete.wav", > "$leg_uuid"); > $con->setEventLock("0"); > $con->execute("break", "playback", $leg_uuid); > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > _________________________________________________________________________ > 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 > -- Thanks & Regards, Godson Gera VoIP Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/661080ca/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: photo.jpg Type: image/jpeg Size: 1380 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/661080ca/attachment.jpg From bote_radio at botecomm.com Sat Jan 17 17:27:42 2015 From: bote_radio at botecomm.com (Bote Man) Date: Sat, 17 Jan 2015 09:27:42 -0500 Subject: [Freeswitch-users] mobile to mobile over 3g/4g In-Reply-To: <779edd43-435f-4d7a-afdf-ced2118a6277@email.bluemailapp.com> References: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> <779edd43-435f-4d7a-afdf-ced2118a6277@email.bluemailapp.com> Message-ID: <005e01d03261$c177ba10$44672e30$@com> I have it on good authority that AT&T in the U.S. assigns public i.p. addresses to each device, but is subject to firewall rules whatever they might be. They offer "enterprise static i.p. service" that allows greater freedom, however. Sounds expensive and scary. I agree with Steven Ayre: I would not depend on having necessary connectivity. Bote From: Assaf Dahary Sent: Thursday, 15 January, 2015 11:47 Subject: Re: [Freeswitch-users] mobile to mobile over 3g/4g It seems that some mobile operators allow internal direct connectivity and some just block it. Direct connectivity will increase call quality and drop call latency. I should start mapping mobile operators by trying first bypass media on internal calls and if fail then to cancel it and call via FS . On Jan 15, 2015, at 18:32, Steven Ayre wrote: I would expect that while it might work on some operators you won't be able to rely on it working for all mobile networks. If your users aren't under your control and therefore aren't on networks you've had an opportunity to test this on then you'll probably see many calls with no audio. On 15 January 2015 at 13:33, Assaf Dahary wrote: Hi, I'm currently using bypass_media to make two mobile call each directly (RTP bypass FS) over the same wifi network. Once I detect that caller and callee are sharing the same WiFi network then I switch to bypass_media in dialplan (works well in big complex buildings). I would like also to use the same bypass_media mode when mobiles connect over 3g/4g data mobile network. I fully understand that both mobiles should be connected to the same mobile operator and should share a common subnet routing (without NAT) for them to 'see' each other private IP addr. Is it possible for two mobiles on the same mobile 3g/4g network to ping each other using their private IP addr? Should mobile operators allow this kind of direct peer connections (I couldn't ping internally with my Orange mobile operator)? Did anyone successfully make a direct bypass_media call between two sip mobiles over 3g/4g network? Regards assaf -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/690e2d35/attachment.html From john.nash778 at gmail.com Sat Jan 17 19:27:58 2015 From: john.nash778 at gmail.com (John Nash) Date: Sat, 17 Jan 2015 21:57:58 +0530 Subject: [Freeswitch-users] Sending call to outbound proxy Message-ID: I am trying to test a case where I am receiving call from opensips to freeswitch and need to route back to opensips again without changing Request URI. How can I do it? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/808609ed/attachment.html From luis.daniel.lucio at gmail.com Sat Jan 17 21:06:11 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sat, 17 Jan 2015 13:06:11 -0500 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? In-Reply-To: References: Message-ID: You will see that billing will drive you to lcr as well On Jan 17, 2015 3:37 AM, "Sina Owolabi" wrote: > Thanks Luis, Dan, Vik, everyone! > > I'm cloning the master. I'm more interested in CDR than billing actually, > though I now think one leads to the other. > > > On Fri Jan 16 2015 at 5:03:33 PM Moishe Grunstein > wrote: > >> FusionPBX does cdr's however not billing, if you are looking for a >> billing solution see https://wiki.freeswitch.org/wiki/Billing >> >> >> >> Anything should work on Cenos, however Debian is currently the >> recommended Freeswitch platform. >> >> >> >> Thanks, >> >> >> >> Moishe Grunstein >> >> Tornado Computer Systems, Inc. >> >> 212.400.7650 888.IPPBX.US >> *Service Request Email: **support at nysolutions.com* >> >> >> [image: cid:image001.jpg at 01C72F94.9EE45D60] >> >> Computer Networking * Managed Services * IP Video Surveillance * Network >> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >> Security * Site Surveys * CMS >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sina >> Owolabi >> *Sent:* Friday, January 16, 2015 9:47 AM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Any CDR tool for Freeswitch that won't >> remove/overwrite existing config? >> >> >> >> Hi List! >> >> I'm checking out GUI billing and CDR records applications for FreeSWITCH. >> Most I've tried (bluebox, fusion, vbilling ) seem to either want to delete >> my existing configuration , or install their own Freeswitch , don't like >> CentOS, and totally ignore my own which I'm quite partial to. >> >> I wonder if there are any suggestions of CDR applications that will just >> fit in to what's already there? I'm running the latest master on a CentOS >> 64bit server. >> >> Thanks in advance ! >> ____________________________________________________________ >> _____________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/2a36aaa4/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/2a36aaa4/attachment-0001.jpg From raphael.lechner at gmail.com Sat Jan 17 22:06:15 2015 From: raphael.lechner at gmail.com (Raphael Lechner) Date: Sat, 17 Jan 2015 20:06:15 +0100 Subject: [Freeswitch-users] voicemail silence_detection is always detecting silence - Solved In-Reply-To: References: Message-ID: Finally I resolved the problem by adding a sleep 1000 before voicemail. Now It works as expected. Raphael > On 16 Jan 2015, at 13:24, Raphael Lechner wrote: > > I missed to write the used FreeSWITCH version. I tried with 1.4.14 and 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit) > > Thank you > >> On 16 Jan 2015, at 11:09, Raphael Lechner > wrote: >> >> Hi, >> >> I configured an extension that first call for some seconds a phone and if nobody is picking up, the caller is hearing a playback and can press 1 for leaving a voicemail and 2 to get redirected to a mobile phone. >> The Problem is that in my test environment with a SIP Provider everything works fine after tuning the voicemail.conf.xml to >> >> >> > After that I can record a message and that works as expected. >> Is there a way do disable the silence_detection or any hint what I can change? >> >> I tried changing the silence-threshold to 1,50 and silence-hits to 300,30000 but nothing has changed >> >> Debug Log >> https://pastebin.freeswitch.org/23851 >> >> The called python script: >> def handler(session, args): >> voicemail = args.split(' ')[0] >> dtmf_pressed = args.split(' ')[1] >> forward_number = args.split(' ')[2] >> callerid = session.getVariable("caller_id_number") >> callername = session.getVariable("caller_id_name").lstrip() >> >> if dtmf_pressed == '1': >> send_sms('377XXXXXXX?,?New Voicemail from %s %s' % (callername, callerid)) >> session.execute("export", "skip_greeting=true") >> session.execute("export", "skip_instructions=true") >> session.execute("answer") >> session.execute("voicemail", "default 192.168.17.252 10?) >> #session.execute("bridge", "loopback/app=voicemail:default %s %s" % (conf['network']['ip'],voicemail)) >> elif dtmf_pressed == '2': >> consoleLog( "info", "Call is forwarded to %s\n" % forward_number) >> session.transfer(forward_number, "XML", "default") >> else: >> consoleLog( "info", "DTMF received is %s and not 1 or 2.Hangup Call\n" % (dtmf_pressed)) >> session.hangup() >> >> Thank you, >> Raphael > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/23891113/attachment.html From notify.sina at gmail.com Sun Jan 18 00:29:49 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Sat, 17 Jan 2015 21:29:49 +0000 Subject: [Freeswitch-users] How to use collected variables in mod_curl in lua script Message-ID: Hi List Newb again. I am trying to write a script that will use variables passed to the script from the dialplan in a http GET or POST URL call. Please what is the proper way of doing this? I've tried a few things and they dont work: variable1 = argv[1]; variable2 = argv[2]; In the script proper: web_url = ("http://webserver//index.php?u=%s&var2=%s", variable1, variable2); session:execute("curl", web_url); Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/92012855/attachment.html From adahary at gmail.com Sun Jan 18 01:23:16 2015 From: adahary at gmail.com (Assaf Dahary) Date: Sun, 18 Jan 2015 00:23:16 +0200 Subject: [Freeswitch-users] mobile to mobile over 3g/4g In-Reply-To: <005e01d03261$c177ba10$44672e30$@com> References: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> <779edd43-435f-4d7a-afdf-ced2118a6277@email.bluemailapp.com> <005e01d03261$c177ba10$44672e30$@com> Message-ID: I desided to check each mobile operator network by trying first bypass mode and if it fails then redial automaticalky with proxy mode (and not trying it again with this operator). For those mobile network (fail bypass) users the alternatives can be: - buying static public ip from operator. - setting up mobile operator VPN? group with public access to FS. - setting up common local VPN server to share direct connectivity for users group. On Jan 17, 2015, 16:30, at 16:30, Bote Man wrote: >I have it on good authority that AT&T in the U.S. assigns public i.p. >addresses to each device, but is subject to firewall rules whatever >they might be. > > > >They offer "enterprise static i.p. service" that allows greater >freedom, however. Sounds expensive and scary. > > > >I agree with Steven Ayre: I would not depend on having necessary >connectivity. > > > >Bote > > > > > >From: Assaf Dahary >Sent: Thursday, 15 January, 2015 11:47 >Subject: Re: [Freeswitch-users] mobile to mobile over 3g/4g > >It seems that some mobile operators allow internal direct connectivity >and some just block it. > >Direct connectivity will increase call quality and drop call latency. > >I should start mapping mobile operators by trying first bypass media on >internal calls and if fail then to cancel it and call via FS . > >On Jan 15, 2015, at 18:32, Steven Ayre wrote: > >I would expect that while it might work on some operators you won't be >able to rely on it working for all mobile networks. > > > >If your users aren't under your control and therefore aren't on >networks you've had an opportunity to test this on then you'll probably >see many calls with no audio. > > > > > > > >On 15 January 2015 at 13:33, Assaf Dahary wrote: > >Hi, > >I'm currently using bypass_media to make two mobile call each directly >(RTP bypass FS) over the same wifi network. > >Once I detect that caller and callee are sharing the same WiFi network >then I switch to bypass_media in dialplan (works well in big complex >buildings). > >I would like also to use the same bypass_media mode when mobiles >connect over 3g/4g data mobile network. > >I fully understand that both mobiles should be connected to the same >mobile operator and should share a common subnet routing (without NAT) >for them to 'see' each other private IP addr. > >Is it possible for two mobiles on the same mobile 3g/4g network to ping >each other using their private IP addr? > >Should mobile operators allow this kind of direct peer connections (I >couldn't ping internally with my Orange mobile operator)? > >Did anyone successfully make a direct bypass_media call between two sip >mobiles over 3g/4g network? > >Regards > >assaf > > > > >------------------------------------------------------------------------ > >_________________________________________________________________________ >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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/e0faad77/attachment-0001.html From kris at kriskinc.com Sun Jan 18 02:38:43 2015 From: kris at kriskinc.com (Kristian Kielhofner) Date: Sat, 17 Jan 2015 18:38:43 -0500 Subject: [Freeswitch-users] mobile to mobile over 3g/4g In-Reply-To: References: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> <779edd43-435f-4d7a-afdf-ced2118a6277@email.bluemailapp.com> <005e01d03261$c177ba10$44672e30$@com> Message-ID: You want ICE. On Saturday, January 17, 2015, Assaf Dahary wrote: > I desided to check each mobile operator network by trying first bypass > mode and if it fails then redial automaticalky with proxy mode (and not > trying it again with this operator). > > For those mobile network (fail bypass) users the alternatives can be: > - buying static public ip from operator. > - setting up mobile operator VPN group with public access to FS. > - setting up common local VPN server to share direct connectivity for > users group. > On Jan 17, 2015, at 16:30, Bote Man > wrote: >> >> I have it on good authority that AT&T in the U.S. assigns public i.p. >> addresses to each device, but is subject to firewall rules whatever they >> might be. >> >> >> >> They offer "enterprise static i.p. service" that allows greater freedom, >> however. Sounds expensive and scary. >> >> >> >> I agree with Steven Ayre: I would not depend on having necessary >> connectivity. >> >> >> >> Bote >> >> >> >> >> >> *From:* Assaf Dahary >> *Sent:* Thursday, 15 January, 2015 11:47 >> *Subject:* Re: [Freeswitch-users] mobile to mobile over 3g/4g >> >> It seems that some mobile operators allow internal direct connectivity >> and some just block it. >> >> Direct connectivity will increase call quality and drop call latency. >> >> I should start mapping mobile operators by trying first bypass media on >> internal calls and if fail then to cancel it and call via FS . >> >> On Jan 15, 2015, at 18:32, Steven Ayre > > wrote: >> >> I would expect that while it might work on some operators you won't be >> able to rely on it working for all mobile networks. >> >> >> >> If your users aren't under your control and therefore aren't on networks >> you've had an opportunity to test this on then you'll probably see many >> calls with no audio. >> >> >> >> >> >> >> >> On 15 January 2015 at 13:33, Assaf Dahary > > wrote: >> >> Hi, >> >> I'm currently using bypass_media to make two mobile call each directly >> (RTP bypass FS) over the same wifi network. >> >> Once I detect that caller and callee are sharing the same WiFi network >> then I switch to bypass_media in dialplan (works well in big complex >> buildings). >> >> I would like also to use the same bypass_media mode when mobiles connect >> over 3g/4g data mobile network. >> >> I fully understand that both mobiles should be connected to the same >> mobile operator and should share a common subnet routing (without NAT) for >> them to 'see' each other private IP addr. >> >> Is it possible for two mobiles on the same mobile 3g/4g network to ping >> each other using their private IP addr? >> >> Should mobile operators allow this kind of direct peer connections (I >> couldn't ping internally with my Orange mobile operator)? >> >> Did anyone successfully make a direct bypass_media call between two sip >> mobiles over 3g/4g network? >> >> Regards >> >> assaf >> >> >> >> ------------------------------ >> >> 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 >> >> -- Sent from mobile device -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/a77d0e0d/attachment.html From brian at freeswitch.org Sun Jan 18 02:49:35 2015 From: brian at freeswitch.org (Brian West) Date: Sat, 17 Jan 2015 17:49:35 -0600 Subject: [Freeswitch-users] mobile to mobile over 3g/4g In-Reply-To: References: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> <779edd43-435f-4d7a-afdf-ced2118a6277@email.bluemailapp.com> <005e01d03261$c177ba10$44672e30$@com> Message-ID: FYI proxy media mode is not needed and not what you think it is, FS always proxies the media, on AT&T you just need stun and it just works for me. On Saturday, January 17, 2015, Kristian Kielhofner wrote: > You want ICE. > > On Saturday, January 17, 2015, Assaf Dahary > wrote: > >> I desided to check each mobile operator network by trying first bypass >> mode and if it fails then redial automaticalky with proxy mode (and not >> trying it again with this operator). >> >> For those mobile network (fail bypass) users the alternatives can be: >> - buying static public ip from operator. >> - setting up mobile operator VPN group with public access to FS. >> - setting up common local VPN server to share direct connectivity for >> users group. >> On Jan 17, 2015, at 16:30, Bote Man wrote: >>> >>> I have it on good authority that AT&T in the U.S. assigns public i.p. >>> addresses to each device, but is subject to firewall rules whatever they >>> might be. >>> >>> >>> >>> They offer "enterprise static i.p. service" that allows greater freedom, >>> however. Sounds expensive and scary. >>> >>> >>> >>> I agree with Steven Ayre: I would not depend on having necessary >>> connectivity. >>> >>> >>> >>> Bote >>> >>> >>> >>> >>> >>> *From:* Assaf Dahary >>> *Sent:* Thursday, 15 January, 2015 11:47 >>> *Subject:* Re: [Freeswitch-users] mobile to mobile over 3g/4g >>> >>> It seems that some mobile operators allow internal direct connectivity >>> and some just block it. >>> >>> Direct connectivity will increase call quality and drop call latency. >>> >>> I should start mapping mobile operators by trying first bypass media on >>> internal calls and if fail then to cancel it and call via FS . >>> >>> On Jan 15, 2015, at 18:32, Steven Ayre wrote: >>> >>> I would expect that while it might work on some operators you won't be >>> able to rely on it working for all mobile networks. >>> >>> >>> >>> If your users aren't under your control and therefore aren't on networks >>> you've had an opportunity to test this on then you'll probably see many >>> calls with no audio. >>> >>> >>> >>> >>> >>> >>> >>> On 15 January 2015 at 13:33, Assaf Dahary wrote: >>> >>> Hi, >>> >>> I'm currently using bypass_media to make two mobile call each directly >>> (RTP bypass FS) over the same wifi network. >>> >>> Once I detect that caller and callee are sharing the same WiFi network >>> then I switch to bypass_media in dialplan (works well in big complex >>> buildings). >>> >>> I would like also to use the same bypass_media mode when mobiles connect >>> over 3g/4g data mobile network. >>> >>> I fully understand that both mobiles should be connected to the same >>> mobile operator and should share a common subnet routing (without NAT) for >>> them to 'see' each other private IP addr. >>> >>> Is it possible for two mobiles on the same mobile 3g/4g network to ping >>> each other using their private IP addr? >>> >>> Should mobile operators allow this kind of direct peer connections (I >>> couldn't ping internally with my Orange mobile operator)? >>> >>> Did anyone successfully make a direct bypass_media call between two sip >>> mobiles over 3g/4g network? >>> >>> Regards >>> >>> assaf >>> >>> >>> >>> ------------------------------ >>> >>> 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 >>> >>> > > -- > Sent from mobile device > -- *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/20150117/f6988885/attachment-0001.html From olenchenko at gmail.com Sat Jan 17 12:09:28 2015 From: olenchenko at gmail.com (Vasyl Olenchenko) Date: Sat, 17 Jan 2015 11:09:28 +0200 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? In-Reply-To: References: Message-ID: Have you tried CDRStats? 17 ???. 2015 10:41, ?????????? "Sina Owolabi" ???????: > Thanks Luis, Dan, Vik, everyone! > > I'm cloning the master. I'm more interested in CDR than billing actually, > though I now think one leads to the other. > > > On Fri Jan 16 2015 at 5:03:33 PM Moishe Grunstein > wrote: > >> FusionPBX does cdr?s however not billing, if you are looking for a >> billing solution see https://wiki.freeswitch.org/wiki/Billing >> >> >> >> Anything should work on Cenos, however Debian is currently the >> recommended Freeswitch platform. >> >> >> >> Thanks, >> >> >> >> Moishe Grunstein >> >> Tornado Computer Systems, Inc. >> >> 212.400.7650 888.IPPBX.US >> *Service Request Email: **support at nysolutions.com* >> >> >> [image: cid:image001.jpg at 01C72F94.9EE45D60] >> >> Computer Networking * Managed Services * IP Video Surveillance * Network >> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >> Security * Site Surveys * CMS >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sina >> Owolabi >> *Sent:* Friday, January 16, 2015 9:47 AM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Any CDR tool for Freeswitch that won't >> remove/overwrite existing config? >> >> >> >> Hi List! >> >> I'm checking out GUI billing and CDR records applications for FreeSWITCH. >> Most I've tried (bluebox, fusion, vbilling ) seem to either want to delete >> my existing configuration , or install their own Freeswitch , don't like >> CentOS, and totally ignore my own which I'm quite partial to. >> >> I wonder if there are any suggestions of CDR applications that will just >> fit in to what's already there? I'm running the latest master on a CentOS >> 64bit server. >> >> Thanks in advance ! >> ____________________________________________________________ >> _____________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/713cb3c6/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/713cb3c6/attachment-0001.jpg From olenchenko at gmail.com Sat Jan 17 12:09:29 2015 From: olenchenko at gmail.com (Vasyl Olenchenko) Date: Sat, 17 Jan 2015 11:09:29 +0200 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? In-Reply-To: References: Message-ID: Have you tried CDRStats? 17 ???. 2015 10:41, ?????????? "Sina Owolabi" ???????: > Thanks Luis, Dan, Vik, everyone! > > I'm cloning the master. I'm more interested in CDR than billing actually, > though I now think one leads to the other. > > > On Fri Jan 16 2015 at 5:03:33 PM Moishe Grunstein > wrote: > >> FusionPBX does cdr?s however not billing, if you are looking for a >> billing solution see https://wiki.freeswitch.org/wiki/Billing >> >> >> >> Anything should work on Cenos, however Debian is currently the >> recommended Freeswitch platform. >> >> >> >> Thanks, >> >> >> >> Moishe Grunstein >> >> Tornado Computer Systems, Inc. >> >> 212.400.7650 888.IPPBX.US >> *Service Request Email: **support at nysolutions.com* >> >> >> [image: cid:image001.jpg at 01C72F94.9EE45D60] >> >> Computer Networking * Managed Services * IP Video Surveillance * Network >> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >> Security * Site Surveys * CMS >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sina >> Owolabi >> *Sent:* Friday, January 16, 2015 9:47 AM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Any CDR tool for Freeswitch that won't >> remove/overwrite existing config? >> >> >> >> Hi List! >> >> I'm checking out GUI billing and CDR records applications for FreeSWITCH. >> Most I've tried (bluebox, fusion, vbilling ) seem to either want to delete >> my existing configuration , or install their own Freeswitch , don't like >> CentOS, and totally ignore my own which I'm quite partial to. >> >> I wonder if there are any suggestions of CDR applications that will just >> fit in to what's already there? I'm running the latest master on a CentOS >> 64bit server. >> >> Thanks in advance ! >> ____________________________________________________________ >> _____________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/651427ee/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150117/651427ee/attachment-0001.jpg From notify.sina at gmail.com Sun Jan 18 05:03:12 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Sun, 18 Jan 2015 02:03:12 +0000 Subject: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config? References: Message-ID: CDR-stats? I tried it on CentOS 6.6 and installation borked, couldn't login after it said it was done because it was asking me to input passwords it didn't request I create during the install. No way to uninstall it either, you have to reinstall the entire box. I posted a few questions on their forum and was largely ignored. On Sun, 18 Jan 2015 at 01:18, Vasyl Olenchenko wrote: > Have you tried CDRStats? > 17 ???. 2015 10:41, ?????????? "Sina Owolabi" > ???????: > >> Thanks Luis, Dan, Vik, everyone! > > >> I'm cloning the master. I'm more interested in CDR than billing actually, >> though I now think one leads to the other. >> >> >> On Fri Jan 16 2015 at 5:03:33 PM Moishe Grunstein >> wrote: >> >>> FusionPBX does cdr?s however not billing, if you are looking for a >>> billing solution see https://wiki.freeswitch.org/wiki/Billing >>> >>> >>> >>> Anything should work on Cenos, however Debian is currently the >>> recommended Freeswitch platform. >>> >>> >>> >>> Thanks, >>> >>> >>> >>> Moishe Grunstein >>> >>> Tornado Computer Systems, Inc. >>> >>> 212.400.7650 888.IPPBX.US >>> *Service Request Email: **support at nysolutions.com* >>> >>> >>> [image: cid:image001.jpg at 01C72F94.9EE45D60] >>> >>> >>> Computer Networking * Managed Services * IP Video Surveillance * Network >>> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >>> Security * Site Surveys * CMS >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sina >>> Owolabi >>> *Sent:* Friday, January 16, 2015 9:47 AM >>> *To:* FreeSWITCH Users Help >>> *Subject:* [Freeswitch-users] Any CDR tool for Freeswitch that won't >>> remove/overwrite existing config? >>> >>> >>> >>> Hi List! >>> >>> I'm checking out GUI billing and CDR records applications for >>> FreeSWITCH. Most I've tried (bluebox, fusion, vbilling ) seem to either >>> want to delete my existing configuration , or install their own Freeswitch >>> , don't like CentOS, and totally ignore my own which I'm quite partial to. >>> >>> I wonder if there are any suggestions of CDR applications that will just >>> fit in to what's already there? I'm running the latest master on a CentOS >>> 64bit server. >>> >>> Thanks in advance ! >>> ____________________________________________________________ >>> _____________ >>> 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 >> > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/20a254db/attachment-0001.jpg From sertys at gmail.com Sun Jan 18 11:43:26 2015 From: sertys at gmail.com (Daniel Ivanov) Date: Sun, 18 Jan 2015 10:43:26 +0200 Subject: [Freeswitch-users] mobile to mobile over 3g/4g In-Reply-To: References: <0ab001d030c7$d2c0cbe0$784263a0$@gmail.com> <779edd43-435f-4d7a-afdf-ced2118a6277@email.bluemailapp.com> <005e01d03261$c177ba10$44672e30$@com> Message-ID: I have been following this thread fo some time. Trying to shave off some delay with direct connectivity has cone to my mind a couple of times too. The trouble is that mobile devices are not your typical ethernet nodes. What you see as devices in the same subnet are often several complex hops away in terms of cell roaming connectivity. In relative terms, imagine that the ipv4 address space is already a VPN and the rules across segments are uniquely diverse. If you find it working on some mobile networks, you can still use ICE as suggested beforehand to present media candidates and not needing to retry the call. If device to device fails, it switches to proxy. 18 ???. 2015 ?. 1:50 ???????????? "Brian West" ???????: > FYI proxy media mode is not needed and not what you think it is, FS > always proxies the media, on AT&T you just need stun and it just works for > me. > > On Saturday, January 17, 2015, Kristian Kielhofner > wrote: > >> You want ICE. >> >> On Saturday, January 17, 2015, Assaf Dahary wrote: >> >>> I desided to check each mobile operator network by trying first bypass >>> mode and if it fails then redial automaticalky with proxy mode (and not >>> trying it again with this operator). >>> >>> For those mobile network (fail bypass) users the alternatives can be: >>> - buying static public ip from operator. >>> - setting up mobile operator VPN group with public access to FS. >>> - setting up common local VPN server to share direct connectivity for >>> users group. >>> On Jan 17, 2015, at 16:30, Bote Man wrote: >>>> >>>> I have it on good authority that AT&T in the U.S. assigns public i.p. >>>> addresses to each device, but is subject to firewall rules whatever they >>>> might be. >>>> >>>> >>>> >>>> They offer "enterprise static i.p. service" that allows greater >>>> freedom, however. Sounds expensive and scary. >>>> >>>> >>>> >>>> I agree with Steven Ayre: I would not depend on having necessary >>>> connectivity. >>>> >>>> >>>> >>>> Bote >>>> >>>> >>>> >>>> >>>> >>>> *From:* Assaf Dahary >>>> *Sent:* Thursday, 15 January, 2015 11:47 >>>> *Subject:* Re: [Freeswitch-users] mobile to mobile over 3g/4g >>>> >>>> It seems that some mobile operators allow internal direct connectivity >>>> and some just block it. >>>> >>>> Direct connectivity will increase call quality and drop call latency. >>>> >>>> I should start mapping mobile operators by trying first bypass media on >>>> internal calls and if fail then to cancel it and call via FS . >>>> >>>> On Jan 15, 2015, at 18:32, Steven Ayre wrote: >>>> >>>> I would expect that while it might work on some operators you won't be >>>> able to rely on it working for all mobile networks. >>>> >>>> >>>> >>>> If your users aren't under your control and therefore aren't on >>>> networks you've had an opportunity to test this on then you'll probably see >>>> many calls with no audio. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On 15 January 2015 at 13:33, Assaf Dahary wrote: >>>> >>>> Hi, >>>> >>>> I'm currently using bypass_media to make two mobile call each directly >>>> (RTP bypass FS) over the same wifi network. >>>> >>>> Once I detect that caller and callee are sharing the same WiFi network >>>> then I switch to bypass_media in dialplan (works well in big complex >>>> buildings). >>>> >>>> I would like also to use the same bypass_media mode when mobiles >>>> connect over 3g/4g data mobile network. >>>> >>>> I fully understand that both mobiles should be connected to the same >>>> mobile operator and should share a common subnet routing (without NAT) for >>>> them to 'see' each other private IP addr. >>>> >>>> Is it possible for two mobiles on the same mobile 3g/4g network to ping >>>> each other using their private IP addr? >>>> >>>> Should mobile operators allow this kind of direct peer connections (I >>>> couldn't ping internally with my Orange mobile operator)? >>>> >>>> Did anyone successfully make a direct bypass_media call between two sip >>>> mobiles over 3g/4g network? >>>> >>>> Regards >>>> >>>> assaf >>>> >>>> >>>> >>>> ------------------------------ >>>> >>>> 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 >>>> >>>> >> >> -- >> Sent from mobile device >> > > > -- > > *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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/f302b859/attachment-0001.html From mvar78 at gmail.com Sun Jan 18 13:59:00 2015 From: mvar78 at gmail.com (Jack Cortez) Date: Sun, 18 Jan 2015 11:59:00 +0100 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table Message-ID: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> Hi, I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. Is this possible? In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. I'm looking for days into the wiki but I didn't found anything interesting. Thank you so much! From luis.daniel.lucio at gmail.com Sun Jan 18 16:24:05 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sun, 18 Jan 2015 08:24:05 -0500 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> Message-ID: You may want to install FusionPBX, cdr code does exactly what you need 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 2015-01-18 5:59 GMT-05:00 Jack Cortez : > Hi, > I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. > Is this possible? > > In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. > > And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. > I'm looking for days into the wiki but I didn't found anything interesting. > > Thank you so much! > > > > > _________________________________________________________________________ > 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 From mvar78 at gmail.com Sun Jan 18 17:08:02 2015 From: mvar78 at gmail.com (Jack Cortez) Date: Sun, 18 Jan 2015 15:08:02 +0100 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> Message-ID: <1ADB93E3-160E-4901-869C-D1793EEA5173@gmail.com> Hi Luis! If this was a newest installation I will choose FusionPBX but as this is a running box I do prefer to not install Fusion, but just find a solution for adding a columns to channels table based on a custom variables (maybe declared into the dialplan or with a lua script). Can you give me a clue on this? Thanks Il giorno 18/gen/2015, alle ore 14:24, Luis Daniel Lucio Quiroz ha scritto: > You may want to install FusionPBX, cdr code does exactly what you need > 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 > > > 2015-01-18 5:59 GMT-05:00 Jack Cortez : >> Hi, >> I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. >> Is this possible? >> >> In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. >> >> And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. >> I'm looking for days into the wiki but I didn't found anything interesting. >> >> Thank you so much! >> >> >> >> >> _________________________________________________________________________ >> 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 From lloyd.aloysius at gmail.com Sun Jan 18 19:37:06 2015 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Sun, 18 Jan 2015 11:37:06 -0500 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: <1ADB93E3-160E-4901-869C-D1793EEA5173@gmail.com> References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> <1ADB93E3-160E-4901-869C-D1793EEA5173@gmail.com> Message-ID: Jack There is a channel variable for this purpose *sip_gateway_name.* LLoyd On Sun, Jan 18, 2015 at 9:08 AM, Jack Cortez wrote: > Hi Luis! > If this was a newest installation I will choose FusionPBX but as this is a > running box I do prefer to not install Fusion, but just find a solution for > adding a columns to channels table based on a custom variables (maybe > declared into the dialplan or with a lua script). > Can you give me a clue on this? > > Thanks > > > Il giorno 18/gen/2015, alle ore 14:24, Luis Daniel Lucio Quiroz ha scritto: > > > You may want to install FusionPBX, cdr code does exactly what you need > > 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 > > > > > > 2015-01-18 5:59 GMT-05:00 Jack Cortez : > >> Hi, > >> I would like to add a column to channel table (created by FS > automatically using ODBC) and populate this new columns with my own custom > value. > >> Is this possible? > >> > >> In particular, as I'm creating a simple PHP page to show active calls, > what I would like to add is a column with the gateway_name to identify the > Outbound Provider Name. So, in my PHP page I can show where the call is > going out. > >> > >> And also, I found that the application_data columns is showing the > sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this > columns is overwritten by last action I can't rely on this field. > >> I'm looking for days into the wiki but I didn't found anything > interesting. > >> > >> Thank you so much! > >> > >> > >> > >> > >> > _________________________________________________________________________ > >> 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 > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/a531a250/attachment.html From krice at freeswitch.org Sun Jan 18 20:16:47 2015 From: krice at freeswitch.org (Ken Rice) Date: Sun, 18 Jan 2015 11:16:47 -0600 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> Message-ID: <386E6570-1212-496F-BD13-072B8EACAE00@freeswitch.org> Ok luis while i understand that you are a fan of fusionpbx can we skip that as you suggested answer for everything on this list? Sent from my iPhone > On Jan 18, 2015, at 7:24 AM, Luis Daniel Lucio Quiroz wrote: > > You may want to install FusionPBX, cdr code does exactly what you need > 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 > > > 2015-01-18 5:59 GMT-05:00 Jack Cortez : >> Hi, >> I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. >> Is this possible? >> >> In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. >> >> And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. >> I'm looking for days into the wiki but I didn't found anything interesting. >> >> Thank you so much! >> >> >> >> >> _________________________________________________________________________ >> 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 From krice at freeswitch.org Sun Jan 18 20:18:44 2015 From: krice at freeswitch.org (Ken Rice) Date: Sun, 18 Jan 2015 11:18:44 -0600 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> Message-ID: You could possibly do that and not cause problems, but better would be create another table that has your custome datat in it and you can key with the channel's uuid Sent from my iPhone > On Jan 18, 2015, at 4:59 AM, Jack Cortez wrote: > > Hi, > I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. > Is this possible? > > In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. > > And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. > I'm looking for days into the wiki but I didn't found anything interesting. > > Thank you so much! > > > > > _________________________________________________________________________ > 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 From mvar78 at gmail.com Sun Jan 18 22:28:09 2015 From: mvar78 at gmail.com (Jack Cortez) Date: Sun, 18 Jan 2015 20:28:09 +0100 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> Message-ID: <41AC342E-EB3E-4B21-93C2-A568243F29A2@gmail.com> Hi Ken! I'm making some tests and it could be a good idea to create my personal channel table. However, in my tests I'm using a lua script to insert a new record in my table when call start with uuid as key, and delete the record at the end of the call, but I was thinking that is should be better to use the native channel table (that is better managed by FS itself) as I have to use a lua script for this task. Instead I can even use some custom variables that could be set into the XML dialplan, right? So, do you believe it's possible to overwrite for example context or dialplan columns with my custom values? I am suggesting context and dialplan as I found that on channels table they contain same value dialplan=XML and context=router and in my opinion they can be replaced with some more interested value Otherwise the only way it's really use a script and insert/update/delete calls into my new custom channels table. Thank you! Il giorno 18/gen/2015, alle ore 18:18, Ken Rice ha scritto: > You could possibly do that and not cause problems, but better would be create another table that has your custome datat in it and you can key with the channel's uuid > > Sent from my iPhone > >> On Jan 18, 2015, at 4:59 AM, Jack Cortez wrote: >> >> Hi, >> I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. >> Is this possible? >> >> In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. >> >> And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. >> I'm looking for days into the wiki but I didn't found anything interesting. >> >> Thank you so much! >> >> >> >> >> _________________________________________________________________________ >> 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 From zoell at zoell.us Sun Jan 18 22:42:27 2015 From: zoell at zoell.us (=?UTF-8?B?Wm9sdMOhbiBTemFiw7M=?=) Date: Sun, 18 Jan 2015 19:42:27 +0000 Subject: [Freeswitch-users] Freeswitch on VPS Message-ID: Hi, Does anyone have experience running Freeswitch at OVH? I have a VPS Classic 1 but the audio was really bad there. It was faltered. (tested with multiple clients from multiple locations) This is an OpenVZ virtualization if this counts. I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was also a little bit more, and it was a vServer virtualization, it worked perfectly. If you run your FS on a VPS without any problem, can you please recommend me some providers who has servers in the EU? I don't expect high call volume, just like 5-10 concurrent calls on 1-2 SIP trunks which is basically nothing. Many thanks, Zoltan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/c55de120/attachment.html From krice at freeswitch.org Sun Jan 18 22:43:06 2015 From: krice at freeswitch.org (Ken Rice) Date: Sun, 18 Jan 2015 13:43:06 -0600 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: <41AC342E-EB3E-4B21-93C2-A568243F29A2@gmail.com> References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> <41AC342E-EB3E-4B21-93C2-A568243F29A2@gmail.com> Message-ID: <703825C6-4B49-46A9-9CC0-A1A2C95F2AF6@freeswitch.org> Do not overwrite anything in there, its likely there for a reason... But adding a second table with the uuid is fine also keep in mind that if you use the channels table at master source you can clean our everything left over in the custom table every 5 minutes to reduce load on the db Sent from my iPhone > On Jan 18, 2015, at 1:28 PM, Jack Cortez wrote: > > Hi Ken! > I'm making some tests and it could be a good idea to create my personal channel table. > > However, in my tests I'm using a lua script to insert a new record in my table when call start with uuid as key, and delete the record at the end of the call, but I was thinking that is should be better to use the native channel table (that is better managed by FS itself) as I have to use a lua script for this task. > Instead I can even use some custom variables that could be set into the XML dialplan, right? > > So, do you believe it's possible to overwrite for example context or dialplan columns with my custom values? > I am suggesting context and dialplan as I found that on channels table they contain same value dialplan=XML and context=router and in my opinion they can be replaced with some more interested value > > > Otherwise the only way it's really use a script and insert/update/delete calls into my new custom channels table. > > Thank you! > > > > > >> Il giorno 18/gen/2015, alle ore 18:18, Ken Rice ha scritto: >> >> You could possibly do that and not cause problems, but better would be create another table that has your custome datat in it and you can key with the channel's uuid >> >> Sent from my iPhone >> >>> On Jan 18, 2015, at 4:59 AM, Jack Cortez wrote: >>> >>> Hi, >>> I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. >>> Is this possible? >>> >>> In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. >>> >>> And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. >>> I'm looking for days into the wiki but I didn't found anything interesting. >>> >>> Thank you so much! >>> >>> >>> >>> >>> _________________________________________________________________________ >>> 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 > > > _________________________________________________________________________ > 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 From max at nysolutions.com Sun Jan 18 22:49:44 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Sun, 18 Jan 2015 19:49:44 +0000 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: References: Message-ID: I know some users had luck with https://www.digitalocean.com for low volume. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Zolt?n Szab? Sent: Sunday, January 18, 2015 2:42 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch on VPS Hi, Does anyone have experience running Freeswitch at OVH? I have a VPS Classic 1 but the audio was really bad there. It was faltered. (tested with multiple clients from multiple locations) This is an OpenVZ virtualization if this counts. I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was also a little bit more, and it was a vServer virtualization, it worked perfectly. If you run your FS on a VPS without any problem, can you please recommend me some providers who has servers in the EU? I don't expect high call volume, just like 5-10 concurrent calls on 1-2 SIP trunks which is basically nothing. Many thanks, Zoltan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/341c10bb/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/341c10bb/attachment-0001.jpg From darren at aleph-com.net Sun Jan 18 22:54:11 2015 From: darren at aleph-com.net (Darren Wiebe) Date: Sun, 18 Jan 2015 12:54:11 -0700 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: References: Message-ID: I've had success with linode.com for low volume. Darren Wiebe D. 587-789-0634 T. 877-702-2900 C. 780-808-3320 E. darren at aleph-com.net w. www.aleph-com.net On Sun, Jan 18, 2015 at 12:49 PM, Moishe Grunstein wrote: > I know some users had luck with https://www.digitalocean.com for low > volume. > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Zolt?n Szab? > *Sent:* Sunday, January 18, 2015 2:42 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch on VPS > > > > Hi, > > > > Does anyone have experience running Freeswitch at OVH? I have a VPS > Classic 1 but the audio was really bad there. It was faltered. (tested with > multiple clients from multiple locations) This is an OpenVZ virtualization > if this counts. > > > > I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was also > a little bit more, and it was a vServer virtualization, it worked perfectly. > > > > If you run your FS on a VPS without any problem, can you please recommend > me some providers who has servers in the EU? I don't expect high call > volume, just like 5-10 concurrent calls on 1-2 SIP trunks which is > basically nothing. > > > > Many thanks, > > Zoltan > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/3a0485e1/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/3a0485e1/attachment.jpg From carlos.ruizdiaz at gmail.com Sun Jan 18 22:55:27 2015 From: carlos.ruizdiaz at gmail.com (=?UTF-8?Q?Carlos_Ruiz_D=C3=ADaz?=) Date: Sun, 18 Jan 2015 13:55:27 -0600 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: References: Message-ID: I can confirm that Digital Ocean works prefect with FS since they guarantee CPU and RAM per every instance running. I understand this isn't a reality with Amazon. What kind of instances are you running with EC2 to perfectly run FS? Regards, Carlos On Jan 18, 2015 1:50 PM, "Moishe Grunstein" wrote: > I know some users had luck with https://www.digitalocean.com for low > volume. > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Zolt?n Szab? > *Sent:* Sunday, January 18, 2015 2:42 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch on VPS > > > > Hi, > > > > Does anyone have experience running Freeswitch at OVH? I have a VPS > Classic 1 but the audio was really bad there. It was faltered. (tested with > multiple clients from multiple locations) This is an OpenVZ virtualization > if this counts. > > > > I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was also > a little bit more, and it was a vServer virtualization, it worked perfectly. > > > > If you run your FS on a VPS without any problem, can you please recommend > me some providers who has servers in the EU? I don't expect high call > volume, just like 5-10 concurrent calls on 1-2 SIP trunks which is > basically nothing. > > > > Many thanks, > > Zoltan > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/75398a16/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/75398a16/attachment-0001.jpg From david.villasmil at gmail.com Sun Jan 18 23:25:11 2015 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Sun, 18 Jan 2015 21:25:11 +0100 Subject: [Freeswitch-users] thousands of zombie calls In-Reply-To: References: Message-ID: Thanks for answering Italo, But that was no it. I've had it like that always.. ... On Sat, Jan 17, 2015 at 1:57 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Thanks Italo, I'll try that and let you know. > On Jan 17, 2015 3:21 AM, "?talo Rossi" wrote: > >> Set abandoned-resume-allowed to false >> Em 16/01/2015 18:42, "David Villasmil Govea" >> escreveu: >> >>> What's even weirder, if I try to uuid_kill one of those channels, fs >>> responds "no such channel".. so there's really no call there! >>> >>> Anyone? >>> >>> On Fri, Jan 16, 2015 at 10:26 PM, David Villasmil Govea < >>> david.villasmil at gmail.com> wrote: >>> >>>> Hello guys, >>>> >>>> I have a FS which when i execute on the cli: >>>> >>>> callcenter_config queue list members agents_queue >>>> >>>> I got THOUSANDS of "Abandoned" calls like this: >>>> >>>> agents_queue|single_box|e2685c96-6f85-4024-bd9a-5b2ac5b413d1||123456|Outbound >>>> Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned >>>> agents_queue|single_box|7bc18797-2913-476b-90ed-3f6905b99eda||123456|Outbound >>>> Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned >>>> agents_queue|single_box|96826416-0a90-4012-a68f-083534247424||123456|Outbound >>>> Call|0|1421429675|0|0|1421429676|0|0|||Abandoned >>>> >>>> This bot is used to generate calls to a test server via originate. But >>>> at this time the application which send the calls out is not even running! >>>> >>>> Furthermore, I stopped and started FS and the calls come back! WTF?? >>>> Where could all these calls come from?? >>>> >>>> Thanks for your help >>>> >>>> David >>>> >>>> >>>> -- >>>> DVG >>>> >>>> -- >>>> Imagination is more important than knowledge >>>> Albert Einstein >>>> >>> >>> >>> >>> -- >>> DVG >>> >>> -- >>> Imagination is more important than knowledge >>> Albert Einstein >>> >>> _________________________________________________________________________ >>> 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 >> > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/d8ea3f51/attachment.html From alhakeem at ipextelecom.net Sun Jan 18 23:32:02 2015 From: alhakeem at ipextelecom.net (Abdul Hakeem) Date: Sun, 18 Jan 2015 20:32:02 -0000 Subject: [Freeswitch-users] Fork( ) and Exec ( ) functions Message-ID: Hello again, The reason I ask is because I am trying to port FS to OSv. Unfortunately, Fork ( ) is unsupported in OSv. Regards, Abdul Hakeem From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva Sent: Tuesday, January 13, 2015 5:11 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fork( ) and Exec ( ) functions On Tue, Jan 13, 2015 at 8:11 AM, Abdul Hakeem wrote: Hello, I understand FS makes system calls for sending mails, voicemail and fax. Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on system calls & also, a list of functions which require FS to make system calls ? Not making much sense here. Everything in FS relies heavily on system calls as it's a multi-threaded-I/O-driven system for the most part. I think you're asking the wrong question. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/ee618d96/attachment-0001.html From sos at sokhapkin.dyndns.org Sun Jan 18 23:48:07 2015 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 18 Jan 2015 15:48:07 -0500 Subject: [Freeswitch-users] Fork( ) and Exec ( ) functions In-Reply-To: References: Message-ID: <6053000.JvnxKzh3j9@sos> Usually fork-less OSes provide spawn() family of syscalls to execute a new process. On Sunday 18 January 2015 20:32:02 Abdul Hakeem wrote: > Hello again, > The reason I ask is because I am trying to port FS to OSv. > Unfortunately, Fork ( ) is unsupported in OSv. > Regards, > Abdul Hakeem > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises > Silva Sent: Tuesday, January 13, 2015 5:11 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Fork( ) and Exec ( ) functions > > On Tue, Jan 13, 2015 at 8:11 AM, Abdul Hakeem wrote: > Hello, > > I understand FS makes system calls for sending mails, voicemail and fax. > Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on > system calls & also, a list of functions which require FS to make system > calls ? > > Not making much sense here. Everything in FS relies heavily on system calls > as it's a multi-threaded-I/O-driven system for the most part. I think > you're asking the wrong question. From italorossib at gmail.com Mon Jan 19 03:39:42 2015 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Sun, 18 Jan 2015 21:39:42 -0300 Subject: [Freeswitch-users] thousands of zombie calls In-Reply-To: References: Message-ID: By default mod_callcenter removes these abandoned members from db with this condition: if (abandoned_epoch + discard_abandoned_after < local_epoch_time_now(NULL)) { discard_abandoned_after should be by default 60 seconds. You don't need to worry with this, if you don't want these entries in db set discard-abandoned-after to a low value, like 1sec. You can't uuid_kill, they don't exists anymore, this feature allow a caller to recover their original position when they call again the queue (if they call again before discard_abandon_after expires..). On Sun, Jan 18, 2015 at 5:25 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: > Thanks for answering Italo, But that was no it. I've had it like that > always.. > > > > ... > > > > > On Sat, Jan 17, 2015 at 1:57 PM, David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> Thanks Italo, I'll try that and let you know. >> On Jan 17, 2015 3:21 AM, "?talo Rossi" wrote: >> >>> Set abandoned-resume-allowed to false >>> Em 16/01/2015 18:42, "David Villasmil Govea" >>> escreveu: >>> >>>> What's even weirder, if I try to uuid_kill one of those channels, fs >>>> responds "no such channel".. so there's really no call there! >>>> >>>> Anyone? >>>> >>>> On Fri, Jan 16, 2015 at 10:26 PM, David Villasmil Govea < >>>> david.villasmil at gmail.com> wrote: >>>> >>>>> Hello guys, >>>>> >>>>> I have a FS which when i execute on the cli: >>>>> >>>>> callcenter_config queue list members agents_queue >>>>> >>>>> I got THOUSANDS of "Abandoned" calls like this: >>>>> >>>>> agents_queue|single_box|e2685c96-6f85-4024-bd9a-5b2ac5b413d1||123456|Outbound >>>>> Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned >>>>> agents_queue|single_box|7bc18797-2913-476b-90ed-3f6905b99eda||123456|Outbound >>>>> Call|1421429675|123456|0|0|1421432074|0|0|||Abandoned >>>>> agents_queue|single_box|96826416-0a90-4012-a68f-083534247424||123456|Outbound >>>>> Call|0|1421429675|0|0|1421429676|0|0|||Abandoned >>>>> >>>>> This bot is used to generate calls to a test server via originate. But >>>>> at this time the application which send the calls out is not even running! >>>>> >>>>> Furthermore, I stopped and started FS and the calls come back! WTF?? >>>>> Where could all these calls come from?? >>>>> >>>>> Thanks for your help >>>>> >>>>> David >>>>> >>>>> >>>>> -- >>>>> DVG >>>>> >>>>> -- >>>>> Imagination is more important than knowledge >>>>> Albert Einstein >>>>> >>>> >>>> >>>> >>>> -- >>>> DVG >>>> >>>> -- >>>> Imagination is more important than knowledge >>>> Albert Einstein >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> > > > -- > DVG > > -- > Imagination is more important than knowledge > Albert Einstein > > _________________________________________________________________________ > 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 > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150118/232d811d/attachment.html From babak.freeswitch at gmail.com Mon Jan 19 10:32:47 2015 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Mon, 19 Jan 2015 11:02:47 +0330 Subject: [Freeswitch-users] Startup script is not respecting real time priority In-Reply-To: References: Message-ID: adding the line below to "/etc/security/limits.conf" solves my problem: freeswitch - nice -20 On Sat, Jan 17, 2015 at 4:24 PM, Babak Yakhchali wrote: > Hi > I'm using startup script supplied in repo for centos 6.5 and setting the > -rp like this: > > PROG_NAME=freeswitch > PID_FILE=/usr/local/freeswitch/run/freeswitch.pid > FS_USER=freeswitch > FS_FILE=/usr/local/freeswitch/bin/freeswitch > FS_HOME=/usr/local/freeswitch/run > LOCK_FILE=/var/lock/subsys/freeswitch > *FREESWITCH_ARGS="-nc -rp"* > RETVAL=0 > > but freeswitch is always running with default priority as seen in top: > > 16263 freeswit 39 19 875m 30m 7416 S 1.0 13.0 0:17.44 freeswitch > > and in "ps aux|grep freeswitch" I see: > 500 16263 0.8 12.9 897004 31120 ? SNl 15:38 0:18 > /usr/local/freeswitch/bin/freeswitch -nc -rp > so it seems freeswitch is starting with proper args but priority is not > set. > If I run freeswitch as root in bash it is working will: > runing ./freeswitch -nc -rp results in: > 16550 root -2 -10 865m 28m 6988 S 6.3 12.2 0:00.52 freeswitch > Should I change anything else in the script? or add any special permission > to user "freeswitch"? > Thanks > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/9f290720/attachment.html From notify.sina at gmail.com Mon Jan 19 10:32:47 2015 From: notify.sina at gmail.com (Sina Owolabi) Date: Mon, 19 Jan 2015 07:32:47 +0000 Subject: [Freeswitch-users] FreeSWITCH based Contact Center Software Suggestions Message-ID: Hi List! I'm wondering if there are any freeSWITCH based contact center application suites out there that are easily deployed, and open source that you can recommend. I'm helping a friend figure out options and I'd like your advice. Thanks for the help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/978c1866/attachment.html From nagalenoj at gmail.com Mon Jan 19 12:38:02 2015 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Mon, 19 Jan 2015 15:08:02 +0530 Subject: [Freeswitch-users] spandsp_start_dtmf detects # key many times Message-ID: Hi, When I use spandsp_start_dtmf along with play_and_get_digits, '#' is getting detected many times, though it is pressed once. Other DTMFs are getting detected properly, whereas the '#' is getting detected 2 or 3 times. I have checked the SIP phone's settings, I do not find anything wrong there. What could be the reason? Is this because of some SIP phone setting? Here is the freeswitch log, 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [1] 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF 1:2000 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [1], duration = 191 ms 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [4] 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [4], duration = 153 ms 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [7] 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [7], duration = 204 ms 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [4] 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [4], duration = 127 ms 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:278 at 192.168.2.86:5060 [BREAK] 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [7] 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [7], duration = 191 ms 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 EXECUTE sofia/internal/392 at 3427.vbiz.mundio.com flush_dtmf() 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 30ms 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 38 ms 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done playing file file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 30ms 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 25 ms 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 76 ms -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/a360520d/attachment-0001.html From groysem at gmail.com Mon Jan 19 15:56:38 2015 From: groysem at gmail.com (Shai Perelman) Date: Mon, 19 Jan 2015 14:56:38 +0200 Subject: [Freeswitch-users] looking to pay for help in configuration Message-ID: Hi, Im new to freeswitch and the list, so hi everybody. I,m learning the subject of fusionpbx/freeswitch and have a server up and running, got a few problems. and need someone to assist me on a hourly rate. is there any one interested? thanks Shai -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/8fd4a558/attachment.html From avi at avimarcus.net Mon Jan 19 16:29:17 2015 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Jan 2015 13:29:17 +0000 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: References: Message-ID: <0000014b0262139c-6d06a296-5485-4878-80b7-cff4ab8c5817-000000@email.amazonses.com> I've been using Linode in Europe for years now, on low volume. It's possible that some audio glitches are due to the virtualization, but I've never positively identified anything. OVH has some low-end cheap dedicated servers in Europe via Kimsufi (France I think, but some are in Canada)- you might want to try that if you want to be sure - it might be worth the price. -Avi On Sun, Jan 18, 2015 at 9:55 PM, Carlos Ruiz D?az wrote: > I can confirm that Digital Ocean works prefect with FS since they > guarantee CPU and RAM per every instance running. I understand this isn't a > reality with Amazon. > > What kind of instances are you running with EC2 to perfectly run FS? > > Regards, > Carlos > On Jan 18, 2015 1:50 PM, "Moishe Grunstein" wrote: > >> I know some users had luck with https://www.digitalocean.com for low >> volume. >> >> >> >> Thanks, >> >> >> >> Moishe Grunstein >> >> Tornado Computer Systems, Inc. >> >> 212.400.7650 888.IPPBX.US >> *Service Request Email: support at nysolutions.com >> * >> >> [image: cid:image001.jpg at 01C72F94.9EE45D60] >> >> Computer Networking * Managed Services * IP Video Surveillance * Network >> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >> Security * Site Surveys * CMS >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Zolt?n >> Szab? >> *Sent:* Sunday, January 18, 2015 2:42 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Freeswitch on VPS >> >> >> >> Hi, >> >> >> >> Does anyone have experience running Freeswitch at OVH? I have a VPS >> Classic 1 but the audio was really bad there. It was faltered. (tested with >> multiple clients from multiple locations) This is an OpenVZ virtualization >> if this counts. >> >> >> >> I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was also >> a little bit more, and it was a vServer virtualization, it worked perfectly. >> >> >> >> If you run your FS on a VPS without any problem, can you please recommend >> me some providers who has servers in the EU? I don't expect high call >> volume, just like 5-10 concurrent calls on 1-2 SIP trunks which is >> basically nothing. >> >> >> >> Many thanks, >> >> Zoltan >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/a2c76819/attachment.html From max at nysolutions.com Mon Jan 19 16:58:53 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Mon, 19 Jan 2015 13:58:53 +0000 Subject: [Freeswitch-users] looking to pay for help in configuration In-Reply-To: References: Message-ID: There is a special mailing list for this type of request http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz Also Freeswitch Solutions now offers installation services, https://freeswitch.com/cart.php?gid=4 I am not sure if they will help with FusionPBX though. FusionPBX has their own IRC channel and support website. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shai Perelman Sent: Monday, January 19, 2015 7:57 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] looking to pay for help in configuration Hi, Im new to freeswitch and the list, so hi everybody. I,m learning the subject of fusionpbx/freeswitch and have a server up and running, got a few problems. and need someone to assist me on a hourly rate. is there any one interested? thanks Shai -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/e01b4647/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/e01b4647/attachment-0001.jpg From luis.daniel.lucio at gmail.com Mon Jan 19 17:01:20 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 19 Jan 2015 09:01:20 -0500 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: <0000014b0262139c-6d06a296-5485-4878-80b7-cff4ab8c5817-000000@email.amazonses.com> References: <0000014b0262139c-6d06a296-5485-4878-80b7-cff4ab8c5817-000000@email.amazonses.com> Message-ID: I have been using openvz + 729 with moderate load without a problem 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 2015-01-19 8:29 GMT-05:00 Avi Marcus : > I've been using Linode in Europe for years now, on low volume. It's > possible that some audio glitches are due to the virtualization, but I've > never positively identified anything. > > OVH has some low-end cheap dedicated servers in Europe via Kimsufi > (France I think, but some are in Canada)- > you might want to try that if you want to be sure - it might be worth the > price. > > -Avi > > On Sun, Jan 18, 2015 at 9:55 PM, Carlos Ruiz D?az < > carlos.ruizdiaz at gmail.com> wrote: > >> I can confirm that Digital Ocean works prefect with FS since they >> guarantee CPU and RAM per every instance running. I understand this isn't a >> reality with Amazon. >> >> What kind of instances are you running with EC2 to perfectly run FS? >> >> Regards, >> Carlos >> On Jan 18, 2015 1:50 PM, "Moishe Grunstein" wrote: >> >>> I know some users had luck with https://www.digitalocean.com for low >>> volume. >>> >>> >>> >>> Thanks, >>> >>> >>> >>> Moishe Grunstein >>> >>> Tornado Computer Systems, Inc. >>> >>> 212.400.7650 888.IPPBX.US >>> *Service Request Email: support at nysolutions.com >>> * >>> >>> [image: cid:image001.jpg at 01C72F94.9EE45D60] >>> >>> >>> Computer Networking * Managed Services * IP Video Surveillance * Network >>> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >>> Security * Site Surveys * CMS >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Zolt?n >>> Szab? >>> *Sent:* Sunday, January 18, 2015 2:42 PM >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Subject:* [Freeswitch-users] Freeswitch on VPS >>> >>> >>> >>> Hi, >>> >>> >>> >>> Does anyone have experience running Freeswitch at OVH? I have a VPS >>> Classic 1 but the audio was really bad there. It was faltered. (tested with >>> multiple clients from multiple locations) This is an OpenVZ virtualization >>> if this counts. >>> >>> >>> >>> I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was >>> also a little bit more, and it was a vServer virtualization, it worked >>> perfectly. >>> >>> >>> >>> If you run your FS on a VPS without any problem, can you please >>> recommend me some providers who has servers in the EU? I don't expect high >>> call volume, just like 5-10 concurrent calls on 1-2 SIP trunks which is >>> basically nothing. >>> >>> >>> >>> Many thanks, >>> >>> Zoltan >>> >>> _________________________________________________________________________ >>> 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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/432c91aa/attachment.html From luis.daniel.lucio at gmail.com Mon Jan 19 17:11:43 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 19 Jan 2015 09:11:43 -0500 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: <703825C6-4B49-46A9-9CC0-A1A2C95F2AF6@freeswitch.org> References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> <41AC342E-EB3E-4B21-93C2-A568243F29A2@gmail.com> <703825C6-4B49-46A9-9CC0-A1A2C95F2AF6@freeswitch.org> Message-ID: Ken, I dont suggest in all my answers to install FUsion. But when it is useful i do. Jack. Download the fusion and review the file named: v_xml_cdr_import.php you will find there is a code that use the variables lcr_* those variables are pushed with an export using dialplans. You will figure out how to do your own code. LD 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 2015-01-18 14:43 GMT-05:00 Ken Rice : > Do not overwrite anything in there, its likely there for a reason... But adding a second table with the uuid is fine also keep in mind that if you use the channels table at master source you can clean our everything left over in the custom table every 5 minutes to reduce load on the db > > Sent from my iPhone > >> On Jan 18, 2015, at 1:28 PM, Jack Cortez wrote: >> >> Hi Ken! >> I'm making some tests and it could be a good idea to create my personal channel table. >> >> However, in my tests I'm using a lua script to insert a new record in my table when call start with uuid as key, and delete the record at the end of the call, but I was thinking that is should be better to use the native channel table (that is better managed by FS itself) as I have to use a lua script for this task. >> Instead I can even use some custom variables that could be set into the XML dialplan, right? >> >> So, do you believe it's possible to overwrite for example context or dialplan columns with my custom values? >> I am suggesting context and dialplan as I found that on channels table they contain same value dialplan=XML and context=router and in my opinion they can be replaced with some more interested value >> >> >> Otherwise the only way it's really use a script and insert/update/delete calls into my new custom channels table. >> >> Thank you! >> >> >> >> >> >>> Il giorno 18/gen/2015, alle ore 18:18, Ken Rice ha scritto: >>> >>> You could possibly do that and not cause problems, but better would be create another table that has your custome datat in it and you can key with the channel's uuid >>> >>> Sent from my iPhone >>> >>>> On Jan 18, 2015, at 4:59 AM, Jack Cortez wrote: >>>> >>>> Hi, >>>> I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. >>>> Is this possible? >>>> >>>> In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. >>>> >>>> And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. >>>> I'm looking for days into the wiki but I didn't found anything interesting. >>>> >>>> Thank you so much! >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >> >> >> _________________________________________________________________________ >> 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 From magnus.kelly at gmail.com Mon Jan 19 17:24:16 2015 From: magnus.kelly at gmail.com (Magnus Kelly) Date: Mon, 19 Jan 2015 14:24:16 +0000 Subject: [Freeswitch-users] Mod-Sofia ability to dial Alphanumeric characters in invite? Message-ID: Hello Guys, Is it possible to use FS with alphanumeric characters in the SIP invite? As in I would like to send an invite addressed to +C27344xxxxxxxxxx as one of our upstream platforms uses a C prefix. What I observe with default is " [ERR] mod_sofia.c:4431 Invalid URL " If possible to support what would need to be changed? I am assuming that this is outside the SIP standards, but confirmation would equally be useful. Thanks Magnus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/fc884b03/attachment.html From kris at kriskinc.com Mon Jan 19 19:03:40 2015 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 19 Jan 2015 11:03:40 -0500 Subject: [Freeswitch-users] Mod-Sofia ability to dial Alphanumeric characters in invite? In-Reply-To: References: Message-ID: It's well within standard and actually required for many platforms. What does your bridge string look like? On Mon, Jan 19, 2015 at 9:24 AM, Magnus Kelly wrote: > Hello Guys, > > Is it possible to use FS with alphanumeric characters in the SIP invite? > > As in I would like to send an invite addressed to +C27344xxxxxxxxxx as one > of our upstream platforms uses a C prefix. > > What I observe with default is " [ERR] mod_sofia.c:4431 Invalid URL " > > If possible to support what would need to be changed? I am assuming that > this is outside the SIP standards, but confirmation would equally be useful. > > Thanks > Magnus > > _________________________________________________________________________ > 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 -- Kristian Kielhofner From mike at jerris.com Mon Jan 19 20:06:12 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jan 2015 12:06:12 -0500 Subject: [Freeswitch-users] Fork( ) and Exec ( ) functions In-Reply-To: <6053000.JvnxKzh3j9@sos> References: <6053000.JvnxKzh3j9@sos> Message-ID: <1C00AF43-07EF-4A86-8CB0-286C49617B5F@jerris.com> You can also look at how we have addressed this issue for windows. The places I know of that use these functions. switch_system (used for executing external commands and sending emails), and daemonizing FreeSWITCH. > On Jan 18, 2015, at 3:48 PM, Sergey Okhapkin wrote: > > Usually fork-less OSes provide spawn() family of syscalls to execute a new > process. > > On Sunday 18 January 2015 20:32:02 Abdul Hakeem wrote: >> Hello again, >> The reason I ask is because I am trying to port FS to OSv. >> Unfortunately, Fork ( ) is unsupported in OSv. >> Regards, >> Abdul Hakeem >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises >> Silva Sent: Tuesday, January 13, 2015 5:11 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Fork( ) and Exec ( ) functions >> >> On Tue, Jan 13, 2015 at 8:11 AM, Abdul Hakeem wrote: >> Hello, >> >> I understand FS makes system calls for sending mails, voicemail and fax. >> Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on >> system calls & also, a list of functions which require FS to make system >> calls ? >> >> Not making much sense here. Everything in FS relies heavily on system calls >> as it's a multi-threaded-I/O-driven system for the most part. I think >> you're asking the wrong question. From mike at jerris.com Mon Jan 19 20:10:55 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jan 2015 12:10:55 -0500 Subject: [Freeswitch-users] spandsp_start_dtmf detects # key many times In-Reply-To: References: Message-ID: <33221776-F8EB-4329-BC1A-5413478DBD16@jerris.com> If you are using a sip phone, you are MUCH better off using non audio based dtmf like rfc-2833. If you are already using this, its possible that what is happening is the phone is sending the dtmf as audio and 2833, at which point, you should not have to use spandsp_start_dtmf at all. If the phone is sending only audio, and is not capable of using any other method, you will need to pull a pcap of this call, use wireshark to extract the audio, and do some analysis of the audio to figure out why it is detecting multiple digits. > On Jan 19, 2015, at 4:38 AM, Nagalenoj H. wrote: > > Hi, > > When I use spandsp_start_dtmf along with play_and_get_digits, '#' is getting detected many times, though it is pressed once. > > Other DTMFs are getting detected properly, whereas the '#' is getting detected 2 or 3 times. > > I have checked the SIP phone's settings, I do not find anything wrong there. > > What could be the reason? Is this because of some SIP phone setting? Here is the freeswitch log, > > 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [1] > 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF 1:2000 > 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [1], duration = 191 ms > 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [4] > 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 > 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [4], duration = 153 ms > 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [7] > 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 > 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [7], duration = 204 ms > 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [4] > 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 > 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [4], duration = 127 ms > 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:278 at 192.168.2.86:5060 [BREAK] > 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [7] > 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 > 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [7], duration = 191 ms > 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms > 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] > 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > EXECUTE sofia/internal/392 at 3427.vbiz.mundio.com flush_dtmf() > 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 30ms > 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 38 ms > 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] > 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done playing file file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 30ms > 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 25 ms > 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] > 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 76 ms -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/68b393b5/attachment.html From anthony.minessale at gmail.com Mon Jan 19 20:12:19 2015 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 19 Jan 2015 11:12:19 -0600 Subject: [Freeswitch-users] Fork( ) and Exec ( ) functions In-Reply-To: <1C00AF43-07EF-4A86-8CB0-286C49617B5F@jerris.com> References: <6053000.JvnxKzh3j9@sos> <1C00AF43-07EF-4A86-8CB0-286C49617B5F@jerris.com> Message-ID: We use threads by default in the switch system. There is a setting to use fork instead but it's disabled by default. The system syscall uses fork and exec on its own so maybe you just need to make sure ifdefs properly disable any such functionality. On Monday, January 19, 2015, Michael Jerris wrote: > You can also look at how we have addressed this issue for windows. The > places I know of that use these functions. switch_system (used for > executing external commands and sending emails), and daemonizing FreeSWITCH. > > > > On Jan 18, 2015, at 3:48 PM, Sergey Okhapkin > wrote: > > > > Usually fork-less OSes provide spawn() family of syscalls to execute a > new > > process. > > > > On Sunday 18 January 2015 20:32:02 Abdul Hakeem wrote: > >> Hello again, > >> The reason I ask is because I am trying to port FS to OSv. > >> Unfortunately, Fork ( ) is unsupported in OSv. > >> Regards, > >> Abdul Hakeem > >> > >> From: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org ] > On Behalf Of Moises > >> Silva Sent: Tuesday, January 13, 2015 5:11 PM > >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] Fork( ) and Exec ( ) functions > >> > >> On Tue, Jan 13, 2015 at 8:11 AM, Abdul Hakeem > wrote: > >> Hello, > >> > >> I understand FS makes system calls for sending mails, voicemail and fax. > >> Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) > on > >> system calls & also, a list of functions which require FS to make > system > >> calls ? > >> > >> Not making much sense here. Everything in FS relies heavily on system > calls > >> as it's a multi-threaded-I/O-driven system for the most part. I think > >> you're asking the wrong question. > > _________________________________________________________________________ > 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 > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/8307c40b/attachment.html From mike at jerris.com Mon Jan 19 20:12:31 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jan 2015 12:12:31 -0500 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> <41AC342E-EB3E-4B21-93C2-A568243F29A2@gmail.com> <703825C6-4B49-46A9-9CC0-A1A2C95F2AF6@freeswitch.org> Message-ID: <0CE94105-05A9-48FE-A90A-E668094E4254@jerris.com> if you are just messing with 1 additional column, check out using presence_data for this. > On Jan 19, 2015, at 9:11 AM, Luis Daniel Lucio Quiroz wrote: > > Ken, I dont suggest in all my answers to install FUsion. But when it > is useful i do. > > Jack. Download the fusion and review the file named: > v_xml_cdr_import.php you will find there is a code that use the > variables lcr_* > > those variables are pushed with an export using dialplans. You will > figure out how to do your own code. > > LD > 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 > > > 2015-01-18 14:43 GMT-05:00 Ken Rice : >> Do not overwrite anything in there, its likely there for a reason... But adding a second table with the uuid is fine also keep in mind that if you use the channels table at master source you can clean our everything left over in the custom table every 5 minutes to reduce load on the db >> >> Sent from my iPhone >> >>> On Jan 18, 2015, at 1:28 PM, Jack Cortez wrote: >>> >>> Hi Ken! >>> I'm making some tests and it could be a good idea to create my personal channel table. >>> >>> However, in my tests I'm using a lua script to insert a new record in my table when call start with uuid as key, and delete the record at the end of the call, but I was thinking that is should be better to use the native channel table (that is better managed by FS itself) as I have to use a lua script for this task. >>> Instead I can even use some custom variables that could be set into the XML dialplan, right? >>> >>> So, do you believe it's possible to overwrite for example context or dialplan columns with my custom values? >>> I am suggesting context and dialplan as I found that on channels table they contain same value dialplan=XML and context=router and in my opinion they can be replaced with some more interested value >>> >>> >>> Otherwise the only way it's really use a script and insert/update/delete calls into my new custom channels table. >>> >>> Thank you! >>> >>> >>> >>> >>> >>>> Il giorno 18/gen/2015, alle ore 18:18, Ken Rice ha scritto: >>>> >>>> You could possibly do that and not cause problems, but better would be create another table that has your custome datat in it and you can key with the channel's uuid >>>> >>>> Sent from my iPhone >>>> >>>>> On Jan 18, 2015, at 4:59 AM, Jack Cortez wrote: >>>>> >>>>> Hi, >>>>> I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. >>>>> Is this possible? >>>>> >>>>> In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. >>>>> >>>>> And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. >>>>> I'm looking for days into the wiki but I didn't found anything interesting. >>>>> >>>>> Thank you so much! >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>> >>> >>> _________________________________________________________________________ >>> 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 > > _________________________________________________________________________ > 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 From godson.g at gmail.com Mon Jan 19 20:50:40 2015 From: godson.g at gmail.com (Godson Gera) Date: Mon, 19 Jan 2015 23:20:40 +0530 Subject: [Freeswitch-users] Sending call to outbound proxy In-Reply-To: References: Message-ID: set fs_path in originate or bridge string. Or use outbound proxy option in sofia profile. http://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path On Sat, Jan 17, 2015 at 9:57 PM, John Nash wrote: > I am trying to test a case where I am receiving call from opensips to > freeswitch and need to route back to opensips again without changing > Request URI. How can I do it? > > _________________________________________________________________________ > 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 > -- Thanks & Regards, Godson Gera FreeSWITCH Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/03e9fe55/attachment.html From nico at vthadden.de Mon Jan 19 22:50:54 2015 From: nico at vthadden.de (Nicola von Thadden) Date: Mon, 19 Jan 2015 20:50:54 +0100 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: References: Message-ID: <54BD601E.30603@vthadden.de> I have it running on a machine from netcup.net. They are from germany and use KVM. I have no audio issues but I usually only have one oder two concurrent calls. I did test a conference with about 8 phones and it worked too. Nico On 18.01.2015 20:42, Zolt?n Szab? wrote: > Hi, > > Does anyone have experience running Freeswitch at OVH? I have a VPS > Classic 1 but the audio was really bad there. It was faltered. (tested > with multiple clients from multiple locations) This is an OpenVZ > virtualization if this counts. > > I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was > also a little bit more, and it was a vServer virtualization, it worked > perfectly. > > If you run your FS on a VPS without any problem, can you please > recommend me some providers who has servers in the EU? I don't expect > high call volume, just like 5-10 concurrent calls on 1-2 SIP trunks > which is basically nothing. > > Many thanks, > Zoltan > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/3559543b/attachment.html From krice at freeswitch.org Mon Jan 19 23:09:52 2015 From: krice at freeswitch.org (Ken Rice) Date: Mon, 19 Jan 2015 20:09:52 +0000 Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) January 11th-16th Message-ID: <54bd6490f11d_74e410e133836795@ip-10-33-128-130.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1Gl1ooB FreeSWITCH Week in Review (Master Branch) January 11th-16th Hello, again. This week in the FreeSWITCH master branch we had 15 commits. In an effort to improve the execution of the Week in Review, starting this week I will be using a Saturday-Friday time-line. This will give me more time to review the commits and get answers to questions. That being said, the features for this week are: completing the Linux portion of the update to flite-2.0.0-release and added param add-variables-to-events which will add channel variables to <offer>, <ringing>, <answered>, and <end> if set to true, while the default is false, in mod_rayo. New features that were added: 38d25f1 FS-7149 Linux portion of update to use flite-2.0.0-release complete [Jira: http://ift.tt/1Gl1nRp] 85fbddc FS-7150 Added param add-variables-to-events which will add channel variables to <offer>, <ringing>, <answered>, and <end> if set to true, while the default is false, in mod_rayo [Jira: http://ift.tt/1Gl1nRr] In terms of stability these were the use cases that were fixed: 59a9669 Prevent crash when calling mediaStats JSON function in certain circumstances in mod_conference Improvements to packaging: adb0de9 Update debian utils for flite 2.0.0 #resolve 5c5b9fc FS-7151 If available, use libjpeg-turbo in Debian build dependencies The following bugs were squashed: 3e6ffbc FS-7144 Cleaning up NAT?d RTP where ICE is involved [Jira: http://ift.tt/1CI1nVI] 403d32c FS-7148 Fixed a deadlock [Jira: http://ift.tt/1Gl1ooD] 378d350 FS-7144 More cleaning up NAT?d RTP where ICE is involved [Jira: http://ift.tt/1pgbVcN] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/48d1c529/attachment.html From magnus.kelly at gmail.com Mon Jan 19 23:17:24 2015 From: magnus.kelly at gmail.com (Magnus Kelly) Date: Mon, 19 Jan 2015 20:17:24 +0000 Subject: [Freeswitch-users] Mod-Sofia ability to dial Alphanumeric characters in invite? In-Reply-To: References: Message-ID: On 19/01/2015 16:03, "Kristian Kielhofner" wrote: >It's well within standard and actually required for many platforms. > >What does your bridge string look like? > >On Mon, Jan 19, 2015 at 9:24 AM, Magnus Kelly >wrote: >> Hello Guys, >> >> Is it possible to use FS with alphanumeric characters in the SIP >>invite? >> >> As in I would like to send an invite addressed to +C27344xxxxxxxxxx as >>one >> of our upstream platforms uses a C prefix. >> >> What I observe with default is " [ERR] mod_sofia.c:4431 Invalid URL " >> >> If possible to support what would need to be changed? I am assuming that >> this is outside the SIP standards, but confirmation would equally be >>useful. >> >> Thanks >> Magnus >> >> >>_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: Hello Kristian, I also re-read bit further and figured out it was a silly question as how else would names be used, however I am using a simple dial plan - as in the below - And when the incoming invite that?s destined to dial out on other leg comes as ? +C27344xxxxxxxxxx " I get the error as in ? [ERR] mod_sofia.c:4431 Invalid URL ? I will continue to review but thoughts welcome. my guess is need to study perl expressions. Thanks Magnus From mvar78 at gmail.com Mon Jan 19 23:54:57 2015 From: mvar78 at gmail.com (Jack Cortez) Date: Mon, 19 Jan 2015 21:54:57 +0100 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: <0CE94105-05A9-48FE-A90A-E668094E4254@jerris.com> References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> <41AC342E-EB3E-4B21-93C2-A568243F29A2@gmail.com> <703825C6-4B49-46A9-9CC0-A1A2C95F2AF6@freeswitch.org> <0CE94105-05A9-48FE-A90A-E668094E4254@jerris.com> Message-ID: <277313CA-41FA-4868-9F77-BBAC35BA09E0@gmail.com> Hi Michael, Thank you for your suggestion! I believe I will try your solution as at the end the task of creating my own custom channels table required me too many efforts. I need to put somewhere the gateway name, so, this should be presence_data_cols=sip_gateway_name ? Thank you so much Il giorno 19/gen/2015, alle ore 18:12, Michael Jerris ha scritto: > if you are just messing with 1 additional column, check out using presence_data for this. > >> On Jan 19, 2015, at 9:11 AM, Luis Daniel Lucio Quiroz wrote: >> >> Ken, I dont suggest in all my answers to install FUsion. But when it >> is useful i do. >> >> Jack. Download the fusion and review the file named: >> v_xml_cdr_import.php you will find there is a code that use the >> variables lcr_* >> >> those variables are pushed with an export using dialplans. You will >> figure out how to do your own code. >> >> LD >> 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 >> >> >> 2015-01-18 14:43 GMT-05:00 Ken Rice : >>> Do not overwrite anything in there, its likely there for a reason... But adding a second table with the uuid is fine also keep in mind that if you use the channels table at master source you can clean our everything left over in the custom table every 5 minutes to reduce load on the db >>> >>> Sent from my iPhone >>> >>>> On Jan 18, 2015, at 1:28 PM, Jack Cortez wrote: >>>> >>>> Hi Ken! >>>> I'm making some tests and it could be a good idea to create my personal channel table. >>>> >>>> However, in my tests I'm using a lua script to insert a new record in my table when call start with uuid as key, and delete the record at the end of the call, but I was thinking that is should be better to use the native channel table (that is better managed by FS itself) as I have to use a lua script for this task. >>>> Instead I can even use some custom variables that could be set into the XML dialplan, right? >>>> >>>> So, do you believe it's possible to overwrite for example context or dialplan columns with my custom values? >>>> I am suggesting context and dialplan as I found that on channels table they contain same value dialplan=XML and context=router and in my opinion they can be replaced with some more interested value >>>> >>>> >>>> Otherwise the only way it's really use a script and insert/update/delete calls into my new custom channels table. >>>> >>>> Thank you! >>>> >>>> >>>> >>>> >>>> >>>>> Il giorno 18/gen/2015, alle ore 18:18, Ken Rice ha scritto: >>>>> >>>>> You could possibly do that and not cause problems, but better would be create another table that has your custome datat in it and you can key with the channel's uuid >>>>> >>>>> Sent from my iPhone >>>>> >>>>>> On Jan 18, 2015, at 4:59 AM, Jack Cortez wrote: >>>>>> >>>>>> Hi, >>>>>> I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. >>>>>> Is this possible? >>>>>> >>>>>> In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. >>>>>> >>>>>> And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. >>>>>> I'm looking for days into the wiki but I didn't found anything interesting. >>>>>> >>>>>> Thank you so much! >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >> >> _________________________________________________________________________ >> 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 From magnus.kelly at gmail.com Mon Jan 19 23:56:07 2015 From: magnus.kelly at gmail.com (Magnus Kelly) Date: Mon, 19 Jan 2015 20:56:07 +0000 Subject: [Freeswitch-users] Function - drop_dtmf Message-ID: Hello Guys, Couple of questions on the dtmf masking function that was created/extended recently - is it possible to use half way through a call ? - in terms of invoking the DTMF masking by initiating dtmf as in the below dial plan, and associated features.xml entry - would the call be able to start as normal and then on demand mask dumf? and an entry in features.xml as in - ------------snip I guess then I would need an alternative bind digit to turn the feature feature off again? Is this possible or is this only selectable at bridge time? Any thoughts? Thanks Magnus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/41711cd2/attachment.html From mike at jerris.com Tue Jan 20 00:37:48 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jan 2015 16:37:48 -0500 Subject: [Freeswitch-users] Custom Columns into MySQL Channels table In-Reply-To: <277313CA-41FA-4868-9F77-BBAC35BA09E0@gmail.com> References: <662ED501-722D-43AF-98C7-4384E03666C9@gmail.com> <41AC342E-EB3E-4B21-93C2-A568243F29A2@gmail.com> <703825C6-4B49-46A9-9CC0-A1A2C95F2AF6@freeswitch.org> <0CE94105-05A9-48FE-A90A-E668094E4254@jerris.com> <277313CA-41FA-4868-9F77-BBAC35BA09E0@gmail.com> Message-ID: <3BC68B17-95CC-40F7-A5F7-E5F1A3ECB311@jerris.com> if its only one column you can use presence_data > On Jan 19, 2015, at 3:54 PM, Jack Cortez wrote: > > Hi Michael, > Thank you for your suggestion! I believe I will try your solution as at the end the task of creating my own custom channels table required me too many efforts. > > I need to put somewhere the gateway name, so, this should be presence_data_cols=sip_gateway_name ? > > Thank you so much > > > Il giorno 19/gen/2015, alle ore 18:12, Michael Jerris ha scritto: > >> if you are just messing with 1 additional column, check out using presence_data for this. >> >>> On Jan 19, 2015, at 9:11 AM, Luis Daniel Lucio Quiroz wrote: >>> >>> Ken, I dont suggest in all my answers to install FUsion. But when it >>> is useful i do. >>> >>> Jack. Download the fusion and review the file named: >>> v_xml_cdr_import.php you will find there is a code that use the >>> variables lcr_* >>> >>> those variables are pushed with an export using dialplans. You will >>> figure out how to do your own code. >>> >>> LD >>> 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 >>> >>> >>> 2015-01-18 14:43 GMT-05:00 Ken Rice : >>>> Do not overwrite anything in there, its likely there for a reason... But adding a second table with the uuid is fine also keep in mind that if you use the channels table at master source you can clean our everything left over in the custom table every 5 minutes to reduce load on the db >>>> >>>> Sent from my iPhone >>>> >>>>> On Jan 18, 2015, at 1:28 PM, Jack Cortez wrote: >>>>> >>>>> Hi Ken! >>>>> I'm making some tests and it could be a good idea to create my personal channel table. >>>>> >>>>> However, in my tests I'm using a lua script to insert a new record in my table when call start with uuid as key, and delete the record at the end of the call, but I was thinking that is should be better to use the native channel table (that is better managed by FS itself) as I have to use a lua script for this task. >>>>> Instead I can even use some custom variables that could be set into the XML dialplan, right? >>>>> >>>>> So, do you believe it's possible to overwrite for example context or dialplan columns with my custom values? >>>>> I am suggesting context and dialplan as I found that on channels table they contain same value dialplan=XML and context=router and in my opinion they can be replaced with some more interested value >>>>> >>>>> >>>>> Otherwise the only way it's really use a script and insert/update/delete calls into my new custom channels table. >>>>> >>>>> Thank you! >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> Il giorno 18/gen/2015, alle ore 18:18, Ken Rice ha scritto: >>>>>> >>>>>> You could possibly do that and not cause problems, but better would be create another table that has your custome datat in it and you can key with the channel's uuid >>>>>> >>>>>> Sent from my iPhone >>>>>> >>>>>>> On Jan 18, 2015, at 4:59 AM, Jack Cortez wrote: >>>>>>> >>>>>>> Hi, >>>>>>> I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. >>>>>>> Is this possible? >>>>>>> >>>>>>> In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. >>>>>>> >>>>>>> And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. >>>>>>> I'm looking for days into the wiki but I didn't found anything interesting. >>>>>>> >>>>>>> Thank you so much! >>>>>>> From kris at kriskinc.com Tue Jan 20 00:38:59 2015 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 19 Jan 2015 16:38:59 -0500 Subject: [Freeswitch-users] Mod-Sofia ability to dial Alphanumeric characters in invite? In-Reply-To: References: Message-ID: Why do you have two sets of double quotes in your destination number expression? That could be causing your error. Your guess is correct; you should be able to match your expected string with something like the following: You need to escape the "+", match "C", and match any number of digits. Note the one set of double quotes surrounding the expression. On Mon, Jan 19, 2015 at 3:17 PM, Magnus Kelly wrote: > > > On 19/01/2015 16:03, "Kristian Kielhofner" wrote: > >>It's well within standard and actually required for many platforms. >> >>What does your bridge string look like? >> >>On Mon, Jan 19, 2015 at 9:24 AM, Magnus Kelly >>wrote: >>> Hello Guys, >>> >>> Is it possible to use FS with alphanumeric characters in the SIP >>>invite? >>> >>> As in I would like to send an invite addressed to +C27344xxxxxxxxxx as >>>one >>> of our upstream platforms uses a C prefix. >>> >>> What I observe with default is " [ERR] mod_sofia.c:4431 Invalid URL " >>> >>> If possible to support what would need to be changed? I am assuming that >>> this is outside the SIP standards, but confirmation would equally be >>>useful. >>> >>> Thanks >>> Magnus >>> >>> >>>_________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: > > Hello Kristian, > > I also re-read bit further and figured out it was a silly question as how > else would names be used, however I am using a simple dial plan - as in > the below - > > > > > > > > > > > > > > > And when the incoming invite that?s destined to dial out on other leg > comes as ? +C27344xxxxxxxxxx " I get the error as in ? [ERR] > mod_sofia.c:4431 Invalid URL ? > > I will continue to review but thoughts welcome. my guess is need to study > perl expressions. > Thanks > Magnus > > > > > > _________________________________________________________________________ > 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 -- Kristian Kielhofner From magnus.kelly at gmail.com Tue Jan 20 00:59:45 2015 From: magnus.kelly at gmail.com (Magnus Kelly) Date: Mon, 19 Jan 2015 21:59:45 +0000 Subject: [Freeswitch-users] Mod-Sofia ability to dial Alphanumeric characters in invite? In-Reply-To: References: Message-ID: Kristian, Appreciate the help - that fixed it. Thanks Magnus On 19/01/2015 21:38, "Kristian Kielhofner" wrote: >Why do you have two sets of double quotes in your destination number >expression? That could be causing your error. > >Your guess is correct; you should be able to match your expected >string with something like the following: > > > >You need to escape the "+", match "C", and match any number of digits. >Note the one set of double quotes surrounding the expression. > > From GeorgePhelps at gfphelps.com Tue Jan 20 01:02:57 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Mon, 19 Jan 2015 17:02:57 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <053301d03189$dc9b9a10$95d2ce30$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <14 7601d02f32$37a9f5a0$ a 6 fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> <007901d02ff5$67075090$3515f1b0$@gfphelps.com> <041401d030e5$ea3b6730$beb23590$@gfphelps.com> <04ce01d0311e$272fac40$758f04c0$@gfphelps.com> <053301d03189$dc9b9a10$95d2ce30$@gfphelps.com> Message-ID: <08b001d03433$af9d6a50$0ed83ef0$@gfphelps.com> The following set of (test) code changes seems to resolve the issue. Could someone familiar with the FreeSWITCH code base please review the changes and provide their feedback? diff switch_ivr_originate.c switch_ivr_originate.c.ORIG 1371a1372 > handle->done = 0; 1380,1382d1380 < < return NULL; < 1559d1556 < handles[i].done = 0; 1633a1631 > 1660,1667c1658 < if (hp == &handles[i]) { < continue; < } < < if (handles[i].bleg) { < switch_channel_t *channel = switch_core_session_get_channel(handles[i].bleg); < < switch_channel_set_variable(channel, "group_dial_status", "loser"); --- > if (channel) { 1669d1659 < switch_core_session_rwunlock(handles[i].bleg); 1671a1662,1664 > if (hp == &handles[i]) { > continue; > } Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Friday, January 16, 2015 7:42 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Michael Collins, Any feedback on the debug log that I uploaded yesterday? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Thursday, January 15, 2015 6:51 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? New failed call log at: http://pastebin.com/cErTyGht Scenario? Two registered extensions, 1000 and 1001. Inbound call, to simultaneously ring both extensions. Both extensions start ringing. I answer on extension 1001. The call immediately drops, and then begins ringing again in a couple of seconds. >From the log? Line #754: extension 1001 OKs its INVITE. Line #835: debug messages indicates extension 1001 answered the call. Line #854: FreeSWITCH CANCELs the call to extension 1000. Line #887: FreeSWITCH terminates the inbound call with a ?480 Temporarily Unavailable? response. Line #930: FreeSWITCH sends a BYE to extension 1001. Line #1117: a new (regenerated), inbound call request. Without simultaneous ring enabled to two extensions, i.e., ringing only extension 1001 ? the call is handled with no problems. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Thursday, January 15, 2015 12:09 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Michael Collins, I had previously tried the ?Enterprise Originate? and syntax, but it did not make any difference at the time. I retested today with that syntax, and I am still seeing the same problem. If that is the recommended configuration for my situation, I will kept that syntax in my dialplan. I also tested with a difference VoIP service provider. Better results in that the INVITE timer now runs for 28 seconds, as opposed to just 10 seconds with the previous VoIP service provider. During the 28 seconds ? with both extensions ringing ? I was able to answer one extension. However, the call disconnected just as soon as I answered it, and then immediately started ringing again. This continued until the 28 seconds ran out. I will look into this later, and gather additional logs. I also retested, with the new VoIP service provider ? and dialing just one extension works fine. Dialing just one extension also worked with the previous VoIP service provider, even with the 10 second INVITE timer, as long as I answered within the 10 seconds window. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 10:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Glad you have the book. On page 19 it covers the use of enterprise originate. I think possibly you need to use the method as discussed on page 21. Try something like this: Could just be that early media is not being ignored on both user dialout attempts. -MC On Wed, Jan 14, 2015 at 4:27 AM, George F. Phelps wrote: Michael Collins, I already have the book. Thanks! Here?s my dialplan: New log file uploaded to: http://pastebin.com/gnEpPzk9 To me, the most significant event in the log file is the SIP CANCEL message ? starting at line #321: tport.c:3023 tport_deliver() tport_deliver(0x95daa0): msg 0xad8fb0 (437 bytes) from udp/169.XX.XX.XX:5080/sip next=(nil) nta.c:2880 agent_recv_request() nta: received CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (CSeq 1) nta.c:3026 agent_recv_request() nta: CANCEL (1) is going to INVITE (1) I don?t think it?s related, but I am also curious about log file line #285: sres.c:2987 sres_query_report_error() sres(q=0x98b050): reporting error NAME_ERR for SRV _sip._udp.sip.switch2voip.us Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 2:41 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? We covered this nicely in chapter 1 of the FreeSWITCH Cookbook I'm sorry that I'm late to the party so I am missing some information. Can you pastebin not only the call log but also the dialplan code for the example in question? One other tip: it appears that the log that you are pasting is coming directly from the FreeSWITCH console. By default the console does not have debug level output enabled. Try entering the command "console loglevel debug" and you'll see way more log lines, mostly yellow text. Those lines will most likely contain the clues needed to unravel this mystery. Thanks, MC On Tue, Jan 13, 2015 at 2:42 PM, George F. Phelps wrote: New logfile uploaded to: http://pastebin.com/CFFvVarS The log contains default Freeswitch console log messages, plus a SIP trace of a failed call. BTW, both extensions were ringing ? prior to the CANCEL message (see context below). In the log I see the INVITE from my VoIP service provider: recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: ------------------------------------------------------------------------ INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: ------------------------------------------------------------------------ SIP/2.0 100 Trying (Then, subsequent INVITE messages to my two extensions. But other no messages to/from my VoIP service provider.) And then, a spontaneous CANCEL from my VoIP service provider, approximately 10 seconds after the initial INVITE message. Due to a SIP ?Timer B? timeout? Seems way too short. recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: ------------------------------------------------------------------------ CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (Freeswitch cleanup of SIP sessions to my extensions?) Bote Man--> I have two local extensions. Individually, the extensions can make and receive both internal and external calls. It?s only the simultaneous ringing for external, inbound calls that is not working at the moment. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Tuesday, January 13, 2015 2:33 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I suggest you configure and register 3 total local phones to your FS installation, configure 2 of them as the target of your simultaneous ring group, and call them with the 3rd phone. Until you can get that working, calling through a carrier is adding another layer of complexity to the problem and confusing the issue. Out of the box FreeSWITCH does not utter voice codes, they must be coming from your carrier. Also, the debug-level logs very likely tell you exactly what is happening, even though they can be staggering to decipher as a newcomer to FS. Learning how to read them pays off in so many ways, though. I find the color-coded logs on the console or viewed via FS_cli to be helpful in these instances. Bote From: George F. Phelps Sent: Tuesday, 13 January, 2015 08:10 Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I tried? ?but that did not resolve the problem. 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: Tuesday, January 13, 2015 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" wrote: David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: Thanks, George -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/277d8072/attachment-0001.html From brian at freeswitch.org Tue Jan 20 01:06:06 2015 From: brian at freeswitch.org (Brian West) Date: Mon, 19 Jan 2015 16:06:06 -0600 Subject: [Freeswitch-users] looking to pay for help in configuration In-Reply-To: References: Message-ID: consulting at freeswitch.org? ;) On Mon, Jan 19, 2015 at 7:58 AM, Moishe Grunstein wrote: > There is a special mailing list for this type of request > http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz > > > > Also Freeswitch Solutions now offers installation services, > https://freeswitch.com/cart.php?gid=4 I am not sure if they will help > with FusionPBX though. > > > > FusionPBX has their own IRC channel and support website. > > > > Thanks, > > > > Moishe Grunstein > > Tornado Computer Systems, Inc. > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com * > > [image: cid:image001.jpg at 01C72F94.9EE45D60] > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shai > Perelman > *Sent:* Monday, January 19, 2015 7:57 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] looking to pay for help in configuration > > > > Hi, > > Im new to freeswitch and the list, so hi everybody. > > I,m learning the subject of fusionpbx/freeswitch and have a server up and > running, got a few problems. and need someone to assist me on a hourly rate. > > > > is there any one interested? > > > > thanks > > Shai > > > > > > > > _________________________________________________________________________ > 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/20150119/d3319959/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/d3319959/attachment.jpg From blessendor at gmail.com Tue Jan 20 01:23:08 2015 From: blessendor at gmail.com (Alexandr Usov) Date: Tue, 20 Jan 2015 00:23:08 +0200 Subject: [Freeswitch-users] looking to pay for help in configuration In-Reply-To: References: Message-ID: I can try to help you with FusionPBX. You can contact me direct to blessendor at gmail.com 2015-01-20 0:06 GMT+02:00 Brian West : > consulting at freeswitch.org? ;) > > On Mon, Jan 19, 2015 at 7:58 AM, Moishe Grunstein > wrote: > >> There is a special mailing list for this type of request >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz >> >> >> >> Also Freeswitch Solutions now offers installation services, >> https://freeswitch.com/cart.php?gid=4 I am not sure if they will help >> with FusionPBX though. >> >> >> >> FusionPBX has their own IRC channel and support website. >> >> >> >> Thanks, >> >> >> >> Moishe Grunstein >> >> Tornado Computer Systems, Inc. >> >> 212.400.7650 888.IPPBX.US >> *Service Request Email: support at nysolutions.com >> * >> >> [image: cid:image001.jpg at 01C72F94.9EE45D60] >> >> Computer Networking * Managed Services * IP Video Surveillance * Network >> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >> Security * Site Surveys * CMS >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shai >> Perelman >> *Sent:* Monday, January 19, 2015 7:57 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] looking to pay for help in configuration >> >> >> >> Hi, >> >> Im new to freeswitch and the list, so hi everybody. >> >> I,m learning the subject of fusionpbx/freeswitch and have a server up and >> running, got a few problems. and need someone to assist me on a hourly rate. >> >> >> >> is there any one interested? >> >> >> >> thanks >> >> Shai >> >> >> >> >> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/aa409561/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/aa409561/attachment-0001.jpg From ssinyagin at gmail.com Tue Jan 20 01:24:35 2015 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Mon, 19 Jan 2015 23:24:35 +0100 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: References: Message-ID: I run my PBX on a VPS at softronics.ch , which is not exactly EU, but it's in the middle of Europe. The support staff communicates in English freely. I've been with them for many years, and my PBX is hosted there since 2011, without problems. On Sun, Jan 18, 2015 at 8:42 PM, Zolt?n Szab? wrote: > Hi, > > Does anyone have experience running Freeswitch at OVH? I have a VPS Classic > 1 but the audio was really bad there. It was faltered. (tested with multiple > clients from multiple locations) This is an OpenVZ virtualization if this > counts. > > I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was also a > little bit more, and it was a vServer virtualization, it worked perfectly. > > If you run your FS on a VPS without any problem, can you please recommend me > some providers who has servers in the EU? I don't expect high call volume, > just like 5-10 concurrent calls on 1-2 SIP trunks which is basically > nothing. > > Many thanks, > Zoltan > > _________________________________________________________________________ > 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 From mike at jerris.com Tue Jan 20 01:29:27 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jan 2015 17:29:27 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <08b001d03433$af9d6a50$0ed83ef0$@gfphelps.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <14 7601d02f32$37a9f5a0$ a 6 fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> <007901d02ff5$67075090$3515f1b0$@gfphelps.com> <041401d030e5$ea3b6730$beb23590$@gfphelps.com> <04ce01d0311e$272fac40$758f04c0$@gfphelps.com> <053301d03189$dc9b9a10$95d2ce30$@gfphelps.com> <08b001d03433$af9d6a50$0ed83ef0$@gfphelps.com> Message-ID: <32C551E0-A264-4F0C-9BB6-D4BB5E8D8271@jerris.com> Please submit patches for review by creating a jira and a pull request in stash. > On Jan 19, 2015, at 5:02 PM, George F. Phelps wrote: > > The following set of (test) code changes seems to resolve the issue. Could someone familiar with the FreeSWITCH code base please review the changes and provide their feedback? > > diff switch_ivr_originate.c switch_ivr_originate.c.ORIG > 1371a1372 > > handle->done = 0; > 1380,1382d1380 > < > < return NULL; > < > 1559d1556 > < handles[i].done = 0; > 1633a1631 > > > 1660,1667c1658 > < if (hp == &handles[i]) { > < continue; > < } > < > < if (handles[i].bleg) { > < switch_channel_t *channel = switch_core_session_get_channel(handles[i].bleg); > < > < switch_channel_set_variable(channel, "group_dial_status", "loser"); > --- > > if (channel) { > 1669d1659 > < switch_core_session_rwunlock(handles[i].bleg); > 1671a1662,1664 > > if (hp == &handles[i]) { > > continue; > > } > > Thanks, > > George > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps > Sent: Friday, January 16, 2015 7:42 AM > To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > Michael Collins, > > Any feedback on the debug log that I uploaded yesterday? > > Thanks, > > George > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of George F. Phelps > Sent: Thursday, January 15, 2015 6:51 PM > To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > New failed call log at: > > http://pastebin.com/cErTyGht > > Scenario? Two registered extensions, 1000 and 1001. Inbound call, to simultaneously ring both extensions. Both extensions start ringing. I answer on extension 1001. The call immediately drops, and then begins ringing again in a couple of seconds. > > From the log? > > Line #754: extension 1001 OKs its INVITE. > > Line #835: debug messages indicates extension 1001 answered the call. > > Line #854: FreeSWITCH CANCELs the call to extension 1000. > > Line #887: FreeSWITCH terminates the inbound call with a ?480 Temporarily Unavailable? response. > > Line #930: FreeSWITCH sends a BYE to extension 1001. > > Line #1117: a new (regenerated), inbound call request. > > Without simultaneous ring enabled to two extensions, i.e., ringing only extension 1001 ? the call is handled with no problems. > > Thanks, > > George > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of George F. Phelps > Sent: Thursday, January 15, 2015 12:09 PM > To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > Michael Collins, > > I had previously tried the ?Enterprise Originate? and syntax, but it did not make any difference at the time. I retested today with that syntax, and I am still seeing the same problem. If that is the recommended configuration for my situation, I will kept that syntax in my dialplan. > > I also tested with a difference VoIP service provider. Better results in that the INVITE timer now runs for 28 seconds, as opposed to just 10 seconds with the previous VoIP service provider. During the 28 seconds ? with both extensions ringing ? I was able to answer one extension. However, the call disconnected just as soon as I answered it, and then immediately started ringing again. This continued until the 28 seconds ran out. I will look into this later, and gather additional logs. > > I also retested, with the new VoIP service provider ? and dialing just one extension works fine. Dialing just one extension also worked with the previous VoIP service provider, even with the 10 second INVITE timer, as long as I answered within the 10 seconds window. > > Thanks, > > George > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Michael Collins > Sent: Wednesday, January 14, 2015 10:26 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > Glad you have the book. On page 19 it covers the use of enterprise originate. I think possibly you need to use the method as discussed on page 21. Try something like this: > > > > Could just be that early media is not being ignored on both user dialout attempts. > -MC > > On Wed, Jan 14, 2015 at 4:27 AM, George F. Phelps > wrote: > Michael Collins, > > I already have the book. Thanks! > > Here?s my dialplan: > > > > > > > > > > > New log file uploaded to: > > http://pastebin.com/gnEpPzk9 > > To me, the most significant event in the log file is the SIP CANCEL message ? starting at line #321: > > tport.c:3023 tport_deliver() tport_deliver(0x95daa0): msg 0xad8fb0 (437 bytes) from udp/169.XX.XX.XX:5080/sip next=(nil) > nta.c:2880 agent_recv_request() nta: received CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (CSeq 1) > nta.c:3026 agent_recv_request() nta: CANCEL (1) is going to INVITE (1) > > I don?t think it?s related, but I am also curious about log file line #285: > > sres.c:2987 sres_query_report_error() sres(q=0x98b050): reporting error NAME_ERR for SRV _sip._udp.sip.switch2voip.us > > Thanks, > > George > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Michael Collins > Sent: Wednesday, January 14, 2015 2:41 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > > We covered this nicely in chapter 1 of the FreeSWITCH Cookbook > > > I'm sorry that I'm late to the party so I am missing some information. Can you pastebin not only the call log but also the dialplan code for the example in question? One other tip: it appears that the log that you are pasting is coming directly from the FreeSWITCH console. By default the console does not have debug level output enabled. Try entering the command "console loglevel debug" and you'll see way more log lines, mostly yellow text. Those lines will most likely contain the clues needed to unravel this mystery. > > Thanks, > MC > > > On Tue, Jan 13, 2015 at 2:42 PM, George F. Phelps > wrote: > New logfile uploaded to: > > http://pastebin.com/CFFvVarS > > The log contains default Freeswitch console log messages, plus a SIP trace of a failed call. BTW, both extensions were ringing ? prior to the CANCEL message (see context below). > > In the log I see the INVITE from my VoIP service provider: > > recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: > ------------------------------------------------------------------------ > INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 > > send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > > (Then, subsequent INVITE messages to my two extensions. But other no messages to/from my VoIP service provider.) > > And then, a spontaneous CANCEL from my VoIP service provider, approximately 10 seconds after the initial INVITE message. Due to a SIP ?Timer B? timeout? Seems way too short. > > recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: > ------------------------------------------------------------------------ > CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 > > (Freeswitch cleanup of SIP sessions to my extensions?) > > > Bote Man--> I have two local extensions. Individually, the extensions can make and receive both internal and external calls. It?s only the simultaneous ringing for external, inbound calls that is not working at the moment. > > Thanks, > > George > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Bote Man > Sent: Tuesday, January 13, 2015 2:33 PM > > To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > I suggest you configure and register 3 total local phones to your FS installation, configure 2 of them as the target of your simultaneous ring group, and call them with the 3rd phone. Until you can get that working, calling through a carrier is adding another layer of complexity to the problem and confusing the issue. > > Out of the box FreeSWITCH does not utter voice codes, they must be coming from your carrier. > > Also, the debug-level logs very likely tell you exactly what is happening, even though they can be staggering to decipher as a newcomer to FS. Learning how to read them pays off in so many ways, though. I find the color-coded logs on the console or viewed via FS_cli to be helpful in these instances. > > Bote > > > From: George F. Phelps > Sent: Tuesday, 13 January, 2015 08:10 > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > I tried? > > ?but that did not resolve the problem. > > 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: Tuesday, January 13, 2015 7:58 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > Correct, first endpoint providing audio wins, but you're using ignore_early_media... > Try using > > Which is global. And I believe in the dial string also is. > But try it anyway. > > On Jan 13, 2015 1:50 PM, "George F. Phelps" > wrote: > David Govea, > > It appears that the essence of the problem is: > > [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] > [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] > [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] > [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] > > Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. > > What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? > > > > > ?but without success. Or does setting ignore_early_media have to be done somewhere else? > > 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: Tuesday, January 13, 2015 6:36 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > You need to have both extensions registered. Register both and try again and paste de log. > > On Jan 13, 2015 12:30 PM, "George F. Phelps" > wrote: > For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. > > 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: Monday, January 12, 2015 6:16 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > Is 1000 registered? The log says it's not registered... > > On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps > wrote: > David Govea, > > I uploaded a new Freeswitch debug logfile at: > > http://pastebin.com/v17SyXhh > > Notes > > Only extension 1001 was registered for this test. > > Dialstring segment: > > I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. > > I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. > > I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? > > My effective codec is G711U ? fully supported throughout the call chain. > > 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: Monday, January 12, 2015 7:15 AM > > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... > > On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea > wrote: > Are you using freeswitch with its default config or did you install something like fusionpbx? > Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. > (Are you sure your codecs are correct?) > > On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps > wrote: > David Govea, > > Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. > > How do I track down the meaning of ?231?? > > 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: Monday, January 12, 2015 6:14 AM > > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > You can also try: > > bridge user/1001:_:user/1002 > > On Jan 12, 2015 12:04 PM, "George F. Phelps" > wrote: > David Govea, > > That syntax, with more than one extension specified, causes the following Freeswitch warning log message: > > [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. > > However, the call ? to only the first extension on the list ? does work. > > 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: Monday, January 12, 2015 3:21 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > try this: > > > > > > > > > > On Jan 12, 2015 4:33 AM, "George F. Phelps" > wrote: > Here you go: > > > > > > > > > > > Symbol ${domain} resolves to the local LAN, IP address. > > 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, January 11, 2015 10:18 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > Cab you paste your dialplan? > Also, never EVER show your ip addresses. > > On Jan 12, 2015 2:48 AM, "George F. Phelps" > wrote: > Yes, I tested with that dialstring. My extension was registered, and online. > > The call disconnects with verbal error code ?231?. The associated logfile is at: > > http://pastebin.com/BeWhhgSU > > 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, January 11, 2015 8:31 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? > > On Jan 12, 2015 2:01 AM, "George F. Phelps" > wrote: > David Govea, > > I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? > > I am trying to get the first extension to work with ?bridge.? > > This Freeswitch example shows bridging (I thought?) to two (2) extensions: > > Calling multiple destinations > By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) > If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: > > > 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, January 11, 2015 7:31 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > Sorry, > > I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? > > On Jan 12, 2015 1:29 AM, "George F. Phelps" > wrote: > David Govea, > > Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. > > I created a pastebin.com of the failed call log, at: > > http://pastebin.com/BeWhhgSU > > A reminder that this ?transfer? statement works: > > > > 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, January 11, 2015 4:19 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? > > https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user > That's: > > > > Note the % sign..., not @ > > On Jan 11, 2015 10:09 PM, "George F. Phelps" > wrote: > Can someone help me with my question? > > Thanks, > > George > > From: George F. Phelps [mailto:GeorgePhelps at gfphelps.com ] > Sent: Saturday, January 10, 2015 12:02 PM > To: freeswitch-users at lists.freeswitch.org > Subject: How to Bridge To Local Extensions? > > The ?transfer? statement, shown below, works (in my inbound dialplan): > > > > What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: > > > > My extensions are effectively default values and in the default directory location. For example: > > more /usr/local/freeswitch/conf/directory/default/1001.xml > > > > > > > > > > > > > > > > > > > > > My goal is to configure simultaneous ringing for multiple extensions: > > > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150119/0a362c85/attachment-0001.html From john.nash778 at gmail.com Tue Jan 20 04:21:58 2015 From: john.nash778 at gmail.com (John Nash) Date: Tue, 20 Jan 2015 06:51:58 +0530 Subject: [Freeswitch-users] Sending call to outbound proxy In-Reply-To: References: Message-ID: sorry fs_path worked. I posted this question in a hurry I think. Issue was that I was not using "sip:" in uri so may be it was trying to parse uri itself. On Mon, Jan 19, 2015 at 11:20 PM, Godson Gera wrote: > set fs_path in originate or bridge string. Or use outbound proxy option in > sofia profile. > > http://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path > > > On Sat, Jan 17, 2015 at 9:57 PM, John Nash wrote: > >> I am trying to test a case where I am receiving call from opensips to >> freeswitch and need to route back to opensips again without changing >> Request URI. How can I do it? >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Thanks & Regards, > Godson Gera > FreeSWITCH Consultant > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/aff5ed61/attachment.html From nagalenoj at gmail.com Tue Jan 20 09:52:29 2015 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Tue, 20 Jan 2015 12:22:29 +0530 Subject: [Freeswitch-users] spandsp_start_dtmf detects # key many times In-Reply-To: <33221776-F8EB-4329-BC1A-5413478DBD16@jerris.com> References: <33221776-F8EB-4329-BC1A-5413478DBD16@jerris.com> Message-ID: When I changed the Aastra phone settings to use SIP INFO, am not facing this issue. But, when I change it to use RTP for DTMF, then '#' is getting detected more times. I did a pcap and listened to the audio, I could hear only one tone(beep) for each key pressed. Is there anything other configuration to get it working in RTP? On Mon, Jan 19, 2015 at 10:40 PM, Michael Jerris wrote: > If you are using a sip phone, you are MUCH better off using non audio > based dtmf like rfc-2833. If you are already using this, its possible that > what is happening is the phone is sending the dtmf as audio and 2833, at > which point, you should not have to use spandsp_start_dtmf at all. If the > phone is sending only audio, and is not capable of using any other method, > you will need to pull a pcap of this call, use wireshark to extract the > audio, and do some analysis of the audio to figure out why it is detecting > multiple digits. > > > On Jan 19, 2015, at 4:38 AM, Nagalenoj H. wrote: > > Hi, > > When I use spandsp_start_dtmf along with play_and_get_digits, '#' is > getting detected many times, though it is pressed once. > > Other DTMFs are getting detected properly, whereas the '#' is getting > detected 2 or 3 times. > > I have checked the SIP phone's settings, I do not find anything wrong > there. > > What could be the reason? Is this because of some SIP phone setting? Here > is the freeswitch log, > > 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [1] > 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF 1:2000 > 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [1], duration = 191 ms > 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [4] > 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 > 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [4], duration = 153 ms > 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [7] > 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 > 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [7], duration = 204 ms > 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [4] > 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 > 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [4], duration = 127 ms > 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send signal > sofia/internal/sip:278 at 192.168.2.86:5060 [BREAK] > 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [7] > 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 > 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [7], duration = 191 ms > 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec > Activated L16 at 8000hz 1 channels 20ms > 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [#] > 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > EXECUTE sofia/internal/392 at 3427.vbiz.mundio.com flush_dtmf() > 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec > Activated L16 at 8000hz 1 channels 30ms > 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [#], duration = 38 ms > 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [#] > 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done playing > file file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec > Activated L16 at 8000hz 1 channels 30ms > 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [#], duration = 25 ms > 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [#] > 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [#], duration = 76 ms > > > > _________________________________________________________________________ > 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 > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/86d2bac2/attachment.html From covici at ccs.covici.com Tue Jan 20 10:07:41 2015 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 20 Jan 2015 02:07:41 -0500 Subject: [Freeswitch-users] spandsp_start_dtmf detects # key many times In-Reply-To: References: <33221776-F8EB-4329-BC1A-5413478DBD16@jerris.com> Message-ID: <8404.1421737661@ccs.covici.com> Why not use rfc2833 instead? Nagalenoj H. wrote: > When I changed the Aastra phone settings to use SIP INFO, am not facing > this issue. But, when I change it to use RTP for DTMF, then '#' is getting > detected more times. > > I did a pcap and listened to the audio, I could hear only one tone(beep) > for each key pressed. Is there anything other configuration to get it > working in RTP? > > On Mon, Jan 19, 2015 at 10:40 PM, Michael Jerris wrote: > > > If you are using a sip phone, you are MUCH better off using non audio > > based dtmf like rfc-2833. If you are already using this, its possible that > > what is happening is the phone is sending the dtmf as audio and 2833, at > > which point, you should not have to use spandsp_start_dtmf at all. If the > > phone is sending only audio, and is not capable of using any other method, > > you will need to pull a pcap of this call, use wireshark to extract the > > audio, and do some analysis of the audio to figure out why it is detecting > > multiple digits. > > > > > > On Jan 19, 2015, at 4:38 AM, Nagalenoj H. wrote: > > > > Hi, > > > > When I use spandsp_start_dtmf along with play_and_get_digits, '#' is > > getting detected many times, though it is pressed once. > > > > Other DTMFs are getting detected properly, whereas the '#' is getting > > detected 2 or 3 times. > > > > I have checked the SIP phone's settings, I do not find anything wrong > > there. > > > > What could be the reason? Is this because of some SIP phone setting? Here > > is the freeswitch log, > > > > 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > DETECTED: [1] > > 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF 1:2000 > > 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > DETECTED: [1], duration = 191 ms > > 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > DETECTED: [4] > > 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 > > 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > DETECTED: [4], duration = 153 ms > > 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > DETECTED: [7] > > 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 > > 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > DETECTED: [7], duration = 204 ms > > 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > DETECTED: [4] > > 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 > > 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > DETECTED: [4], duration = 127 ms > > 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send signal > > sofia/internal/sip:278 at 192.168.2.86:5060 [BREAK] > > 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > DETECTED: [7] > > 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 > > 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > DETECTED: [7], duration = 191 ms > > 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec > > Activated L16 at 8000hz 1 channels 20ms > > 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > DETECTED: [#] > > 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > > EXECUTE sofia/internal/392 at 3427.vbiz.mundio.com flush_dtmf() > > 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec > > Activated L16 at 8000hz 1 channels 30ms > > 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > DETECTED: [#], duration = 38 ms > > 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > DETECTED: [#] > > 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done playing > > file file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec > > Activated L16 at 8000hz 1 channels 30ms > > 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > DETECTED: [#], duration = 25 ms > > 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > DETECTED: [#] > > 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > > 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > DETECTED: [#], duration = 76 ms > > > > > > > > _________________________________________________________________________ > > 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 > > > > > > -- > Regards, > Nagalenoj H. > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From nagalenoj at gmail.com Tue Jan 20 10:54:53 2015 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Tue, 20 Jan 2015 13:24:53 +0530 Subject: [Freeswitch-users] spandsp_start_dtmf detects # key many times In-Reply-To: <8404.1421737661@ccs.covici.com> References: <33221776-F8EB-4329-BC1A-5413478DBD16@jerris.com> <8404.1421737661@ccs.covici.com> Message-ID: My phone does not have an option to select RFC2833. On Tue, Jan 20, 2015 at 12:37 PM, wrote: > Why not use rfc2833 instead? > > Nagalenoj H. wrote: > > > When I changed the Aastra phone settings to use SIP INFO, am not facing > > this issue. But, when I change it to use RTP for DTMF, then '#' is > getting > > detected more times. > > > > I did a pcap and listened to the audio, I could hear only one tone(beep) > > for each key pressed. Is there anything other configuration to get it > > working in RTP? > > > > On Mon, Jan 19, 2015 at 10:40 PM, Michael Jerris > wrote: > > > > > If you are using a sip phone, you are MUCH better off using non audio > > > based dtmf like rfc-2833. If you are already using this, its possible > that > > > what is happening is the phone is sending the dtmf as audio and 2833, > at > > > which point, you should not have to use spandsp_start_dtmf at all. If > the > > > phone is sending only audio, and is not capable of using any other > method, > > > you will need to pull a pcap of this call, use wireshark to extract the > > > audio, and do some analysis of the audio to figure out why it is > detecting > > > multiple digits. > > > > > > > > > On Jan 19, 2015, at 4:38 AM, Nagalenoj H. wrote: > > > > > > Hi, > > > > > > When I use spandsp_start_dtmf along with play_and_get_digits, '#' is > > > getting detected many times, though it is pressed once. > > > > > > Other DTMFs are getting detected properly, whereas the '#' is getting > > > detected 2 or 3 times. > > > > > > I have checked the SIP phone's settings, I do not find anything wrong > > > there. > > > > > > What could be the reason? Is this because of some SIP phone setting? > Here > > > is the freeswitch log, > > > > > > 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [1] > > > 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF > 1:2000 > > > 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [1], duration = 191 ms > > > 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [4] > > > 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF > 4:2000 > > > 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [4], duration = 153 ms > > > 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [7] > > > 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF > 7:2000 > > > 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [7], duration = 204 ms > > > 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [4] > > > 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF > 4:2000 > > > 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [4], duration = 127 ms > > > 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send > signal > > > sofia/internal/sip:278 at 192.168.2.86:5060 [BREAK] > > > 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [7] > > > 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF > 7:2000 > > > 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [7], duration = 191 ms > > > 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec > > > Activated L16 at 8000hz 1 channels 20ms > > > 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [#] > > > 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF > #:2000 > > > EXECUTE sofia/internal/392 at 3427.vbiz.mundio.com flush_dtmf() > > > 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec > > > Activated L16 at 8000hz 1 channels 30ms > > > 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [#], duration = 38 ms > > > 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [#] > > > 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF > #:2000 > > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done > playing > > > file > file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav > > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec > > > Activated L16 at 8000hz 1 channels 30ms > > > 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [#], duration = 25 ms > > > 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [#] > > > 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF > #:2000 > > > 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [#], duration = 76 ms > > > > > > > > > > > > > _________________________________________________________________________ > > > 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 > > > > > > > > > > > -- > > Regards, > > Nagalenoj H. > > > > ---------------------------------------------------- > > Alternatives: > > > > ---------------------------------------------------- > > _________________________________________________________________________ > > 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 > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > 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 > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/e2a05871/attachment-0001.html From brian at freeswitch.org Tue Jan 20 10:58:42 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 20 Jan 2015 01:58:42 -0600 Subject: [Freeswitch-users] spandsp_start_dtmf detects # key many times In-Reply-To: References: <33221776-F8EB-4329-BC1A-5413478DBD16@jerris.com> <8404.1421737661@ccs.covici.com> Message-ID: It should, maybe the call it AVT? On Tuesday, January 20, 2015, Nagalenoj H. wrote: > My phone does not have an option to select RFC2833. > > On Tue, Jan 20, 2015 at 12:37 PM, > wrote: > >> Why not use rfc2833 instead? >> >> Nagalenoj H. > > wrote: >> >> > When I changed the Aastra phone settings to use SIP INFO, am not facing >> > this issue. But, when I change it to use RTP for DTMF, then '#' is >> getting >> > detected more times. >> > >> > I did a pcap and listened to the audio, I could hear only one tone(beep) >> > for each key pressed. Is there anything other configuration to get it >> > working in RTP? >> > >> > On Mon, Jan 19, 2015 at 10:40 PM, Michael Jerris > > wrote: >> > >> > > If you are using a sip phone, you are MUCH better off using non audio >> > > based dtmf like rfc-2833. If you are already using this, its >> possible that >> > > what is happening is the phone is sending the dtmf as audio and 2833, >> at >> > > which point, you should not have to use spandsp_start_dtmf at all. >> If the >> > > phone is sending only audio, and is not capable of using any other >> method, >> > > you will need to pull a pcap of this call, use wireshark to extract >> the >> > > audio, and do some analysis of the audio to figure out why it is >> detecting >> > > multiple digits. >> > > >> > > >> > > On Jan 19, 2015, at 4:38 AM, Nagalenoj H. > > wrote: >> > > >> > > Hi, >> > > >> > > When I use spandsp_start_dtmf along with play_and_get_digits, '#' is >> > > getting detected many times, though it is pressed once. >> > > >> > > Other DTMFs are getting detected properly, whereas the '#' is getting >> > > detected 2 or 3 times. >> > > >> > > I have checked the SIP phone's settings, I do not find anything wrong >> > > there. >> > > >> > > What could be the reason? Is this because of some SIP phone setting? >> Here >> > > is the freeswitch log, >> > > >> > > 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [1] >> > > 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF >> 1:2000 >> > > 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [1], duration = 191 ms >> > > 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [4] >> > > 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF >> 4:2000 >> > > 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [4], duration = 153 ms >> > > 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [7] >> > > 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF >> 7:2000 >> > > 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [7], duration = 204 ms >> > > 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [4] >> > > 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF >> 4:2000 >> > > 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [4], duration = 127 ms >> > > 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send >> signal >> > > sofia/internal/sip:278 at 192.168.2.86:5060 [BREAK] >> > > 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [7] >> > > 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF >> 7:2000 >> > > 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [7], duration = 191 ms >> > > 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec >> > > Activated L16 at 8000hz 1 channels 20ms >> > > 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [#] >> > > 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF >> #:2000 >> > > EXECUTE sofia/internal/392 at 3427.vbiz.mundio.com >> flush_dtmf() >> > > 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec >> > > Activated L16 at 8000hz 1 channels 30ms >> > > 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [#], duration = 38 ms >> > > 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [#] >> > > 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF >> #:2000 >> > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done >> playing >> > > file >> file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav >> > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec >> > > Activated L16 at 8000hz 1 channels 30ms >> > > 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [#], duration = 25 ms >> > > 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [#] >> > > 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF >> #:2000 >> > > 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [#], duration = 76 ms >> > > >> > > >> > > >> > > >> _________________________________________________________________________ >> > > 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 >> > > >> > >> > >> > >> > -- >> > Regards, >> > Nagalenoj H. >> > >> > ---------------------------------------------------- >> > Alternatives: >> > >> > ---------------------------------------------------- >> > >> _________________________________________________________________________ >> > 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 >> -- >> Your life is like a penny. You're going to lose it. The question is: >> How do >> you spend it? >> >> John Covici >> covici at ccs.covici.com >> >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Regards, > Nagalenoj H. > -- *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/20150120/a3102fd0/attachment.html From kamil.nigmatullin at gmail.com Tue Jan 20 11:39:06 2015 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Tue, 20 Jan 2015 14:39:06 +0600 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: References: Message-ID: Yeah we also had some quality problems on openvz. So we had to move to xen. Now it is very stable and what is more impotant is upgradable 20 ???. 2015 ?. 4:25 ???????????? "Stanislav Sinyagin" ???????: > I run my PBX on a VPS at softronics.ch , which is not exactly EU, but > it's in the middle of Europe. The support staff communicates in > English freely. I've been with them for many years, and my PBX is > hosted there since 2011, without problems. > > > > On Sun, Jan 18, 2015 at 8:42 PM, Zolt?n Szab? wrote: > > Hi, > > > > Does anyone have experience running Freeswitch at OVH? I have a VPS > Classic > > 1 but the audio was really bad there. It was faltered. (tested with > multiple > > clients from multiple locations) This is an OpenVZ virtualization if this > > counts. > > > > I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was > also a > > little bit more, and it was a vServer virtualization, it worked > perfectly. > > > > If you run your FS on a VPS without any problem, can you please > recommend me > > some providers who has servers in the EU? I don't expect high call > volume, > > just like 5-10 concurrent calls on 1-2 SIP trunks which is basically > > nothing. > > > > Many thanks, > > Zoltan > > > > _________________________________________________________________________ > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/b83bbabd/attachment-0001.html From covici at ccs.covici.com Tue Jan 20 12:51:12 2015 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 20 Jan 2015 04:51:12 -0500 Subject: [Freeswitch-users] spandsp_start_dtmf detects # key many times In-Reply-To: References: <33221776-F8EB-4329-BC1A-5413478DBD16@jerris.com> <8404.1421737661@ccs.covici.com> Message-ID: <31159.1421747472@ccs.covici.com> I would get another phone, then. Nagalenoj H. wrote: > My phone does not have an option to select RFC2833. > > On Tue, Jan 20, 2015 at 12:37 PM, wrote: > > > Why not use rfc2833 instead? > > > > Nagalenoj H. wrote: > > > > > When I changed the Aastra phone settings to use SIP INFO, am not facing > > > this issue. But, when I change it to use RTP for DTMF, then '#' is > > getting > > > detected more times. > > > > > > I did a pcap and listened to the audio, I could hear only one tone(beep) > > > for each key pressed. Is there anything other configuration to get it > > > working in RTP? > > > > > > On Mon, Jan 19, 2015 at 10:40 PM, Michael Jerris > > wrote: > > > > > > > If you are using a sip phone, you are MUCH better off using non audio > > > > based dtmf like rfc-2833. If you are already using this, its possible > > that > > > > what is happening is the phone is sending the dtmf as audio and 2833, > > at > > > > which point, you should not have to use spandsp_start_dtmf at all. If > > the > > > > phone is sending only audio, and is not capable of using any other > > method, > > > > you will need to pull a pcap of this call, use wireshark to extract the > > > > audio, and do some analysis of the audio to figure out why it is > > detecting > > > > multiple digits. > > > > > > > > > > > > On Jan 19, 2015, at 4:38 AM, Nagalenoj H. wrote: > > > > > > > > Hi, > > > > > > > > When I use spandsp_start_dtmf along with play_and_get_digits, '#' is > > > > getting detected many times, though it is pressed once. > > > > > > > > Other DTMFs are getting detected properly, whereas the '#' is getting > > > > detected 2 or 3 times. > > > > > > > > I have checked the SIP phone's settings, I do not find anything wrong > > > > there. > > > > > > > > What could be the reason? Is this because of some SIP phone setting? > > Here > > > > is the freeswitch log, > > > > > > > > 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > > DETECTED: [1] > > > > 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF > > 1:2000 > > > > 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > > DETECTED: [1], duration = 191 ms > > > > 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > > DETECTED: [4] > > > > 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF > > 4:2000 > > > > 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > > DETECTED: [4], duration = 153 ms > > > > 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > > DETECTED: [7] > > > > 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF > > 7:2000 > > > > 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > > DETECTED: [7], duration = 204 ms > > > > 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > > DETECTED: [4] > > > > 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF > > 4:2000 > > > > 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > > DETECTED: [4], duration = 127 ms > > > > 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send > > signal > > > > sofia/internal/sip:278 at 192.168.2.86:5060 [BREAK] > > > > 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > > DETECTED: [7] > > > > 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF > > 7:2000 > > > > 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > > DETECTED: [7], duration = 191 ms > > > > 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec > > > > Activated L16 at 8000hz 1 channels 20ms > > > > 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > > DETECTED: [#] > > > > 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF > > #:2000 > > > > EXECUTE sofia/internal/392 at 3427.vbiz.mundio.com flush_dtmf() > > > > 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec > > > > Activated L16 at 8000hz 1 channels 30ms > > > > 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > > DETECTED: [#], duration = 38 ms > > > > 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > > DETECTED: [#] > > > > 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF > > #:2000 > > > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done > > playing > > > > file > > file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav > > > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec > > > > Activated L16 at 8000hz 1 channels 30ms > > > > 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > > DETECTED: [#], duration = 25 ms > > > > 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > > DETECTED: [#] > > > > 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF > > #:2000 > > > > 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > > DETECTED: [#], duration = 76 ms > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > 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 > > > > > > > > > > > > > > > > -- > > > Regards, > > > Nagalenoj H. > > > > > > ---------------------------------------------------- > > > Alternatives: > > > > > > ---------------------------------------------------- > > > _________________________________________________________________________ > > > 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 > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici > > covici at ccs.covici.com > > > > _________________________________________________________________________ > > 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 > > > > > > -- > Regards, > Nagalenoj H. > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From ahabiba at gmail.com Tue Jan 20 13:49:20 2015 From: ahabiba at gmail.com (Ahmed habiba) Date: Tue, 20 Jan 2015 13:49:20 +0300 Subject: [Freeswitch-users] Performance test using Sipp In-Reply-To: References: Message-ID: Dears, I'm looking how can I use sipp for freeswitch performance testing in term of number of concurrent calls. Your kind help will be appreciated. Thanks, Ahmed Habiba. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/7a3d3f8b/attachment.html From gb at cm.nl Tue Jan 20 14:40:20 2015 From: gb at cm.nl (Grant Bagdasarian) Date: Tue, 20 Jan 2015 11:40:20 +0000 Subject: [Freeswitch-users] Performance test using Sipp In-Reply-To: References: Message-ID: Hello Ahmed, Take a look at sippy-cup. It makes creating scenarios for Sipp much easier. https://mojolingo.com/blog/2013/introducing-sippy-cup-sipp-load-testing-made-easy/ Regards, Grant From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ahmed habiba Sent: Tuesday, January 20, 2015 11:49 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Performance test using Sipp Dears, I'm looking how can I use sipp for freeswitch performance testing in term of number of concurrent calls. Your kind help will be appreciated. Thanks, Ahmed Habiba. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/02cc5398/attachment.html From john.nash778 at gmail.com Tue Jan 20 15:04:43 2015 From: john.nash778 at gmail.com (John Nash) Date: Tue, 20 Jan 2015 17:34:43 +0530 Subject: [Freeswitch-users] Sound files from cache Message-ID: I am using "playback" to play native sound file () My question is when it plays the file does it read file every time from the disk? or there is some way to "Load at the startup" and play directly from memory. I also saw the "mod_http_cache" but not sure if I should use it (I am unable to understand in what scenario it will be logical to use it) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/353cf611/attachment.html From italorossib at gmail.com Tue Jan 20 16:49:41 2015 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 20 Jan 2015 10:49:41 -0300 Subject: [Freeswitch-users] Sound files from cache In-Reply-To: References: Message-ID: Are you having problem playing files? Why worry with this? On Tue, Jan 20, 2015 at 9:04 AM, John Nash wrote: > I am using "playback" to play native sound file ( application="playback" data="/tmp/hello"/>) > > My question is when it plays the file does it read file every time from > the disk? or there is some way to "Load at the startup" and play directly > from memory. > Yes, it does. You can put your file in a ram disk. I also saw the "mod_http_cache" but not sure if I should use it (I am > unable to understand in what scenario it will be logical to use it) > It caches a remote file (http) in your local filesystem. > _________________________________________________________________________ > 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 > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/077dc906/attachment-0001.html From mike at jerris.com Tue Jan 20 16:56:22 2015 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jan 2015 08:56:22 -0500 Subject: [Freeswitch-users] spandsp_start_dtmf detects # key many times In-Reply-To: References: <33221776-F8EB-4329-BC1A-5413478DBD16@jerris.com> <8404.1421737661@ccs.covici.com> Message-ID: <82880EB6-F9B2-4FAA-B709-A403612A869B@jerris.com> Which phone is it? look for a checkbox about force 2833 out of band dtmf. > On Jan 20, 2015, at 2:54 AM, Nagalenoj H. wrote: > > My phone does not have an option to select RFC2833. > > On Tue, Jan 20, 2015 at 12:37 PM, > wrote: > Why not use rfc2833 instead? > > Nagalenoj H. > wrote: > > > When I changed the Aastra phone settings to use SIP INFO, am not facing > > this issue. But, when I change it to use RTP for DTMF, then '#' is getting > > detected more times. > > > > I did a pcap and listened to the audio, I could hear only one tone(beep) > > for each key pressed. Is there anything other configuration to get it > > working in RTP? > > > > On Mon, Jan 19, 2015 at 10:40 PM, Michael Jerris > wrote: > > > > > If you are using a sip phone, you are MUCH better off using non audio > > > based dtmf like rfc-2833. If you are already using this, its possible that > > > what is happening is the phone is sending the dtmf as audio and 2833, at > > > which point, you should not have to use spandsp_start_dtmf at all. If the > > > phone is sending only audio, and is not capable of using any other method, > > > you will need to pull a pcap of this call, use wireshark to extract the > > > audio, and do some analysis of the audio to figure out why it is detecting > > > multiple digits. > > > > > > > > > On Jan 19, 2015, at 4:38 AM, Nagalenoj H. > wrote: > > > > > > Hi, > > > > > > When I use spandsp_start_dtmf along with play_and_get_digits, '#' is > > > getting detected many times, though it is pressed once. > > > > > > Other DTMFs are getting detected properly, whereas the '#' is getting > > > detected 2 or 3 times. > > > > > > I have checked the SIP phone's settings, I do not find anything wrong > > > there. > > > > > > What could be the reason? Is this because of some SIP phone setting? Here > > > is the freeswitch log, > > > > > > 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [1] > > > 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF 1:2000 > > > 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [1], duration = 191 ms > > > 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [4] > > > 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 > > > 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [4], duration = 153 ms > > > 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [7] > > > 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 > > > 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [7], duration = 204 ms > > > 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [4] > > > 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 > > > 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [4], duration = 127 ms > > > 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send signal > > > sofia/internal/sip:278 at 192.168.2.86:5060 [BREAK] > > > 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [7] > > > 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 > > > 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [7], duration = 191 ms > > > 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec > > > Activated L16 at 8000hz 1 channels 20ms > > > 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [#] > > > 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > > > EXECUTE sofia/internal/392 at 3427.vbiz.mundio.com flush_dtmf() > > > 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec > > > Activated L16 at 8000hz 1 channels 30ms > > > 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [#], duration = 38 ms > > > 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [#] > > > 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done playing > > > file file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav > > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec > > > Activated L16 at 8000hz 1 channels 30ms > > > 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [#], duration = 25 ms > > > 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > > > DETECTED: [#] > > > 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 > > > 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > > > DETECTED: [#], duration = 76 ms > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/105125f2/attachment.html From mike at jerris.com Tue Jan 20 16:57:53 2015 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jan 2015 08:57:53 -0500 Subject: [Freeswitch-users] Sound files from cache In-Reply-To: References: Message-ID: <6E5BF6EF-EB0E-4CBC-B29A-B453FA77F4C8@jerris.com> Also.. pretty much all modern operating systems automatically handle appropriate file caching into memory, so in most cases where you are reusing the same files over and over again, they will end up in memory without any tricks. > On Jan 20, 2015, at 8:49 AM, ?talo Rossi wrote: > > Are you having problem playing files? Why worry with this? > > On Tue, Jan 20, 2015 at 9:04 AM, John Nash > wrote: > I am using "playback" to play native sound file () > > My question is when it plays the file does it read file every time from the disk? or there is some way to "Load at the startup" and play directly from memory. > > Yes, it does. You can put your file in a ram disk. > > I also saw the "mod_http_cache" but not sure if I should use it (I am unable to understand in what scenario it will be logical to use it) > > It caches a remote file (http) in your local filesystem. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/600054e5/attachment.html From john.nash778 at gmail.com Tue Jan 20 17:06:16 2015 From: john.nash778 at gmail.com (John Nash) Date: Tue, 20 Jan 2015 19:36:16 +0530 Subject: [Freeswitch-users] Sound files from cache In-Reply-To: <6E5BF6EF-EB0E-4CBC-B29A-B453FA77F4C8@jerris.com> References: <6E5BF6EF-EB0E-4CBC-B29A-B453FA77F4C8@jerris.com> Message-ID: OK. Thank you Italo and Micheal. I think I will first test without RAM disk and see how it goes. Another related question is I looked at "say" application and I think it is quite useful as I need to play digits like "One hundred forty two" (by passing 142) but I planned to use "native" files (already encoded media files). Looks like "say" wont work with native files. I was studying "lua" to make script to convert number into words and then play native files. Is there any other easy way to do it? On Tue, Jan 20, 2015 at 7:27 PM, Michael Jerris wrote: > Also.. pretty much all modern operating systems automatically handle > appropriate file caching into memory, so in most cases where you are > reusing the same files over and over again, they will end up in memory > without any tricks. > > On Jan 20, 2015, at 8:49 AM, ?talo Rossi wrote: > > Are you having problem playing files? Why worry with this? > > On Tue, Jan 20, 2015 at 9:04 AM, John Nash wrote: > >> I am using "playback" to play native sound file (> application="playback" data="/tmp/hello"/>) >> >> My question is when it plays the file does it read file every time from >> the disk? or there is some way to "Load at the startup" and play directly >> from memory. >> > > Yes, it does. You can put your file in a ram disk. > > I also saw the "mod_http_cache" but not sure if I should use it (I am >> unable to understand in what scenario it will be logical to use it) >> > > It caches a remote file (http) in your local filesystem. > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/cedb2b9a/attachment-0001.html From hi-tecc at hotmail.com Tue Jan 20 18:54:27 2015 From: hi-tecc at hotmail.com (DP .) Date: Tue, 20 Jan 2015 10:54:27 -0500 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: References: , , Message-ID: OpenVZ sucks if your neighbors are noisy. You'll have a some better luck with a KVM or XEN isolated container. Check your your Steal Time (ST) in Top to see if you are on a busy node. Date: Sun, 18 Jan 2015 12:54:11 -0700 From: darren at aleph-com.net To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch on VPS I've had success with linode.com for low volume.Darren Wiebe D. 587-789-0634 T. 877-702-2900 C. 780-808-3320 E. darren at aleph-com.net w. www.aleph-com.net On Sun, Jan 18, 2015 at 12:49 PM, Moishe Grunstein wrote: I know some users had luck with https://www.digitalocean.com for low volume. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Zolt?n Szab? Sent: Sunday, January 18, 2015 2:42 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch on VPS Hi, Does anyone have experience running Freeswitch at OVH? I have a VPS Classic 1 but the audio was really bad there. It was faltered. (tested with multiple clients from multiple locations) This is an OpenVZ virtualization if this counts. I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was also a little bit more, and it was a vServer virtualization, it worked perfectly. If you run your FS on a VPS without any problem, can you please recommend me some providers who has servers in the EU? I don't expect high call volume, just like 5-10 concurrent calls on 1-2 SIP trunks which is basically nothing. Many thanks, Zoltan _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/fb5c1057/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/fb5c1057/attachment.jpg From victor.chukalovskiy at gmail.com Wed Jan 21 00:16:47 2015 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 20 Jan 2015 16:16:47 -0500 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? Message-ID: <54BEC5BF.3060300@gmail.com> Greetings, Just discovered that fail_on_single_reject does not work with "|" operator... Not sure if it was always the case or a recent "improvement" Is there something equivalent that can be used with sequential bridge, that is with | operator? The goal is that if I bridge to "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one of the rejection codes I expect over gw1 it should not attempt gw2 Thx! -Victor From ssinyagin at gmail.com Wed Jan 21 00:59:37 2015 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Tue, 20 Jan 2015 22:59:37 +0100 Subject: [Freeswitch-users] Freeswitch on VPS In-Reply-To: References: Message-ID: forgot to mention that it's a XEN VPS at softronics.ch On Mon, Jan 19, 2015 at 11:24 PM, Stanislav Sinyagin wrote: > I run my PBX on a VPS at softronics.ch , which is not exactly EU, but > it's in the middle of Europe. The support staff communicates in > English freely. I've been with them for many years, and my PBX is > hosted there since 2011, without problems. > > > > On Sun, Jan 18, 2015 at 8:42 PM, Zolt?n Szab? wrote: >> Hi, >> >> Does anyone have experience running Freeswitch at OVH? I have a VPS Classic >> 1 but the audio was really bad there. It was faltered. (tested with multiple >> clients from multiple locations) This is an OpenVZ virtualization if this >> counts. >> >> I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was also a >> little bit more, and it was a vServer virtualization, it worked perfectly. >> >> If you run your FS on a VPS without any problem, can you please recommend me >> some providers who has servers in the EU? I don't expect high call volume, >> just like 5-10 concurrent calls on 1-2 SIP trunks which is basically >> nothing. >> >> Many thanks, >> Zoltan >> >> _________________________________________________________________________ >> 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 From bdfoster at davri.com Wed Jan 21 01:58:13 2015 From: bdfoster at davri.com (Brian Foster) Date: Tue, 20 Jan 2015 17:58:13 -0500 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: Sorry to bring up a really old thread... Cal, have you implemented this? I've got a really basic implementation now, looking for a param called 'follow-me' in the directory user's configs, and if it exists it sends out a call about 5 seconds after the leg to the internal extension starts. No LUA scripts, all XML dialplan. Honestly, it's not that flexible but it works well enough for our purposes (until today, but I'm still researching). Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana On Thu, Apr 25, 2013 at 10:55 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Sorry to hi-jack, just wanted to say thank you to everyone getting > involved on this ticket, and OP for raising this. > > The requirements that OP specified are almost exactly the same as what we > are doing, and this item was next in my list to figure out. > > Once the thread is finished, we should definitely get it put into a > cookbook page on the wiki! > > Cal > > > On Wed, Apr 24, 2013 at 7:46 PM, Brian Foster wrote: > >> Does anyone have ideas on this issue? >> >> -BDF >> >> Thank you, >> >> Brian Foster >> Project Manager/Owner's Representative >> Davri Investments, Incorporated >> P: +1-317-787-2686 >> M: +1-317-600-9753 >> Indianapolis, Indiana >> >> >> On Wed, Apr 10, 2013 at 8:16 PM, Brian Foster wrote: >> >>> Also forgot to add, I'd have to use Loopback because I would have to >>> send the follow me number back through the dialplan in order to figure out >>> which gateway it should go out (we select a gateway based on a few >>> different variables, but there's no way to tell until you send it through >>> the dialplan). I know Loopback is kinda evil but I don't see another way of >>> doing it. But theoretically it makes things a little more modular and >>> keeps maintenance of the dialplan to a minimum. >>> >>> Thank you, >>> >>> Brian Foster >>> Project Manager/Owner's Representative >>> Davri Investments, Incorporated >>> P: +1-317-787-2686 >>> M: +1-317-600-9753 >>> Indianapolis, Indiana >>> >>> >>> On Wed, Apr 10, 2013 at 8:13 PM, Brian Foster >>> wrote: >>> >>>> MSC: Yea, I've heard of it but never really used it. I've taken a look >>>> at some documentation here: >>>> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Enterprise_originate >>>> >>>> Avi's Suggestion: >>>> >>>> {ignore_early_media=true,group_ >>>> confirm_file=ivr/ivr-accept_reject_voicemail.wav,group_ >>>> confirm_key=1}sofia/gateway/flowroute/12223334444:_:user/1000 >>>> >>>> Avi: We're close :) >>>> >>>> >>> data="${sofia_contact(${dialed_extension})},{group_confirm_key=exec,group_confirm_file=lua >>>> menu.lua}loopback/${follow_me_number}/xml/default"/> >>>> >>>> Now, I'm not sure if that will actually work (might not take much to >>>> get it to work) but theoretically it should. To get it clean, I'll probably >>>> have to do a script that takes a look at the user's variables set in the >>>> directory and figure out if the user even has a follow_me_number set. >>>> >>>> Notice "{group_confirm_key=exec,group_confirm_file=lua menu.lua}". >>>> That was taken from >>>> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm. >>>> Anyone know if that works? That would sure as hell make a dent in this. >>>> >>>> Another problem is that this dial-string will likely have to be set in >>>> the directory in order to keep things consistent. What's the best route to >>>> take? I'd have to use a script to set the dial string, not sure how I could >>>> call the script from there. >>>> >>>> >>>> >>>> Thank you, >>>> >>>> Brian Foster >>>> Project Manager/Owner's Representative >>>> Davri Investments, Incorporated >>>> P: +1-317-787-2686 >>>> M: +1-317-600-9753 >>>> Indianapolis, Indiana >>>> >>>> >>>> On Wed, Apr 10, 2013 at 7:36 PM, Brian Foster >>>> wrote: >>>> >>>>> LinuxMCE has done the Bluetooth tracking bit, albeit with extra code >>>>> and asterisk. >>>>> >>>>> >>>>> On Wednesday, April 10, 2013, Guillermo Ruiz Camauer wrote: >>>>> >>>>>> I don't want to hijack your thread, but since the subject matter is >>>>>> FollowMe, I would like to add that I am looking for something along the >>>>>> lines of: >>>>>> >>>>>> A FollowMe that somehow knows what room you are in within a building >>>>>> and rings the nearest extension with a special ringtone which is assigned >>>>>> to each user. >>>>>> The "somehow" could be through a Bluetooth dongle attached to a PC in >>>>>> the room that detects the User's cell phone and updates a DB that FS has >>>>>> access to, or an Mobile Phone App that triangulates on WiFi Access Points >>>>>> and updates a DB, etc. >>>>>> >>>>>> Has anyone heard of such a system? Experiences? >>>>>> >>>>>> Thank you, >>>>>> >>>>>> Guillermo Ruiz Camauer >>>>>> >>>>>> >>>>>> On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster >>>>>> wrote: >>>>>> >>>>>>> I've been given an assignment. It's a little rough, and honestly >>>>>>> I've been working on other projects and at the same time loosing my >>>>>>> freeswitch-fu. So, here it goes. >>>>>>> >>>>>>> Company owner wants to be able to implement a follow me function. >>>>>>> He's asking for the deskphones to begin ringing, then have cell phones ring >>>>>>> N seconds later WHILE the deskphones continue to ring. The function has to >>>>>>> be able to work using a couple different ways of dialing (we've got call >>>>>>> groups implemented, >>>>>>> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). >>>>>>> When the mobile phone is answered, we need to be able to get some feedback >>>>>>> from the callee to figure out if they're human. We'll use AVMD to kill the >>>>>>> call if it detects a voicemail beep. >>>>>>> >>>>>>> I've looked at several different examples on the wiki and mailing >>>>>>> list, and the only way I can figure out how to do it while keeping the >>>>>>> requirements in mind is to at some point resort to using Loopback >>>>>>> (something i didnt want to do). >>>>>>> >>>>>>> Requirements are: >>>>>>> 1. Use a custom IVR/menu/something to get a confirmation from the >>>>>>> callee that they are human (while also keeping it available >>>>>>> for customization he's wanting a way to blacklist numbers on that same >>>>>>> menu). So that rules out group_confirm_file, etc. >>>>>>> 2. Use AVMD to kill the call if we detect the call was picked up by >>>>>>> voicemail. >>>>>>> 3. The custom IVR/menu/something isn't used on the deskphones >>>>>>> 4. Deskphones need to continue to ring after the external number leg >>>>>>> is started. I can't timeout the call on the deskphone then call the cell >>>>>>> phone, or call the deskphone, time it out, then call the deskphone and cell >>>>>>> phone. >>>>>>> 5. It has to work on any type of calling method (so basically, if >>>>>>> the deskphone rings then eventually the cell phone needs to ring to if it's >>>>>>> assigned). >>>>>>> >>>>>>> Has anyone done something similar, and if so, how did you do it? >>>>>>> >>>>>>> Thank you, >>>>>>> >>>>>>> Brian Foster >>>>>>> Project Manager/Owner's Representative >>>>>>> Davri Investments, Incorporated >>>>>>> P: +1-317-787-2686 >>>>>>> M: +1-317-600-9753 >>>>>>> Indianapolis, Indiana >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Guillermo Ruiz Camauer >>>>>> >>>>> >>>>> >>>>> -- >>>>> Thank you, >>>>> >>>>> Brian Foster >>>>> Project Manager/Owner's Representative >>>>> Davri Investments, Incorporated >>>>> P: +1-317-787-2686 >>>>> M: +1-317-600-9753 >>>>> Indianapolis, Indiana >>>>> >>>>> >>>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.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 > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150120/3bcb90bd/attachment-0001.html From luis.daniel.lucio at gmail.com Wed Jan 21 06:32:15 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 20 Jan 2015 22:32:15 -0500 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: Guillermo, I dont think bluethood will work on your need. Reason is that Bt has 1 to 1 relationship. This means, you need as many BT radios as many people are in the room. Putting 1 BT radio in a PC will work with first hit, but second person wont be able to be detected. I think best option is to install a softphone in your smartphone and allow multiple registration. When dialing to your extension, it will ring on your extension and on your smartphone as well. 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 2013-04-10 17:49 GMT-04:00 Guillermo Ruiz Camauer : > I don't want to hijack your thread, but since the subject matter is > FollowMe, I would like to add that I am looking for something along the > lines of: > > A FollowMe that somehow knows what room you are in within a building and > rings the nearest extension with a special ringtone which is assigned to > each user. > The "somehow" could be through a Bluetooth dongle attached to a PC in the > room that detects the User's cell phone and updates a DB that FS has access > to, or an Mobile Phone App that triangulates on WiFi Access Points and > updates a DB, etc. > > Has anyone heard of such a system? Experiences? > > Thank you, > > Guillermo Ruiz Camauer > > > On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: >> >> I've been given an assignment. It's a little rough, and honestly I've been >> working on other projects and at the same time loosing my freeswitch-fu. So, >> here it goes. >> >> Company owner wants to be able to implement a follow me function. He's >> asking for the deskphones to begin ringing, then have cell phones ring N >> seconds later WHILE the deskphones continue to ring. The function has to be >> able to work using a couple different ways of dialing (we've got call groups >> implemented, >> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When the >> mobile phone is answered, we need to be able to get some feedback from the >> callee to figure out if they're human. We'll use AVMD to kill the call if it >> detects a voicemail beep. >> >> I've looked at several different examples on the wiki and mailing list, >> and the only way I can figure out how to do it while keeping the >> requirements in mind is to at some point resort to using Loopback (something >> i didnt want to do). >> >> Requirements are: >> 1. Use a custom IVR/menu/something to get a confirmation from the callee >> that they are human (while also keeping it available for customization he's >> wanting a way to blacklist numbers on that same menu). So that rules out >> group_confirm_file, etc. >> 2. Use AVMD to kill the call if we detect the call was picked up by >> voicemail. >> 3. The custom IVR/menu/something isn't used on the deskphones >> 4. Deskphones need to continue to ring after the external number leg is >> started. I can't timeout the call on the deskphone then call the cell phone, >> or call the deskphone, time it out, then call the deskphone and cell phone. >> 5. It has to work on any type of calling method (so basically, if the >> deskphone rings then eventually the cell phone needs to ring to if it's >> assigned). >> >> Has anyone done something similar, and if so, how did you do it? >> >> Thank you, >> >> Brian Foster >> Project Manager/Owner's Representative >> Davri Investments, Incorporated >> P: +1-317-787-2686 >> M: +1-317-600-9753 >> Indianapolis, Indiana >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.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 >> > > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.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 > From grcamauer at gmail.com Wed Jan 21 07:22:25 2015 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 20 Jan 2015 23:22:25 -0500 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: Good observation regarding Bluetooth. I was looking for a solution that did not require a cell phone (just a Bluetooth fob). Any another indoor tracking technology come to mind? Thanks, Guillermo Sent from my iPhone > On 20/1/2015, at 22:32, Luis Daniel Lucio Quiroz wrote: > > Guillermo, > > I dont think bluethood will work on your need. Reason is that Bt has 1 > to 1 relationship. This means, you need as many BT radios as many > people are in the room. Putting 1 BT radio in a PC will work with > first hit, but second person wont be able to be detected. > > I think best option is to install a softphone in your smartphone and > allow multiple registration. When dialing to your extension, it will > ring on your extension and on your smartphone as well. > > > 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 > > > 2013-04-10 17:49 GMT-04:00 Guillermo Ruiz Camauer : >> I don't want to hijack your thread, but since the subject matter is >> FollowMe, I would like to add that I am looking for something along the >> lines of: >> >> A FollowMe that somehow knows what room you are in within a building and >> rings the nearest extension with a special ringtone which is assigned to >> each user. >> The "somehow" could be through a Bluetooth dongle attached to a PC in the >> room that detects the User's cell phone and updates a DB that FS has access >> to, or an Mobile Phone App that triangulates on WiFi Access Points and >> updates a DB, etc. >> >> Has anyone heard of such a system? Experiences? >> >> Thank you, >> >> Guillermo Ruiz Camauer >> >> >>> On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: >>> >>> I've been given an assignment. It's a little rough, and honestly I've been >>> working on other projects and at the same time loosing my freeswitch-fu. So, >>> here it goes. >>> >>> Company owner wants to be able to implement a follow me function. He's >>> asking for the deskphones to begin ringing, then have cell phones ring N >>> seconds later WHILE the deskphones continue to ring. The function has to be >>> able to work using a couple different ways of dialing (we've got call groups >>> implemented, >>> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). When the >>> mobile phone is answered, we need to be able to get some feedback from the >>> callee to figure out if they're human. We'll use AVMD to kill the call if it >>> detects a voicemail beep. >>> >>> I've looked at several different examples on the wiki and mailing list, >>> and the only way I can figure out how to do it while keeping the >>> requirements in mind is to at some point resort to using Loopback (something >>> i didnt want to do). >>> >>> Requirements are: >>> 1. Use a custom IVR/menu/something to get a confirmation from the callee >>> that they are human (while also keeping it available for customization he's >>> wanting a way to blacklist numbers on that same menu). So that rules out >>> group_confirm_file, etc. >>> 2. Use AVMD to kill the call if we detect the call was picked up by >>> voicemail. >>> 3. The custom IVR/menu/something isn't used on the deskphones >>> 4. Deskphones need to continue to ring after the external number leg is >>> started. I can't timeout the call on the deskphone then call the cell phone, >>> or call the deskphone, time it out, then call the deskphone and cell phone. >>> 5. It has to work on any type of calling method (so basically, if the >>> deskphone rings then eventually the cell phone needs to ring to if it's >>> assigned). >>> >>> Has anyone done something similar, and if so, how did you do it? >>> >>> Thank you, >>> >>> Brian Foster >>> Project Manager/Owner's Representative >>> Davri Investments, Incorporated >>> P: +1-317-787-2686 >>> M: +1-317-600-9753 >>> Indianapolis, Indiana >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.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 >> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.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 From max at nysolutions.com Wed Jan 21 08:27:47 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 21 Jan 2015 05:27:47 +0000 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: You can control presence using the mac address of the Bluetooth device, see http://nerdvittles.com/?p=803 and http://nerdvittles.com/?p=185 Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis Daniel Lucio Quiroz Sent: Tuesday, January 20, 2015 10:32 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Follow me implementation Guillermo, I dont think bluethood will work on your need. Reason is that Bt has 1 to 1 relationship. This means, you need as many BT radios as many people are in the room. Putting 1 BT radio in a PC will work with first hit, but second person wont be able to be detected. I think best option is to install a softphone in your smartphone and allow multiple registration. When dialing to your extension, it will ring on your extension and on your smartphone as well. 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 2013-04-10 17:49 GMT-04:00 Guillermo Ruiz Camauer : > I don't want to hijack your thread, but since the subject matter is > FollowMe, I would like to add that I am looking for something along > the lines of: > > A FollowMe that somehow knows what room you are in within a building > and rings the nearest extension with a special ringtone which is > assigned to each user. > The "somehow" could be through a Bluetooth dongle attached to a PC in > the room that detects the User's cell phone and updates a DB that FS > has access to, or an Mobile Phone App that triangulates on WiFi Access > Points and updates a DB, etc. > > Has anyone heard of such a system? Experiences? > > Thank you, > > Guillermo Ruiz Camauer > > > On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster wrote: >> >> I've been given an assignment. It's a little rough, and honestly I've >> been working on other projects and at the same time loosing my >> freeswitch-fu. So, here it goes. >> >> Company owner wants to be able to implement a follow me function. >> He's asking for the deskphones to begin ringing, then have cell >> phones ring N seconds later WHILE the deskphones continue to ring. >> The function has to be able to work using a couple different ways of >> dialing (we've got call groups implemented, >> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). >> When the mobile phone is answered, we need to be able to get some >> feedback from the callee to figure out if they're human. We'll use >> AVMD to kill the call if it detects a voicemail beep. >> >> I've looked at several different examples on the wiki and mailing >> list, and the only way I can figure out how to do it while keeping >> the requirements in mind is to at some point resort to using Loopback >> (something i didnt want to do). >> >> Requirements are: >> 1. Use a custom IVR/menu/something to get a confirmation from the >> callee that they are human (while also keeping it available for >> customization he's wanting a way to blacklist numbers on that same >> menu). So that rules out group_confirm_file, etc. >> 2. Use AVMD to kill the call if we detect the call was picked up by >> voicemail. >> 3. The custom IVR/menu/something isn't used on the deskphones 4. >> Deskphones need to continue to ring after the external number leg is >> started. I can't timeout the call on the deskphone then call the cell >> phone, or call the deskphone, time it out, then call the deskphone and cell phone. >> 5. It has to work on any type of calling method (so basically, if the >> deskphone rings then eventually the cell phone needs to ring to if >> it's assigned). >> >> Has anyone done something similar, and if so, how did you do it? >> >> Thank you, >> >> Brian Foster >> Project Manager/Owner's Representative Davri Investments, >> Incorporated >> P: +1-317-787-2686 >> M: +1-317-600-9753 >> Indianapolis, Indiana >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.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-us >> ers >> http://www.freeswitch.org >> > > > > -- > Guillermo Ruiz Camauer > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.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-use > rs > 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 From achinthau at gmail.com Wed Jan 21 11:20:57 2015 From: achinthau at gmail.com (Achintha) Date: Wed, 21 Jan 2015 13:50:57 +0530 Subject: [Freeswitch-users] media bypass on freeswitch Message-ID: I configured freeswitch (FreeSWITCH Version 1.5.15b+git~20141117T202539Z~424df19083~64bit) on Ubuntu 12.04. It works properly. I?m using xml_curl module to generate the dialplan. Then I set the media bypass parameter on freeswitch dialplan. But the freeswitch crashed when following dialplan was given.
References: Message-ID: Please file a Jira. A segfault is always a bug. /Peter 2015-01-21 9:20 GMT+01:00 Achintha : > > I configured freeswitch (FreeSWITCH Version > 1.5.15b+git~20141117T202539Z~424df19083~64bit) on Ubuntu 12.04. It works > properly. I?m using xml_curl module to generate the dialplan. Then I set > the media bypass parameter on freeswitch dialplan. But the freeswitch > crashed when following dialplan was given. > > > > >
> > > > > > > > > > > > > > data="FromGuUserId=14111215533245504" /> > > /> > > data="FromUser=sanasains2023_sanasa_lk" /> > > > > > > data="nolocal:ToUser=sanasains2022_sanasa_lk" /> > > > > > > > > > > > > > > > > > > data="[leg_timeout=60,origination_caller_id_name=sanasains2023_sanasa_lk,origination_caller_id_number=2023]user/ > 2022 at 192.168.51.X <2022 at 192.168.51.86>X,pickup/3003" /> > > > > > > > > > >
> > > > > In here I ?m using attendant transfer feature. > > Here I have attached the print screen of freeswitch console just after > it crashed. > > Kindly advise me to solve this problem. > > > -- > Thanking You.. > Achintha > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150121/3c6f4a26/attachment.html From steveayre at gmail.com Wed Jan 21 13:18:57 2015 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 21 Jan 2015 10:18:57 +0000 Subject: [Freeswitch-users] media bypass on freeswitch In-Reply-To: References: Message-ID: Also make sure you can reproduce it on the latest version - 20141117T202539Z is 2 months out of date and it's possible your bug has already been found and fixed. On 21 January 2015 at 10:06, Peter Olsson wrote: > Please file a Jira. A segfault is always a bug. > > /Peter > > 2015-01-21 9:20 GMT+01:00 Achintha : > >> >> I configured freeswitch (FreeSWITCH Version >> 1.5.15b+git~20141117T202539Z~424df19083~64bit) on Ubuntu 12.04. It works >> properly. I?m using xml_curl module to generate the dialplan. Then I set >> the media bypass parameter on freeswitch dialplan. But the freeswitch >> crashed when following dialplan was given. >> >> >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> > data="FromGuUserId=14111215533245504" /> >> >> > data="ToGuUserId=14111215495317100" /> >> >> > data="FromUser=sanasains2023_sanasa_lk" /> >> >> >> >> >> >> > data="nolocal:ToUser=sanasains2022_sanasa_lk" /> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > data="[leg_timeout=60,origination_caller_id_name=sanasains2023_sanasa_lk,origination_caller_id_number=2023]user/ >> 2022 at 192.168.51.X <2022 at 192.168.51.86>X,pickup/3003" /> >> >> >> >> >> >> >> >> >> >>
>> >> > >> >> >> In here I ?m using attendant transfer feature. >> >> Here I have attached the print screen of freeswitch console just after >> it crashed. >> >> Kindly advise me to solve this problem. >> >> >> -- >> Thanking You.. >> Achintha >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150121/55eaf23b/attachment-0001.html From avi at avimarcus.net Wed Jan 21 13:52:35 2015 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 21 Jan 2015 10:52:35 +0000 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? In-Reply-To: <54BEC5BF.3060300@gmail.com> References: <54BEC5BF.3060300@gmail.com> Message-ID: <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> I've never been clear on how this works. However, the most typical case - you get a USER_BUSY so don't try any more carriers - I've found to my surprise that other carriers sometimes are able to complete the call... -Avi On Tue, Jan 20, 2015 at 11:16 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Greetings, > > Just discovered that fail_on_single_reject does not work with "|" > operator... > > Not sure if it was always the case or a recent "improvement" > > Is there something equivalent that can be used with sequential bridge, > that is with | operator? > The goal is that if I bridge to > "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one of the > rejection codes I expect over gw1 it should not attempt gw2 > > Thx! > -Victor > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150121/815e5231/attachment.html From dp.siddharth at eng.knowlarity.com Wed Jan 21 14:58:05 2015 From: dp.siddharth at eng.knowlarity.com (DP Siddharth) Date: Wed, 21 Jan 2015 17:28:05 +0530 Subject: [Freeswitch-users] Getting Redirecting number in Events (ISDN) Message-ID: Hi, I am using ISDN (PRI) for handling calls. In a use case we need Redirecting number as part of freeswitch events. I verified various events but we get only caller/called. I was looking into the code & found for SS7 this is available but not for ISDN, just need confirmation if any way possible to get this. I am using 1.2.24 (1.2.stable) freeswitch for testing. following is ISDN SETUP message for reference: Prot Disc:Q.931/I.451 (0x08) Call Ref:242E (Originating side) Type:SETUP (0x5) Sending complete: Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3) Channel Id:No:13 Type:B-chans(3) Preferred/Implicit Calling Party Number:8860128000(l:10) plan:isdn(1) type:national(2)scr:network, provided(3) pres:allowed(0) Called Party Number:01725218700(l:11) plan:isdn(1) type:subscriber(4) Redirecting Number:53703417(l:8) plan:isdn(1) type:national(2)scr:user, not screened(0) pres:restricted(1)reason:Call forwarding unconditional(15) Redirecting Number:53703417(l:8) plan:isdn(1) type:national(2)scr:user, not screened(0) pres:restricted(1)reason:Call forwarding unconditional(15) [ 08 02 24 2e 05 a1 04 03 80 90 a3 18 03 a1 83 8d 6c 0c 21 83 38 38 36 30 31 32 38 30 30 30 70 0c c1 30 31 37 32 35 32 31 38 37 30 30 74 0d 21 20 8f 39 36 35 33 37 30 33 34 31 37 74 0d 21 20 8f 39 36 35 33 37 30 33 34 31 37 ] -- Thanks & Regards, D P Siddharth Director (Platform) Knowlarity Communications Ph: +919999115231 dp.siddharth at eng.knowlarity.com *"Come together to build a lasting world-class cloud telephony company that helps businesses grow"* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150121/ae9ec739/attachment.html From vipkilla at gmail.com Wed Jan 21 16:38:20 2015 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 21 Jan 2015 08:38:20 -0500 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? In-Reply-To: <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> References: <54BEC5BF.3060300@gmail.com> <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> Message-ID: try setting continue_on_fail=true On Wed, Jan 21, 2015 at 5:52 AM, Avi Marcus wrote: > I've never been clear on how this works. > > However, the most typical case - you get a USER_BUSY so don't try any more > carriers - I've found to my surprise that other carriers sometimes are able > to complete the call... > > -Avi > > On Tue, Jan 20, 2015 at 11:16 PM, Victor Chukalovskiy < > victor.chukalovskiy at gmail.com> wrote: > >> Greetings, >> >> Just discovered that fail_on_single_reject does not work with "|" >> operator... >> >> Not sure if it was always the case or a recent "improvement" >> >> Is there something equivalent that can be used with sequential bridge, >> that is with | operator? >> The goal is that if I bridge to >> "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one of the >> rejection codes I expect over gw1 it should not attempt gw2 >> >> Thx! >> -Victor >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150121/ef2b2c4e/attachment.html From victor.chukalovskiy at gmail.com Wed Jan 21 18:34:04 2015 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 21 Jan 2015 10:34:04 -0500 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? In-Reply-To: References: <54BEC5BF.3060300@gmail.com> <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> Message-ID: <54BFC6EC.1080306@gmail.com> @Avi - Thanks for the feedback. Yes, in typical use scenarios you let it fail-over using defaults. However, I'm working on something quite different hence looking for more controls. If anyone could shed more light, would be great. My understanding is that today there is no channel variable in FS that lets control reject causes to stop bridge iterations over | separated endpoints. Am I right or am I wrong? @Vik - I'm using continue_on_fail=true, however believe it's not relevant for my question. It controls what happens after bridge is done. I'm looking for control between multiple endpoints within the same bridge. Thanks all! On 15-01-21 08:38 AM, Vik Killa wrote: > try setting continue_on_fail=true > > On Wed, Jan 21, 2015 at 5:52 AM, Avi Marcus > wrote: > > I've never been clear on how this works. > > However, the most typical case - you get a USER_BUSY so don't try > any more carriers - I've found to my surprise that other carriers > sometimes are able to complete the call... > > -Avi > > On Tue, Jan 20, 2015 at 11:16 PM, Victor Chukalovskiy > > wrote: > > Greetings, > > Just discovered that fail_on_single_reject does not work with "|" > operator... > > Not sure if it was always the case or a recent "improvement" > > Is there something equivalent that can be used with sequential > bridge, > that is with | operator? > The goal is that if I bridge to > "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one of the > rejection codes I expect over gw1 it should not attempt gw2 > > Thx! > -Victor > > _________________________________________________________________________ > 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 > > > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150121/d05fb0e5/attachment-0001.html From vipkilla at gmail.com Wed Jan 21 18:52:59 2015 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 21 Jan 2015 10:52:59 -0500 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? In-Reply-To: <54BFC6EC.1080306@gmail.com> References: <54BEC5BF.3060300@gmail.com> <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> <54BFC6EC.1080306@gmail.com> Message-ID: continue_on_fail=true is relevant if a bridge fails. You almost always need to use it with enterprise originate On Wed, Jan 21, 2015 at 10:34 AM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > @Avi - Thanks for the feedback. Yes, in typical use scenarios you let it > fail-over using defaults. > However, I'm working on something quite different hence looking for more > controls. > > If anyone could shed more light, would be great. > My understanding is that today there is no channel variable in FS that > lets control reject causes to stop bridge iterations over | separated > endpoints. > Am I right or am I wrong? > > @Vik - I'm using continue_on_fail=true, however believe it's not relevant > for my question. > It controls what happens after bridge is done. I'm looking for control > between multiple endpoints within the same bridge. > > Thanks all! > > > > On 15-01-21 08:38 AM, Vik Killa wrote: > > try setting continue_on_fail=true > > On Wed, Jan 21, 2015 at 5:52 AM, Avi Marcus wrote: > >> I've never been clear on how this works. >> >> However, the most typical case - you get a USER_BUSY so don't try any >> more carriers - I've found to my surprise that other carriers sometimes are >> able to complete the call... >> >> -Avi >> >> On Tue, Jan 20, 2015 at 11:16 PM, Victor Chukalovskiy < >> victor.chukalovskiy at gmail.com> wrote: >> >>> Greetings, >>> >>> Just discovered that fail_on_single_reject does not work with "|" >>> operator... >>> >>> Not sure if it was always the case or a recent "improvement" >>> >>> Is there something equivalent that can be used with sequential bridge, >>> that is with | operator? >>> The goal is that if I bridge to >>> "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one of the >>> rejection codes I expect over gw1 it should not attempt gw2 >>> >>> Thx! >>> -Victor >>> >>> _________________________________________________________________________ >>> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150121/c72f1dc3/attachment.html From alipey at gmail.com Wed Jan 21 19:14:18 2015 From: alipey at gmail.com (Ali Pey) Date: Wed, 21 Jan 2015 11:14:18 -0500 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? In-Reply-To: References: <54BEC5BF.3060300@gmail.com> <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> <54BFC6EC.1080306@gmail.com> Message-ID: Victor, I've had the same issue for a long time. I haven't been able to find a way to stop the bridge from continuing dialling destinations if the call was sent to voicemail for instance. I'd love to find a solution for this. Please let me know if you find a way to do this. I have a hack to do this but screws up many other things. Thanks, Ali Pey On Wed, Jan 21, 2015 at 10:52 AM, Vik Killa wrote: > continue_on_fail=true is relevant if a bridge fails. > You almost always need to use it with enterprise originate > > On Wed, Jan 21, 2015 at 10:34 AM, Victor Chukalovskiy < > victor.chukalovskiy at gmail.com> wrote: > >> @Avi - Thanks for the feedback. Yes, in typical use scenarios you let >> it fail-over using defaults. >> However, I'm working on something quite different hence looking for more >> controls. >> >> If anyone could shed more light, would be great. >> My understanding is that today there is no channel variable in FS that >> lets control reject causes to stop bridge iterations over | separated >> endpoints. >> Am I right or am I wrong? >> >> @Vik - I'm using continue_on_fail=true, however believe it's not relevant >> for my question. >> It controls what happens after bridge is done. I'm looking for control >> between multiple endpoints within the same bridge. >> >> Thanks all! >> >> >> >> On 15-01-21 08:38 AM, Vik Killa wrote: >> >> try setting continue_on_fail=true >> >> On Wed, Jan 21, 2015 at 5:52 AM, Avi Marcus wrote: >> >>> I've never been clear on how this works. >>> >>> However, the most typical case - you get a USER_BUSY so don't try any >>> more carriers - I've found to my surprise that other carriers sometimes are >>> able to complete the call... >>> >>> -Avi >>> >>> On Tue, Jan 20, 2015 at 11:16 PM, Victor Chukalovskiy < >>> victor.chukalovskiy at gmail.com> wrote: >>> >>>> Greetings, >>>> >>>> Just discovered that fail_on_single_reject does not work with "|" >>>> operator... >>>> >>>> Not sure if it was always the case or a recent "improvement" >>>> >>>> Is there something equivalent that can be used with sequential bridge, >>>> that is with | operator? >>>> The goal is that if I bridge to >>>> "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one of the >>>> rejection codes I expect over gw1 it should not attempt gw2 >>>> >>>> Thx! >>>> -Victor >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150121/7bd74421/attachment-0001.html From victor.chukalovskiy at gmail.com Wed Jan 21 19:16:25 2015 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 21 Jan 2015 11:16:25 -0500 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? In-Reply-To: References: <54BEC5BF.3060300@gmail.com> <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> <54BFC6EC.1080306@gmail.com> Message-ID: <54BFD0D9.3010900@gmail.com> I'm not doing enterprise originate here On 15-01-21 10:52 AM, Vik Killa wrote: > continue_on_fail=true is relevant if a bridge fails. > You almost always need to use it with enterprise originate > > On Wed, Jan 21, 2015 at 10:34 AM, Victor Chukalovskiy > > > wrote: > > @Avi - Thanks for the feedback. Yes, in typical use scenarios you > let it fail-over using defaults. > However, I'm working on something quite different hence looking > for more controls. > > If anyone could shed more light, would be great. > My understanding is that today there is no channel variable in FS > that lets control reject causes to stop bridge iterations over | > separated endpoints. > Am I right or am I wrong? > > @Vik - I'm using continue_on_fail=true, however believe it's not > relevant for my question. > It controls what happens after bridge is done. I'm looking for > control between multiple endpoints within the same bridge. > > Thanks all! > > > > On 15-01-21 08:38 AM, Vik Killa wrote: >> try setting continue_on_fail=true >> >> On Wed, Jan 21, 2015 at 5:52 AM, Avi Marcus > > wrote: >> >> I've never been clear on how this works. >> >> However, the most typical case - you get a USER_BUSY so don't >> try any more carriers - I've found to my surprise that other >> carriers sometimes are able to complete the call... >> >> -Avi >> >> On Tue, Jan 20, 2015 at 11:16 PM, Victor Chukalovskiy >> > > wrote: >> >> Greetings, >> >> Just discovered that fail_on_single_reject does not work >> with "|" >> operator... >> >> Not sure if it was always the case or a recent "improvement" >> >> Is there something equivalent that can be used with >> sequential bridge, >> that is with | operator? >> The goal is that if I bridge to >> "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one >> of the >> rejection codes I expect over gw1 it should not attempt gw2 >> >> Thx! >> -Victor >> >> _________________________________________________________________________ >> 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 >> >> >> >> >> _________________________________________________________________________ >> 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 > > > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150121/b12c719c/attachment.html From steveayre at gmail.com Thu Jan 22 03:19:14 2015 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 22 Jan 2015 00:19:14 +0000 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? In-Reply-To: References: <54BEC5BF.3060300@gmail.com> <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> Message-ID: That is evaluated at the end of the bridge app to decide whether to return to dialplan or hangup, not inbetween sequential (|) destinations. On 21 January 2015 at 13:38, Vik Killa wrote: > try setting continue_on_fail=true > > On Wed, Jan 21, 2015 at 5:52 AM, Avi Marcus wrote: > >> I've never been clear on how this works. >> >> However, the most typical case - you get a USER_BUSY so don't try any >> more carriers - I've found to my surprise that other carriers sometimes are >> able to complete the call... >> >> -Avi >> >> On Tue, Jan 20, 2015 at 11:16 PM, Victor Chukalovskiy < >> victor.chukalovskiy at gmail.com> wrote: >> >>> Greetings, >>> >>> Just discovered that fail_on_single_reject does not work with "|" >>> operator... >>> >>> Not sure if it was always the case or a recent "improvement" >>> >>> Is there something equivalent that can be used with sequential bridge, >>> that is with | operator? >>> The goal is that if I bridge to >>> "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one of the >>> rejection codes I expect over gw1 it should not attempt gw2 >>> >>> Thx! >>> -Victor >>> >>> _________________________________________________________________________ >>> 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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/1a206eab/attachment-0001.html From steveayre at gmail.com Thu Jan 22 03:26:05 2015 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 22 Jan 2015 00:26:05 +0000 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? In-Reply-To: <54BFC6EC.1080306@gmail.com> References: <54BEC5BF.3060300@gmail.com> <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> <54BFC6EC.1080306@gmail.com> Message-ID: > > My understanding is that today there is no channel variable in FS that > lets control reject causes to stop bridge iterations over | separated > endpoints. Am I right or am I wrong? fail_on_single_reject should be working for exactly that purpose. For example I use the following: This passes routes via multiple carriers. If the gateway is down/non-existent it will try the next sequential destination(s), but any other failure cause causes the bridge to end without trying the next (fail_on_single_reject). If all the gateways are down it returns to dialplan (continue_on_fail) otherwise it hangs up with the reject cause from the bridge. On 21 January 2015 at 15:34, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > @Avi - Thanks for the feedback. Yes, in typical use scenarios you let it > fail-over using defaults. > However, I'm working on something quite different hence looking for more > controls. > > If anyone could shed more light, would be great. > My understanding is that today there is no channel variable in FS that > lets control reject causes to stop bridge iterations over | separated > endpoints. > Am I right or am I wrong? > > @Vik - I'm using continue_on_fail=true, however believe it's not relevant > for my question. > It controls what happens after bridge is done. I'm looking for control > between multiple endpoints within the same bridge. > > Thanks all! > > > > On 15-01-21 08:38 AM, Vik Killa wrote: > > try setting continue_on_fail=true > > On Wed, Jan 21, 2015 at 5:52 AM, Avi Marcus wrote: > >> I've never been clear on how this works. >> >> However, the most typical case - you get a USER_BUSY so don't try any >> more carriers - I've found to my surprise that other carriers sometimes are >> able to complete the call... >> >> -Avi >> >> On Tue, Jan 20, 2015 at 11:16 PM, Victor Chukalovskiy < >> victor.chukalovskiy at gmail.com> wrote: >> >>> Greetings, >>> >>> Just discovered that fail_on_single_reject does not work with "|" >>> operator... >>> >>> Not sure if it was always the case or a recent "improvement" >>> >>> Is there something equivalent that can be used with sequential bridge, >>> that is with | operator? >>> The goal is that if I bridge to >>> "sofia/gw1/5555555555|sofia/gw2/5555555555 and I get one of the >>> rejection codes I expect over gw1 it should not attempt gw2 >>> >>> Thx! >>> -Victor >>> >>> _________________________________________________________________________ >>> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/ec1f95b4/attachment.html From luis.daniel.lucio at gmail.com Thu Jan 22 05:32:58 2015 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 21 Jan 2015 21:32:58 -0500 Subject: [Freeswitch-users] Install FusionPBX in minutes on Centos 6 & 7 Message-ID: Hello everybody Just to share you this: https://okay.com.mx/en/entrepreneurs/install-fusionpbx-in-a-moment.html With this alternate repo you will be able to install FusionPBX in minutes, just with a yum command. Please read the link for all details. 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 From regis.freeswitch.org at tornad.net Thu Jan 22 11:30:24 2015 From: regis.freeswitch.org at tornad.net (Regis M) Date: Thu, 22 Jan 2015 09:30:24 +0100 Subject: [Freeswitch-users] Register more than 100 gateways on a host Message-ID: Hi, I had a project where I could have to register between 100 and 500 gateways/trunk on a host. Does anyone have already done something similar ? Have you some problems with this configuration or recommandations ? For me, It"s equivalent to 250 sip phones register on me... it's just that : sofia status will give me some lines :) Regards R?gis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/d0ff5211/attachment.html From nagalenoj at gmail.com Thu Jan 22 14:23:53 2015 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Thu, 22 Jan 2015 16:53:53 +0530 Subject: [Freeswitch-users] spandsp_start_dtmf detects # key many times In-Reply-To: <82880EB6-F9B2-4FAA-B709-A403612A869B@jerris.com> References: <33221776-F8EB-4329-BC1A-5413478DBD16@jerris.com> <8404.1421737661@ccs.covici.com> <82880EB6-F9B2-4FAA-B709-A403612A869B@jerris.com> Message-ID: Yes.. Thanks. It works. On Tue, Jan 20, 2015 at 7:26 PM, Michael Jerris wrote: > Which phone is it? look for a checkbox about force 2833 out of band dtmf. > > > On Jan 20, 2015, at 2:54 AM, Nagalenoj H. wrote: > > My phone does not have an option to select RFC2833. > > On Tue, Jan 20, 2015 at 12:37 PM, wrote: > >> Why not use rfc2833 instead? >> >> Nagalenoj H. wrote: >> >> > When I changed the Aastra phone settings to use SIP INFO, am not facing >> > this issue. But, when I change it to use RTP for DTMF, then '#' is >> getting >> > detected more times. >> > >> > I did a pcap and listened to the audio, I could hear only one tone(beep) >> > for each key pressed. Is there anything other configuration to get it >> > working in RTP? >> > >> > On Mon, Jan 19, 2015 at 10:40 PM, Michael Jerris >> wrote: >> > >> > > If you are using a sip phone, you are MUCH better off using non audio >> > > based dtmf like rfc-2833. If you are already using this, its >> possible that >> > > what is happening is the phone is sending the dtmf as audio and 2833, >> at >> > > which point, you should not have to use spandsp_start_dtmf at all. >> If the >> > > phone is sending only audio, and is not capable of using any other >> method, >> > > you will need to pull a pcap of this call, use wireshark to extract >> the >> > > audio, and do some analysis of the audio to figure out why it is >> detecting >> > > multiple digits. >> > > >> > > >> > > On Jan 19, 2015, at 4:38 AM, Nagalenoj H. >> wrote: >> > > >> > > Hi, >> > > >> > > When I use spandsp_start_dtmf along with play_and_get_digits, '#' is >> > > getting detected many times, though it is pressed once. >> > > >> > > Other DTMFs are getting detected properly, whereas the '#' is getting >> > > detected 2 or 3 times. >> > > >> > > I have checked the SIP phone's settings, I do not find anything wrong >> > > there. >> > > >> > > What could be the reason? Is this because of some SIP phone setting? >> Here >> > > is the freeswitch log, >> > > >> > > 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [1] >> > > 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF >> 1:2000 >> > > 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [1], duration = 191 ms >> > > 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [4] >> > > 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF >> 4:2000 >> > > 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [4], duration = 153 ms >> > > 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [7] >> > > 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF >> 7:2000 >> > > 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [7], duration = 204 ms >> > > 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [4] >> > > 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF >> 4:2000 >> > > 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [4], duration = 127 ms >> > > 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send >> signal >> > > sofia/internal/sip:278 at 192.168.2.86:5060 [BREAK] >> > > 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [7] >> > > 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF >> 7:2000 >> > > 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [7], duration = 191 ms >> > > 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec >> > > Activated L16 at 8000hz 1 channels 20ms >> > > 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [#] >> > > 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF >> #:2000 >> > > EXECUTE sofia/internal/392 at 3427.vbiz.mundio.com flush_dtmf() >> > > 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec >> > > Activated L16 at 8000hz 1 channels 30ms >> > > 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [#], duration = 38 ms >> > > 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [#] >> > > 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF >> #:2000 >> > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done >> playing >> > > file >> file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav >> > > 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec >> > > Activated L16 at 8000hz 1 channels 30ms >> > > 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [#], duration = 25 ms >> > > 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> > > DETECTED: [#] >> > > 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF >> #:2000 >> > > 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> > > DETECTED: [#], duration = 76 ms >> > > >> > > >> > > >> > > _________________________________________________________________________ > 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 > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/4144f088/attachment-0001.html From GeorgePhelps at gfphelps.com Thu Jan 22 15:28:03 2015 From: GeorgePhelps at gfphelps.com (George F. Phelps) Date: Thu, 22 Jan 2015 07:28:03 -0500 Subject: [Freeswitch-users] How to Bridge To Local Extensions? In-Reply-To: <32C551E0-A264-4F0C-9BB6-D4BB5E8D8271@jerris.com> References: <11b701d02de2$a4d8e100$ee8aa300$@gfphelps.com> <11d801d02dfe$9cbd5020$d637f060$@gfphelps.com> <120701d02e03$2c900540$85b00fc0$@gfphelps.com> <121e01d02e09$b72011e0$256035a0$@gfphelps.com> <124f01d02e18$78d1c2d0$6a754870$@gfphelps.com> <128601d02e57$6b73fba0$425bf2e0$@gfphelps.com> <12ad01d02e60$28da2ef0$7a8e8cd0$@gfphelps.com> <13c701d02ebd$09f54530$1ddfcf90$@gfphelps.com> <145d01d02f2f$5e7293c0$1b57bb40$@gfphelps.com> <14 7601d02f32$37a9f5a0$ a 6 fde0e0$@gfphelps.com> <02bb01d02f67$b21fcd70$165f6850$@com> <153801d02f82$4695a2d0$d3c0e870$@gfphelps.com> <007901d02ff5$67075090$3515f1b0$@gfphelps.com> <041401d030e5$ea3b6730$beb23590$@gfphelps.com> <04ce01d0311e$272fac40$758f04c0$@gfphelps.com> <053301d03189$dc9b9a10$95d2ce30$@gfphelps.com> <08b001d03433$af9d6a50$0ed83ef0$@gfphelps.com> <32C551E0-A264-4F0C-9BB6-D4BB5E8D8271@jerris.com> Message-ID: <03e201d0363e$de9d7e00$9bd87a00$@gfphelps.com> A Jira has been created: https://freeswitch.org/jira/browse/FS-7186 Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, January 19, 2015 5:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Please submit patches for review by creating a jira and a pull request in stash. On Jan 19, 2015, at 5:02 PM, George F. Phelps wrote: The following set of (test) code changes seems to resolve the issue. Could someone familiar with the FreeSWITCH code base please review the changes and provide their feedback? diff switch_ivr_originate.c switch_ivr_originate.c.ORIG 1371a1372 > handle->done = 0; 1380,1382d1380 < < return NULL; < 1559d1556 < handles[i].done = 0; 1633a1631 > 1660,1667c1658 < if (hp == &handles[i]) { < continue; < } < < if (handles[i].bleg) { < switch_channel_t *channel = switch_core_session_get_channel(handles[i].bleg); < < switch_channel_set_variable(channel, "group_dial_status", "loser"); --- > if (channel) { 1669d1659 < switch_core_session_rwunlock(handles[i].bleg); 1671a1662,1664 > if (hp == &handles[i]) { > continue; > } Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Friday, January 16, 2015 7:42 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Michael Collins, Any feedback on the debug log that I uploaded yesterday? Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [ mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Thursday, January 15, 2015 6:51 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? New failed call log at: http://pastebin.com/cErTyGht Scenario? Two registered extensions, 1000 and 1001. Inbound call, to simultaneously ring both extensions. Both extensions start ringing. I answer on extension 1001. The call immediately drops, and then begins ringing again in a couple of seconds. >From the log? Line #754: extension 1001 OKs its INVITE. Line #835: debug messages indicates extension 1001 answered the call. Line #854: FreeSWITCH CANCELs the call to extension 1000. Line #887: FreeSWITCH terminates the inbound call with a ?480 Temporarily Unavailable? response. Line #930: FreeSWITCH sends a BYE to extension 1001. Line #1117: a new (regenerated), inbound call request. Without simultaneous ring enabled to two extensions, i.e., ringing only extension 1001 ? the call is handled with no problems. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [ mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps Sent: Thursday, January 15, 2015 12:09 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Michael Collins, I had previously tried the ?Enterprise Originate? and syntax, but it did not make any difference at the time. I retested today with that syntax, and I am still seeing the same problem. If that is the recommended configuration for my situation, I will kept that syntax in my dialplan. I also tested with a difference VoIP service provider. Better results in that the INVITE timer now runs for 28 seconds, as opposed to just 10 seconds with the previous VoIP service provider. During the 28 seconds ? with both extensions ringing ? I was able to answer one extension. However, the call disconnected just as soon as I answered it, and then immediately started ringing again. This continued until the 28 seconds ran out. I will look into this later, and gather additional logs. I also retested, with the new VoIP service provider ? and dialing just one extension works fine. Dialing just one extension also worked with the previous VoIP service provider, even with the 10 second INVITE timer, as long as I answered within the 10 seconds window. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [ mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 10:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Glad you have the book. On page 19 it covers the use of enterprise originate. I think possibly you need to use the method as discussed on page 21. Try something like this: Could just be that early media is not being ignored on both user dialout attempts. -MC On Wed, Jan 14, 2015 at 4:27 AM, George F. Phelps < GeorgePhelps at gfphelps.com> wrote: Michael Collins, I already have the book. Thanks! Here?s my dialplan: New log file uploaded to: http://pastebin.com/gnEpPzk9 To me, the most significant event in the log file is the SIP CANCEL message ? starting at line #321: tport.c:3023 tport_deliver() tport_deliver(0x95daa0): msg 0xad8fb0 (437 bytes) from udp/169.XX.XX.XX:5080/sip next=(nil) nta.c:2880 agent_recv_request() nta: received CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (CSeq 1) nta.c:3026 agent_recv_request() nta: CANCEL (1) is going to INVITE (1) I don?t think it?s related, but I am also curious about log file line #285: sres.c:2987 sres_query_report_error() sres(q=0x98b050): reporting error NAME_ERR for SRV _sip._ udp.sip.switch2voip.us Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 14, 2015 2:41 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? We covered this nicely in chapter 1 of the FreeSWITCH Cookbook I'm sorry that I'm late to the party so I am missing some information. Can you pastebin not only the call log but also the dialplan code for the example in question? One other tip: it appears that the log that you are pasting is coming directly from the FreeSWITCH console. By default the console does not have debug level output enabled. Try entering the command "console loglevel debug" and you'll see way more log lines, mostly yellow text. Those lines will most likely contain the clues needed to unravel this mystery. Thanks, MC On Tue, Jan 13, 2015 at 2:42 PM, George F. Phelps < GeorgePhelps at gfphelps.com> wrote: New logfile uploaded to: http://pastebin.com/CFFvVarS The log contains default Freeswitch console log messages, plus a SIP trace of a failed call. BTW, both extensions were ringing ? prior to the CANCEL message (see context below). In the log I see the INVITE from my VoIP service provider: recv 746 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:14.941233: ------------------------------------------------------------------------ INVITE sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 send 405 bytes to udp/[169.XX.XX.XX]:5060 at 16:22:14.941450: ------------------------------------------------------------------------ SIP/2.0 100 Trying (Then, subsequent INVITE messages to my two extensions. But other no messages to/from my VoIP service provider.) And then, a spontaneous CANCEL from my VoIP service provider, approximately 10 seconds after the initial INVITE message. Due to a SIP ?Timer B? timeout? Seems way too short. recv 435 bytes from udp/[169.XX.XX.XX]:5060 at 16:22:24.104375: ------------------------------------------------------------------------ CANCEL sip:gw+switch2voip.us at 54.XX.XX.XX:5080 SIP/2.0 (Freeswitch cleanup of SIP sessions to my extensions?) Bote Man--> I have two local extensions. Individually, the extensions can make and receive both internal and external calls. It?s only the simultaneous ringing for external, inbound calls that is not working at the moment. Thanks, George From: freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Tuesday, January 13, 2015 2:33 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I suggest you configure and register 3 total local phones to your FS installation, configure 2 of them as the target of your simultaneous ring group, and call them with the 3rd phone. Until you can get that working, calling through a carrier is adding another layer of complexity to the problem and confusing the issue. Out of the box FreeSWITCH does not utter voice codes, they must be coming from your carrier. Also, the debug-level logs very likely tell you exactly what is happening, even though they can be staggering to decipher as a newcomer to FS. Learning how to read them pays off in so many ways, though. I find the color-coded logs on the console or viewed via FS_cli to be helpful in these instances. Bote From: George F. Phelps Sent: Tuesday, 13 January, 2015 08:10 Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? I tried? ?but that did not resolve the problem. 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: Tuesday, January 13, 2015 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Correct, first endpoint providing audio wins, but you're using ignore_early_media... Try using Which is global. And I believe in the dial string also is. But try it anyway. On Jan 13, 2015 1:50 PM, "George F. Phelps" < GeorgePhelps at gfphelps.com> wrote: David Govea, It appears that the essence of the problem is: [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49714 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 192.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:1001 at 50.XX.XX.XX:49219 [CS_CONSUME_MEDIA] [LOSE_RACE] Various Freeswitch web comments, related to the same problem, indicate that I should: ?Ok. Setting it per leg didn't help [ignore_early_media=true], but per channel {ignore_early_media=true} worked?. What dialplan(?) syntax do I use to correctly ?set ignore_early_media=true? on a per channel basis? I tried, within my dialplan? ?but without success. Or does setting ignore_early_media have to be done somewhere else? 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: Tuesday, January 13, 2015 6:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You need to have both extensions registered. Register both and try again and paste de log. On Jan 13, 2015 12:30 PM, "George F. Phelps" < GeorgePhelps at gfphelps.com> wrote: For the most recent test/logfile, only extension 1001 was registered ? to reduce the number of debug messages in the logfile. 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: Monday, January 12, 2015 6:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Is 1000 registered? The log says it's not registered... On Tue, Jan 13, 2015 at 12:11 AM, George F. Phelps < GeorgePhelps at gfphelps.com> wrote: David Govea, I uploaded a new Freeswitch debug logfile at: http://pastebin.com/v17SyXhh Notes Only extension 1001 was registered for this test. Dialstring segment: I?m guessing that ?verbal error code 231? is being generated by my VoIP service provider. I am running Freeswitch with (mostly) the default configuration. Changed passphrases, added my gateway, etc. I downloaded the source code from git and built it unmodified, from scratch. ?FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)? My effective codec is G711U ? fully supported throughout the call chain. 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: Monday, January 12, 2015 7:15 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? BTW, I've never heard of verbal error code 231, that's why I ask whether you downloaded and freeswitch from the git... On Mon, Jan 12, 2015 at 1:12 PM, David Villasmil Govea < david.villasmil at gmail.com> wrote: Are you using freeswitch with its default config or did you install something like fusionpbx? Can you please post your log now? the log for the last dial string, where calls go out and then get hung up. (Are you sure your codecs are correct?) On Mon, Jan 12, 2015 at 1:06 PM, George F. Phelps < GeorgePhelps at gfphelps.com> wrote: David Govea, Still fails; both extensions rang. However, before I can answer either one, I heard the same verbal error code: ?231?. How do I track down the meaning of ?231?? 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: Monday, January 12, 2015 6:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? You can also try: bridge user/1001:_:user/1002 On Jan 12, 2015 12:04 PM, "George F. Phelps" < GeorgePhelps at gfphelps.com> wrote: David Govea, That syntax, with more than one extension specified, causes the following Freeswitch warning log message: [WARNING] switch_ivr_originate.c:2531 Only calling the first element in the list in this mode. However, the call ? to only the first extension on the list ? does work. 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: Monday, January 12, 2015 3:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? try this: On Jan 12, 2015 4:33 AM, "George F. Phelps" < GeorgePhelps at gfphelps.com> wrote: Here you go: Symbol ${domain} resolves to the local LAN, IP address. 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, January 11, 2015 10:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Cab you paste your dialplan? Also, never EVER show your ip addresses. On Jan 12, 2015 2:48 AM, "George F. Phelps" < GeorgePhelps at gfphelps.com> wrote: Yes, I tested with that dialstring. My extension was registered, and online. The call disconnects with verbal error code ?231?. The associated logfile is at: http://pastebin.com/BeWhhgSU 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, January 11, 2015 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Did you try the dialstring with /sofia/internal/1001% your_ip ? If extension 1001 is registered they should get the call. What happens when you do that? On Jan 12, 2015 2:01 AM, "George F. Phelps" < GeorgePhelps at gfphelps.com> wrote: David Govea, I am attempting to implement simultaneous ringing ? where when one of my inbound DIDs is called, then two SIP extensions and one outbound DID are all rung at the same time. Simultaneous ringing is also referred, in the Freeswitch documentation, as ?forked dialing? and ?calling multiple destinations.? I am trying to get the first extension to work with ?bridge.? This Freeswitch example shows bridging (I thought?) to two (2) extensions: Calling multiple destinations By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialed in a multi-threaded manner (this is referred to as "Enterprise Origination") - this gives more flexibility (and avoids the "Only calling the first element in the list in this mode" warning) If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example: 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, January 11, 2015 7:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? Sorry, I thought you wanted to call the user 1001, because you spoke about bridge. You can't "bridge" to an extension. Can you please explain in detail what you want to do? On Jan 12, 2015 1:29 AM, "George F. Phelps" < GeorgePhelps at gfphelps.com> wrote: David Govea, Thanks for your input. I tried that coding yesterday, and the call failed. I wasn?t 100 percent sure I was using the correct coding. When I call, I hear spoken error ?231? and then the call hangs up. I created a pastebin.com of the failed call log, at: http://pastebin.com/BeWhhgSU A reminder that this ?transfer? statement works: 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, January 11, 2015 4:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to Bridge To Local Extensions? https://wiki.freeswitch.org/wiki/Dialplan_XML#Example_6:_Calling_registered_user That's: Note the % sign..., not @ On Jan 11, 2015 10:09 PM, "George F. Phelps" < GeorgePhelps at gfphelps.com> wrote: Can someone help me with my question? Thanks, George From: George F. Phelps [mailto: GeorgePhelps at gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users at lists.freeswitch.org Subject: How to Bridge To Local Extensions? The ?transfer? statement, shown below, works (in my inbound dialplan): What is the correct syntax for using ?bridge? instead of ?transfer?? The following statement does not work for me: My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml My goal is to configure simultaneous ringing for multiple extensions: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/c29928f4/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/20150122/c29928f4/attachment-0001.bin From gb at cm.nl Thu Jan 22 16:14:08 2015 From: gb at cm.nl (Grant Bagdasarian) Date: Thu, 22 Jan 2015 13:14:08 +0000 Subject: [Freeswitch-users] Issue in .NET call control using ESL DLL Message-ID: Hello, I'm experiencing a rather weird issue when using the ESL libraries for .NET. When running the below code outside of Visual Studio in its own executable(Console Application) I'm receiving the below exception. When the code is run inside Visual Studio, it works perfect. Using ESL library contained in Freeswitch 1.2.13. Unhandled Exception: System.AccessViolationException: Attempted to read or write protected memory. This is often an indication that other memory is corrupt. at CSharp_ESLevent_Serialize(Void* jarg1, SByte* jarg2) in c:\users\gb\downloads\freeswitch-1.2.13\freeswitch-1.2.13\libs\esl\managed\esl_wrap.cpp:line 413 at ManagedEsl.ESLPINVOKE.ESLevent_Serialize(HandleRef jarg1, String jarg2) at VoiceEventSocketDemoClient.Program.Main(String[] args) in c:\Users\gb\Documents\Visual Studio 2012\Projects\VoiceEventSocketDemoClient\VoiceEventSocketDemoClient\Program.cs:line 26 TcpListener server = new TcpListener(IPAddress.Parse("LISTEN.IP.GOES.HERE"), 5000); server.Start(); TcpClient eslClient = server.AcceptTcpClient(); ESLconnection eslConnection = new ESLconnection(eslClient.Client.Handle.ToInt32()); ESLevent e = eslConnection.GetInfo(); Console.WriteLine(e.Serialize("json")); Does anyone know why this is happening? Maybe someone can reproduce the problem on their own PC using the above code example. Thank you! Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/d5fb70f9/attachment.html From peter at olssononline.se Thu Jan 22 16:26:27 2015 From: peter at olssononline.se (Peter Olsson) Date: Thu, 22 Jan 2015 14:26:27 +0100 Subject: [Freeswitch-users] Issue in .NET call control using ESL DLL In-Reply-To: References: Message-ID: Version 1.2.13 is ancient. Please start by using a current version, and then report back to the list. /Peter 2015-01-22 14:14 GMT+01:00 Grant Bagdasarian : > Hello, > > > > I?m experiencing a rather weird issue when using the ESL libraries for > .NET. > > > > When running the below code outside of Visual Studio in its own > executable(Console Application) I?m receiving the below exception. > > When the code is run inside Visual Studio, it works perfect. > > > > Using ESL library contained in Freeswitch 1.2.13. > > > > Unhandled Exception: System.AccessViolationException: Attempted to read or > write protected memory. This is often an indication that other memory is > corrupt. > > at CSharp_ESLevent_Serialize(Void* jarg1, SByte* jarg2) in > c:\users\gb\downloads\freeswitch-1.2.13\freeswitch-1.2.13\libs\esl\managed\esl_wrap.cpp:line > 413 > > at ManagedEsl.ESLPINVOKE.ESLevent_Serialize(HandleRef jarg1, String > jarg2) > > at VoiceEventSocketDemoClient.Program.Main(String[] args) in > c:\Users\gb\Documents\Visual Studio > 2012\Projects\VoiceEventSocketDemoClient\VoiceEventSocketDemoClient\Program.cs:line > 26 > > > > TcpListener server = new TcpListener(IPAddress.Parse("LISTEN.IP.GOES.HERE" > ), 5000); > > > > server.Start(); > > TcpClient eslClient = server.AcceptTcpClient(); > > ESLconnection eslConnection = new ESLconnection(eslClient.Client.Handle. > ToInt32()); > > ESLevent e = eslConnection.GetInfo(); > > Console.WriteLine(e.Serialize("json")); > > > > Does anyone know why this is happening? Maybe someone can reproduce the > problem on their own PC using the above code example. > > > > Thank you! > > > > Regards, > > > > Grant > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/21706d8b/attachment.html From kris at kriskinc.com Thu Jan 22 16:35:25 2015 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 22 Jan 2015 08:35:25 -0500 Subject: [Freeswitch-users] Register more than 100 gateways on a host In-Reply-To: References: Message-ID: I've done a lot worse for various testing and simulation scenarios. FS will be fine. On Thursday, January 22, 2015, Regis M wrote: > Hi, > > I had a project where I could have to register between 100 and 500 > gateways/trunk on a host. > > Does anyone have already done something similar ? > Have you some problems with this configuration or recommandations ? > > For me, It"s equivalent to 250 sip phones register on me... it's just that > : > > sofia status > > > will give me some lines :) > > Regards > > R?gis > > -- Sent from mobile device -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/37003b8b/attachment-0001.html From gb at cm.nl Thu Jan 22 17:20:40 2015 From: gb at cm.nl (Grant Bagdasarian) Date: Thu, 22 Jan 2015 14:20:40 +0000 Subject: [Freeswitch-users] Issue in .NET call control using ESL DLL In-Reply-To: References: Message-ID: <4f56b44ef6de464787066b4146d77268@CM-EX-V01.cm.local> Will the version of the ESL library contained in the current version of Freeswitch work with the 1.2.13 version of Freeswitch? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Thursday, January 22, 2015 2:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Issue in .NET call control using ESL DLL Version 1.2.13 is ancient. Please start by using a current version, and then report back to the list. /Peter 2015-01-22 14:14 GMT+01:00 Grant Bagdasarian >: Hello, I?m experiencing a rather weird issue when using the ESL libraries for .NET. When running the below code outside of Visual Studio in its own executable(Console Application) I?m receiving the below exception. When the code is run inside Visual Studio, it works perfect. Using ESL library contained in Freeswitch 1.2.13. Unhandled Exception: System.AccessViolationException: Attempted to read or write protected memory. This is often an indication that other memory is corrupt. at CSharp_ESLevent_Serialize(Void* jarg1, SByte* jarg2) in c:\users\gb\downloads\freeswitch-1.2.13\freeswitch-1.2.13\libs\esl\managed\esl_wrap.cpp:line 413 at ManagedEsl.ESLPINVOKE.ESLevent_Serialize(HandleRef jarg1, String jarg2) at VoiceEventSocketDemoClient.Program.Main(String[] args) in c:\Users\gb\Documents\Visual Studio 2012\Projects\VoiceEventSocketDemoClient\VoiceEventSocketDemoClient\Program.cs:line 26 TcpListener server = new TcpListener(IPAddress.Parse("LISTEN.IP.GOES.HERE"), 5000); server.Start(); TcpClient eslClient = server.AcceptTcpClient(); ESLconnection eslConnection = new ESLconnection(eslClient.Client.Handle.ToInt32()); ESLevent e = eslConnection.GetInfo(); Console.WriteLine(e.Serialize("json")); Does anyone know why this is happening? Maybe someone can reproduce the problem on their own PC using the above code example. Thank you! Regards, Grant _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/162fd1e9/attachment.html From krice at freeswitch.org Thu Jan 22 17:31:09 2015 From: krice at freeswitch.org (Ken Rice) Date: Thu, 22 Jan 2015 08:31:09 -0600 Subject: [Freeswitch-users] Issue in .NET call control using ESL DLL In-Reply-To: <4f56b44ef6de464787066b4146d77268@CM-EX-V01.cm.local> Message-ID: The entire 1.2 branch is no longer supported and has been moved to EOL status. You should seriously consider upgrading to atleast the latest 1.4 On 1/22/15, 8:20 AM, "Grant Bagdasarian" wrote: > Will the version of the ESL library contained in the current version of > Freeswitch work with the 1.2.13 version of Freeswitch? > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter > Olsson > Sent: Thursday, January 22, 2015 2:26 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Issue in .NET call control using ESL DLL > > > Version 1.2.13 is ancient. Please start by using a current version, and then > report back to the list. > > > > /Peter > > > > 2015-01-22 14:14 GMT+01:00 Grant Bagdasarian : > > Hello, > > I?m experiencing a rather weird issue when using the ESL libraries for .NET. > > When running the below code outside of Visual Studio in its own > executable(Console Application) I?m receiving the below exception. > When the code is run inside Visual Studio, it works perfect. > > Using ESL library contained in Freeswitch 1.2.13. > > Unhandled Exception: System.AccessViolationException: Attempted to read or > write protected memory. This is often an indication that other memory is > corrupt. > at CSharp_ESLevent_Serialize(Void* jarg1, SByte* jarg2) in > c:\users\gb\downloads\freeswitch-1.2.13\freeswitch-1.2.13\libs\esl\managed\esl > _wrap.cpp:line 413 > at ManagedEsl.ESLPINVOKE.ESLevent_Serialize(HandleRef jarg1, String jarg2) > at VoiceEventSocketDemoClient.Program.Main(String[] args) in > c:\Users\gb\Documents\Visual Studio > 2012\Projects\VoiceEventSocketDemoClient\VoiceEventSocketDemoClient\Program.cs > :line 26 > > TcpListener server = new TcpListener(IPAddress.Parse("LISTEN.IP.GOES.HERE"), > 5000); > > server.Start(); > TcpClient eslClient = server.AcceptTcpClient(); > ESLconnection eslConnection = new > ESLconnection(eslClient.Client.Handle.ToInt32()); > ESLevent e = eslConnection.GetInfo(); > Console.WriteLine(e.Serialize("json")); > > Does anyone know why this is happening? Maybe someone can reproduce the > problem on their own PC using the above code example. > > Thank you! > > Regards, > > Grant > > _________________________________________________________________________ > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/dd0d4b49/attachment-0001.html From gb at cm.nl Thu Jan 22 17:53:14 2015 From: gb at cm.nl (Grant Bagdasarian) Date: Thu, 22 Jan 2015 14:53:14 +0000 Subject: [Freeswitch-users] Issue in .NET call control using ESL DLL In-Reply-To: References: <4f56b44ef6de464787066b4146d77268@CM-EX-V01.cm.local> Message-ID: <14bbc70040724add9756c7aba905f8a7@CM-EX-V01.cm.local> Ohh, I didn't know that. Thank you for notifying me about this! Just to answer my own question: the ESL library version in the 1.4.15 version of Freeswitch does work with 1.2.13 of Freeswitch. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, January 22, 2015 3:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Issue in .NET call control using ESL DLL The entire 1.2 branch is no longer supported and has been moved to EOL status. You should seriously consider upgrading to atleast the latest 1.4 On 1/22/15, 8:20 AM, "Grant Bagdasarian" wrote: Will the version of the ESL library contained in the current version of Freeswitch work with the 1.2.13 version of Freeswitch? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Thursday, January 22, 2015 2:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Issue in .NET call control using ESL DLL Version 1.2.13 is ancient. Please start by using a current version, and then report back to the list. /Peter 2015-01-22 14:14 GMT+01:00 Grant Bagdasarian : Hello, I'm experiencing a rather weird issue when using the ESL libraries for .NET. When running the below code outside of Visual Studio in its own executable(Console Application) I'm receiving the below exception. When the code is run inside Visual Studio, it works perfect. Using ESL library contained in Freeswitch 1.2.13. Unhandled Exception: System.AccessViolationException: Attempted to read or write protected memory. This is often an indication that other memory is corrupt. at CSharp_ESLevent_Serialize(Void* jarg1, SByte* jarg2) in c:\users\gb\downloads\freeswitch-1.2.13\freeswitch-1.2.13\libs\esl\managed\esl_wrap.cpp:line 413 at ManagedEsl.ESLPINVOKE.ESLevent_Serialize(HandleRef jarg1, String jarg2) at VoiceEventSocketDemoClient.Program.Main(String[] args) in c:\Users\gb\Documents\Visual Studio 2012\Projects\VoiceEventSocketDemoClient\VoiceEventSocketDemoClient\Program.cs:line 26 TcpListener server = new TcpListener(IPAddress.Parse("LISTEN.IP.GOES.HERE"), 5000); server.Start(); TcpClient eslClient = server.AcceptTcpClient(); ESLconnection eslConnection = new ESLconnection(eslClient.Client.Handle.ToInt32()); ESLevent e = eslConnection.GetInfo(); Console.WriteLine(e.Serialize("json")); Does anyone know why this is happening? Maybe someone can reproduce the problem on their own PC using the above code example. Thank you! Regards, Grant _________________________________________________________________________ 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/40010c61/attachment.html From nbhatti at gmail.com Thu Jan 22 23:53:33 2015 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 22 Jan 2015 23:53:33 +0300 Subject: [Freeswitch-users] Unable to access channel vars after session terminates evens with set_zombie_exec() Message-ID: I need to access some channel variables being set in the dialplan, but the channel is already hanged up too quick, ORIGINATOR_CANCEL. Since the session was still setting up the channel variables. I have tried app, set_zombie_exec but still not able to see any channel vars set in the dial plan. Am i doing something wrong or if there is a better way to see the channel vars if the session terminates too soon? With:?git a067a49 Dialplan: sofia/internal/9401404 at 10.211.55.26 Regex (PASS) [internal] destination_number(1786866) =~ /$/ break=on-false Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set_zombie_exec()? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(hangup_after_bridge=true)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(continue_on_fail=true)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(fail_on_single_reject=USER_BUSY,NO_ANSWER,NO_USER_RESPONSE,ORIGINATOR_CANCEL)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(disable_hold=true)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(failed_json_cdr_prefix=failed_cdr_index)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(debug_cdr=0)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(debug_cdr_sql=1)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(cust_default_lrn=intra)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(cust_lrn_dip_cost=0.01817)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(cust_jurisdiction=INTRASTATE)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit(hash random_xgw 0.0.0.0 10/1 !FACILITY_REJECTED)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit_execute(hash outbound random_xgw 20 bridge [enable_heartbeat_events=5,nibble_rate=0.2192,nibble_increment=5,nibble_account=AB8KA191,carrier_switch=random_xgw,carrier_switch_id=1,carrier_ratecard_id=2,carrier_rate_rev=1,carrier_rate_type=lrn,carrier_id=1,carrier_connection_cost=0,carrier_rate=0.0119,carrier_interstate_cost=0.0119,carrier_intrastate_cost=0.0119,carrier_enable_billing=t,carrier_call_increment=1,carrier_min_duration=5,carrier_balance=17099.88924]sofia/gateway/random_xgw/1786866)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit(hash switch02 0.0.0.0 10/1 !FACILITY_REJECTED)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit_execute(hash outbound switch02 20 bridge [enable_heartbeat_events=5,nibble_rate=0.2192,nibble_increment=5,nibble_account=AB8KA191,carrier_switch=switch02,carrier_switch_id=3,carrier_ratecard_id=2,carrier_rate_rev=1,carrier_rate_type=lrn,carrier_id=1,carrier_connection_cost=0,carrier_rate=0.0119,carrier_interstate_cost=0.0119,carrier_intrastate_cost=0.0119,carrier_enable_billing=t,carrier_call_increment=1,carrier_min_duration=5,carrier_balance=17099.88924]sofia/gateway/switch02/1786866)? Dialplan: sofia/internal/9401404 at 10.211.55.26 Action hangup()? 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/9401404 at 10.211.55.26) State Change CS_ROUTING -> CS_EXECUTE 2015-01-22 11:47:14.856013 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/9401404 at 10.211.55.26 [BREAK] 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/9401404 at 10.211.55.26) State ROUTING going to sleep 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/9401404 at 10.211.55.26) Running State Change CS_EXECUTE 2015-01-22 11:47:14.856013 [DEBUG] sofia.c:6614 Channel sofia/internal/9401404 at 10.211.55.26 entering state [terminated][487] 2015-01-22 11:47:14.856013 [NOTICE] sofia.c:7530 Hangup sofia/internal/9401404 at 10.211.55.26 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2015-01-22 11:47:14.856013 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/9401404 at 10.211.55.26 [KILL] 2015-01-22 11:47:14.856013 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/9401404 at 10.211.55.26 [BREAK] 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9401404 at 10.211.55.26) State EXECUTE 2015-01-22 11:47:14.856013 [DEBUG] mod_sofia.c:178 sofia/internal/9401404 at 10.211.55.26 SOFIA EXECUTE 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9401404 at 10.211.55.26) State EXECUTE going to sleep ? ? Thanks, Muhammad Naseer Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/4d2e1d5d/attachment-0001.html From karl-theo_hofer at inteli-sim.com Fri Jan 23 01:55:12 2015 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Thu, 22 Jan 2015 23:55:12 +0100 Subject: [Freeswitch-users] perl script excute hangup causes segfault on freeswitch Message-ID: <54C17FD0.1050700@inteli-sim.com> Hi There we want to hang up a channel after some time-out of not detecting DTMF or typing the wrong DTMF 3 times a perl script sends execute hangup and in deed sometime the channel is being terminated but sometime the free switch is eating all its memory and getting a segfault. we can not see why is FS not executing the hangup nor the reason for the segfault any suggestion/help would be relay appreciated. -- With best regards Karl Theo Hofer From brian at freeswitch.org Fri Jan 23 01:58:27 2015 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2015 16:58:27 -0600 Subject: [Freeswitch-users] perl script excute hangup causes segfault on freeswitch In-Reply-To: <54C17FD0.1050700@inteli-sim.com> References: <54C17FD0.1050700@inteli-sim.com> Message-ID: using DBI? On Thu, Jan 22, 2015 at 4:55 PM, kthofer wrote: > Hi There > > we want to hang up a channel after some time-out of not detecting DTMF > or typing the wrong DTMF 3 times > a perl script sends execute hangup and in deed sometime the channel is > being terminated but sometime the free switch is eating all its memory > and getting a segfault. > > we can not see why is FS not executing the hangup nor the reason for the > segfault > > any suggestion/help would be relay appreciated. > > > -- > With best regards > > Karl Theo Hofer > > > _________________________________________________________________________ > 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/20150122/7d3dde38/attachment.html From karl-theo_hofer at inteli-sim.com Fri Jan 23 02:06:51 2015 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Fri, 23 Jan 2015 00:06:51 +0100 Subject: [Freeswitch-users] perl script excute hangup causes segfault on freeswitch In-Reply-To: References: <54C17FD0.1050700@inteli-sim.com> Message-ID: <54C1828B.6070704@inteli-sim.com> Hi Brian yes we do this to check the amount of potential messages but it does not look like the DBI is causing the issue why should DBI connect be a cause for not processing the execute hangup and the reason for the segfault? Kr With best regards Karl Theo Hofer Brian West skrev den 2015-01-22 23:58: > using DBI? > > On Thu, Jan 22, 2015 at 4:55 PM, kthofer > wrote: > >> Hi There >> >> we want to hang up a channel after some time-out of not detecting DTMF >> or typing the wrong DTMF 3 times >> a perl script sends execute hangup and in deed sometime the channel is >> being terminated but sometime the free switch is eating all its memory >> and getting a segfault. >> >> we can not see why is FS not executing the hangup nor the reason for the >> segfault >> >> any suggestion/help would be relay appreciated. >> >> >> -- >> With best regards >> >> Karl Theo Hofer >> >> >> _________________________________________________________________________ >> 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 >> > > > > > using DBI? > > On Thu, Jan 22, 2015 at 4:55 PM, kthofer > > wrote: > > Hi There > > we want to hang up a channel after some time-out of not detecting DTMF > or typing the wrong DTMF 3 times > a perl script sends execute hangup and in deed sometime the > channel is > being terminated but sometime the free switch is eating all its > memory > and getting a segfault. > > we can not see why is FS not executing the hangup nor the reason > for the > segfault > > any suggestion/help would be relay appreciated. > > > -- > With best regards > > Karl Theo Hofer > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/cfd42d8d/attachment.html From mike at jerris.com Fri Jan 23 02:11:07 2015 From: mike at jerris.com (Michael Jerris) Date: Thu, 22 Jan 2015 18:11:07 -0500 Subject: [Freeswitch-users] Unable to access channel vars after session terminates evens with set_zombie_exec() In-Reply-To: References: Message-ID: <70BDF5FD-9594-437A-A8CF-46B7DAE061E2@jerris.com> if you set them using inline they will be set during routing state, before it goes to execute. Once it is in execute state, it checks status between every action. > On Jan 22, 2015, at 3:53 PM, Muhammad Naseer Bhatti wrote: > > > I need to access some channel variables being set in the dialplan, but the channel is already hanged up too quick, ORIGINATOR_CANCEL. Since the session was still setting up the channel variables. I have tried app, set_zombie_exec but still not able to see any channel vars set in the dial plan. Am i doing something wrong or if there is a better way to see the channel vars if the session terminates too soon? > > With: git a067a49 > > Dialplan: sofia/internal/9401404 at 10.211.55.26 Regex (PASS) [internal] destination_number(1786866) =~ /$/ break=on-false > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set_zombie_exec() > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(hangup_after_bridge=true) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(continue_on_fail=true) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(fail_on_single_reject=USER_BUSY,NO_ANSWER,NO_USER_RESPONSE,ORIGINATOR_CANCEL) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(disable_hold=true) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(failed_json_cdr_prefix=failed_cdr_index) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(debug_cdr=0) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(debug_cdr_sql=1) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(cust_default_lrn=intra) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(cust_lrn_dip_cost=0.01817) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(cust_jurisdiction=INTRASTATE) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit(hash random_xgw 0.0.0.0 10/1 !FACILITY_REJECTED) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit_execute(hash outbound random_xgw 20 bridge [enable_heartbeat_events=5,nibble_rate=0.2192,nibble_increment=5,nibble_account=AB8KA191,carrier_switch=random_xgw,carrier_switch_id=1,carrier_ratecard_id=2,carrier_rate_rev=1,carrier_rate_type=lrn,carrier_id=1,carrier_connection_cost=0,carrier_rate=0.0119,carrier_interstate_cost=0.0119,carrier_intrastate_cost=0.0119,carrier_enable_billing=t,carrier_call_increment=1,carrier_min_duration=5,carrier_balance=17099.88924]sofia/gateway/random_xgw/1786866) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit(hash switch02 0.0.0.0 10/1 !FACILITY_REJECTED) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit_execute(hash outbound switch02 20 bridge [enable_heartbeat_events=5,nibble_rate=0.2192,nibble_increment=5,nibble_account=AB8KA191,carrier_switch=switch02,carrier_switch_id=3,carrier_ratecard_id=2,carrier_rate_rev=1,carrier_rate_type=lrn,carrier_id=1,carrier_connection_cost=0,carrier_rate=0.0119,carrier_interstate_cost=0.0119,carrier_intrastate_cost=0.0119,carrier_enable_billing=t,carrier_call_increment=1,carrier_min_duration=5,carrier_balance=17099.88924]sofia/gateway/switch02/1786866) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action hangup() > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/9401404 at 10.211.55.26 ) State Change CS_ROUTING -> CS_EXECUTE > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/9401404 at 10.211.55.26 [BREAK] > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/9401404 at 10.211.55.26 ) State ROUTING going to sleep > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/9401404 at 10.211.55.26 ) Running State Change CS_EXECUTE > 2015-01-22 11:47:14.856013 [DEBUG] sofia.c:6614 Channel sofia/internal/9401404 at 10.211.55.26 entering state [terminated][487] > 2015-01-22 11:47:14.856013 [NOTICE] sofia.c:7530 Hangup sofia/internal/9401404 at 10.211.55.26 [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2015-01-22 11:47:14.856013 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/9401404 at 10.211.55.26 [KILL] > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/9401404 at 10.211.55.26 [BREAK] > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9401404 at 10.211.55.26 ) State EXECUTE > 2015-01-22 11:47:14.856013 [DEBUG] mod_sofia.c:178 sofia/internal/9401404 at 10.211.55.26 SOFIA EXECUTE > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9401404 at 10.211.55.26 ) State EXECUTE going to sleep > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150122/d609c321/attachment-0001.html From ahabiba at gmail.com Fri Jan 23 02:47:35 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Fri, 23 Jan 2015 02:47:35 +0300 Subject: [Freeswitch-users] Session Limit In-Reply-To: References: Message-ID: <3AB70777-417F-4E3B-9269-5BB1777C89C7@gmail.com> Dears, kindly Im facing some issue with concurrent session limit, below is the status showing the maximum sessions of 355, however I set maximum sessions to 10000, you kind help to identify why freeswitch use lower limit than the configured one will be appreciated. /etc/freeswitch# fs_cli -x "status" UP 0 years, 0 days, 0 hours, 19 minutes, 2 seconds, 490 milliseconds, 264 microseconds FreeSWITCH (Version 1.5.8b+git git 35f2bcc 2014-02-12 23:33:16Z 64bit) is ready 345 session(s) since startup 345 session(s) - peak 345, last 5min 345 0 session(s) per Sec out of max 1000, peak 66, last 5min 0 335 session(s) max min idle cpu 0.00/62.00 Current Stack Size/Max 240K/8192K switch.conf.xml: UP 0 years, 0 days, 0 hours, 19 minutes, 2 seconds, 490 milliseconds, 264 microseconds > FreeSWITCH (Version 1.5.8b+git git 35f2bcc 2014-02-12 23:33:16Z 64bit) is ready > 345 session(s) since startup > 345 session(s) - peak 345, last 5min 345 > 0 session(s) per Sec out of max 1000, peak 66, last 5min 0 > 335 session(s) max > min idle cpu 0.00/62.00 > Current Stack Size/Max 240K/8192K > > switch.conf.xml: > > > > UP 0 years, 0 days, 0 hours, 19 minutes, 2 seconds, 490 milliseconds, 264 microseconds > FreeSWITCH (Version 1.5.8b+git git 35f2bcc 2014-02-12 23:33:16Z 64bit) is ready > 345 session(s) since startup > 345 session(s) - peak 345, last 5min 345 > 0 session(s) per Sec out of max 1000, peak 66, last 5min 0 > 335 session(s) max > min idle cpu 0.00/62.00 > Current Stack Size/Max 240K/8192K > > switch.conf.xml: > > > > UP 0 years, 0 days, 0 hours, 19 minutes, 2 seconds, 490 milliseconds, 264 microseconds > FreeSWITCH (Version 1.5.8b+git git 35f2bcc 2014-02-12 23:33:16Z 64bit) is ready > 345 session(s) since startup > 345 session(s) - peak 345, last 5min 345 > 0 session(s) per Sec out of max 1000, peak 66, last 5min 0 > 335 session(s) max > min idle cpu 0.00/62.00 > Current Stack Size/Max 240K/8192K > > switch.conf.xml: > > > > References: <3841721F-555B-44DF-87B6-540699A2C64E@gmx.net> Message-ID: It's normal for the Call-ID to be different each time - each message is its own dialog. Can you not group conversations based on the sender+receiver pair? On 23 January 2015 at 10:28, mbo wrote: > Hello, > > We have implemented a web based client based on the jsSIP library to do > audio/video and chat via SIP over websocket and freeswitch. With this > client it is also possible to chat only. > When chatting between two clients, the Call-ID in the message is different > for each message. > > MESSAGE sip:1008 at 192.168.0.123 SIP/2.0 > Via: SIP/2.0/WSS rfboa1c9gg6q.invalid;branch=z9hG4bK2512434 > Max-Forwards: 69 > To: > From: ;tag=e472qkncp0 > Call-ID: 0ggfl2u355ebrf77jsrq CSeq: 521 MESSAGE > sessionId: some uuid > Content-Type: text/plain > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS > Supported: outbound > User-Agent: JsSIP 0.6.0 > Content-Length: 2 > > First problem: As the Call-ID is different for each message, I cannot > group the messages to a conversation. > Second problem: I want to store the chat conversation in the database and > link it to a CDR. But there is no CDR written for the conversation. > > As I can see in the RFC (https://www.ietf.org/rfc/rfc3428.txt) on page 3, > it should also be possible to create a dialog and then send all messages in > context of this dialog. I assume this means, it is possible to establish a > chat only (INVITE) call and then send all messages in context of the call. > If there is a call then freeswitch will also write a cdr and if I can send > messages in context of that call I can group them to a conversation, insert > it to the db and link it to the cdr. How can I do this with freeswitch? > > Thanks > > Markus > > > > > > > > > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/fae7a758/attachment-0001.html From steveayre at gmail.com Fri Jan 23 17:01:00 2015 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 23 Jan 2015 14:01:00 +0000 Subject: [Freeswitch-users] bridge after first call answers In-Reply-To: References: Message-ID: Use {ignore_early_media=true}loopback/5XXX/default This means the originate only succeeds on answer (instead of on ringing), so the call to 5YYY won't happen until they answer. On 23 January 2015 at 07:13, Necati Demir wrote: > Hello, > > When I run the following command, freeswitch make call to two of these > numbers concurrently. > fs_cli -x "originate loopback/5XXX/default & bridge(loopback/5YYY/default)" > > But I want a scenario like this: > Freeswitch first calls 5XXX. If 5XXX answers, then freeswitch calls 5YYY. > > How to do this? > > > -- > Necati DEM?R > -------------------- > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/42f0e7f8/attachment.html From krice at freeswitch.org Fri Jan 23 17:03:56 2015 From: krice at freeswitch.org (Ken Rice) Date: Fri, 23 Jan 2015 08:03:56 -0600 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: <110d3c1b48924b28af43b24ce7db189e@CM-EX-V01.cm.local> References: <51c6dda8792f450aa93d4278575b2f02@CM-EX-V01.cm.local> <110d3c1b48924b28af43b24ce7db189e@CM-EX-V01.cm.local> Message-ID: <6981BDFE-DC5C-46ED-9250-D1D72827DEED@freeswitch.org> On what platform are you getting this error? Sent from my iPhone > On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian wrote: > > Nevermind. > I used the 1.4.15 tarball which contained this error. > > Now I?ve used the git clone master which works fine. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian > Sent: Friday, January 23, 2015 11:51 AM > To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) > Subject: [Freeswitch-users] APR issue when configuring > > Hello, > > I?m getting the following error during configure. > > checking for APR... configure: error: the --with-apr parameter is incorrect. It must specify an install prefix, a build directory, or an apr-config file. > configure: error: ./configure.gnu failed for libs/apr-util > > I?m running the following command to configure: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support > > Also, a lot of directories inside the libs require the configure script to be executable. I have to set this 1 by 1. > > Is there a way to disable apr? Or is it required by FS? > > Regards, > > Grant > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/2e3b9443/attachment.html From kamil.nigmatullin at gmail.com Fri Jan 23 17:19:58 2015 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Fri, 23 Jan 2015 20:19:58 +0600 Subject: [Freeswitch-users] SIP for chat messages only In-Reply-To: References: <3841721F-555B-44DF-87B6-540699A2C64E@gmx.net> Message-ID: If you want to dig that deep i think you may want to use sipproxy for that purpose. Are you trying to implement on android? If so it is better to use GCM push system for messaging 23 ???. 2015 ?. 20:01 ???????????? "Steven Ayre" ???????: > It's normal for the Call-ID to be different each time - each message is > its own dialog. > > Can you not group conversations based on the sender+receiver pair? > > On 23 January 2015 at 10:28, mbo wrote: > >> Hello, >> >> We have implemented a web based client based on the jsSIP library to do >> audio/video and chat via SIP over websocket and freeswitch. With this >> client it is also possible to chat only. >> When chatting between two clients, the Call-ID in the message is >> different for each message. >> >> MESSAGE sip:1008 at 192.168.0.123 SIP/2.0 >> Via: SIP/2.0/WSS rfboa1c9gg6q.invalid;branch=z9hG4bK2512434 >> Max-Forwards: 69 >> To: >> From: ;tag=e472qkncp0 >> Call-ID: 0ggfl2u355ebrf77jsrq > CSeq: 521 MESSAGE >> sessionId: some uuid >> Content-Type: text/plain >> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS >> Supported: outbound >> User-Agent: JsSIP 0.6.0 >> Content-Length: 2 >> >> First problem: As the Call-ID is different for each message, I cannot >> group the messages to a conversation. >> Second problem: I want to store the chat conversation in the database and >> link it to a CDR. But there is no CDR written for the conversation. >> >> As I can see in the RFC (https://www.ietf.org/rfc/rfc3428.txt) on page >> 3, it should also be possible to create a dialog and then send all messages >> in context of this dialog. I assume this means, it is possible to establish >> a chat only (INVITE) call and then send all messages in context of the call. >> If there is a call then freeswitch will also write a cdr and if I can >> send messages in context of that call I can group them to a conversation, >> insert it to the db and link it to the cdr. How can I do this with >> freeswitch? >> >> Thanks >> >> Markus >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/c1eb09eb/attachment.html From alipey at gmail.com Fri Jan 23 17:27:45 2015 From: alipey at gmail.com (Ali Pey) Date: Fri, 23 Jan 2015 09:27:45 -0500 Subject: [Freeswitch-users] [ERR] switch_core_io.c:187 sofia/SIPRouter/1917xxxxxxx has no read codec In-Reply-To: References: Message-ID: There is no remote SDP. It sends an Invite and a Cancel right away before it even gets ringing. It's a bridge though, there is an aleg that is already established and this happens for the bleg. Local SDP only has G711 ulaw and alaw with ptime 20ms. Thanks. On Fri, Jan 23, 2015 at 3:57 AM, Steven Ayre wrote: > What do the local and remote SDP look like? > > On 23 January 2015 at 05:53, Ali Pey wrote: > >> Hello, >> >> I get this error time to time to certain numbers. There is an incoming >> call and freeswitch is trying to do a bridge to this number. Codec on all >> calls is only G.711. >> >> When I see this error, I see freeswitch sends an Invite message and then >> Cancel in less than one second after it receives 100 Trying. >> >> Any suggestion what I should look for or why this pops up. It results to >> outbound call failure and hangup cause is 88: INCOMPATIBLE_DESTINATION >> >> Any suggestion or advice would be greatly appreciated. >> >> Thanks, >> Ali Pey >> >> >> _________________________________________________________________________ >> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/2264a0f9/attachment-0001.html From gb at cm.nl Fri Jan 23 17:49:25 2015 From: gb at cm.nl (Grant Bagdasarian) Date: Fri, 23 Jan 2015 14:49:25 +0000 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: <6981BDFE-DC5C-46ED-9250-D1D72827DEED@freeswitch.org> References: <51c6dda8792f450aa93d4278575b2f02@CM-EX-V01.cm.local> <110d3c1b48924b28af43b24ce7db189e@CM-EX-V01.cm.local> <6981BDFE-DC5C-46ED-9250-D1D72827DEED@freeswitch.org> Message-ID: lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 7.8 (wheezy) Release: 7.8 Codename: wheezy uname -a Linux HOSTNAME 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Friday, January 23, 2015 3:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring On what platform are you getting this error? Sent from my iPhone On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian > wrote: Nevermind. I used the 1.4.15 tarball which contained this error. Now I?ve used the git clone master which works fine. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian Sent: Friday, January 23, 2015 11:51 AM To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) Subject: [Freeswitch-users] APR issue when configuring Hello, I?m getting the following error during configure. checking for APR... configure: error: the --with-apr parameter is incorrect. It must specify an install prefix, a build directory, or an apr-config file. configure: error: ./configure.gnu failed for libs/apr-util I?m running the following command to configure: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support Also, a lot of directories inside the libs require the configure script to be executable. I have to set this 1 by 1. Is there a way to disable apr? Or is it required by FS? Regards, Grant _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/bb9f3c00/attachment.html From mbodbg at gmx.net Fri Jan 23 17:50:32 2015 From: mbodbg at gmx.net (mbo) Date: Fri, 23 Jan 2015 15:50:32 +0100 Subject: [Freeswitch-users] SIP for chat messages only In-Reply-To: References: <3841721F-555B-44DF-87B6-540699A2C64E@gmx.net> Message-ID: For grouping I can also create a uuid se custom sip headers, but then I still do not have an cdr for a chat. But if it?s not possible to generate a cdr for this at all, I need to think about another concept. Thanks Markus Am 23.01.2015 um 14:59 schrieb Steven Ayre : > It's normal for the Call-ID to be different each time - each message is its own dialog. > > Can you not group conversations based on the sender+receiver pair? > > On 23 January 2015 at 10:28, mbo wrote: > Hello, > > We have implemented a web based client based on the jsSIP library to do audio/video and chat via SIP over websocket and freeswitch. With this client it is also possible to chat only. > When chatting between two clients, the Call-ID in the message is different for each message. > > MESSAGE sip:1008 at 192.168.0.123 SIP/2.0 > Via: SIP/2.0/WSS rfboa1c9gg6q.invalid;branch=z9hG4bK2512434 > Max-Forwards: 69 > To: > From: ;tag=e472qkncp0 > Call-ID: 0ggfl2u355ebrf77jsrq CSeq: 521 MESSAGE > sessionId: some uuid > Content-Type: text/plain > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS > Supported: outbound > User-Agent: JsSIP 0.6.0 > Content-Length: 2 > > First problem: As the Call-ID is different for each message, I cannot group the messages to a conversation. > Second problem: I want to store the chat conversation in the database and link it to a CDR. But there is no CDR written for the conversation. > > As I can see in the RFC (https://www.ietf.org/rfc/rfc3428.txt) on page 3, it should also be possible to create a dialog and then send all messages in context of this dialog. I assume this means, it is possible to establish a chat only (INVITE) call and then send all messages in context of the call. > If there is a call then freeswitch will also write a cdr and if I can send messages in context of that call I can group them to a conversation, insert it to the db and link it to the cdr. How can I do this with freeswitch? > > Thanks > > Markus > > > > > > > > > > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/c584de84/attachment.html From krice at freeswitch.org Fri Jan 23 18:05:34 2015 From: krice at freeswitch.org (Ken Rice) Date: Fri, 23 Jan 2015 15:05:34 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <54c2633e1426d_4ec07d9330688b9@ip-10-155-230-70.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/95e64552/attachment-0001.html From krice at freeswitch.org Fri Jan 23 18:43:33 2015 From: krice at freeswitch.org (Ken Rice) Date: Fri, 23 Jan 2015 09:43:33 -0600 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: Message-ID: I?m inclined to believe you have something broken on your box here. I just grabbed the tarball and built it right on my Wheezy box... How exactly are you trying to build it from the tarball? On 1/23/15, 8:49 AM, "Grant Bagdasarian" wrote: > lsb_release -a > No LSB modules are available. > Distributor ID: Debian > Description: Debian GNU/Linux 7.8 (wheezy) > Release: 7.8 > Codename: wheezy > > uname -a > Linux HOSTNAME 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice > Sent: Friday, January 23, 2015 3:04 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] APR issue when configuring > > > On what platform are you getting this error? > > > > > Sent from my iPhone > > > On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian wrote: >> >> Nevermind. >> I used the 1.4.15 tarball which contained this error. >> >> Now I?ve used the git clone master which works fine. >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant >> Bagdasarian >> Sent: Friday, January 23, 2015 11:51 AM >> To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) >> Subject: [Freeswitch-users] APR issue when configuring >> >> Hello, >> >> I?m getting the following error during configure. >> >> checking for APR... configure: error: the --with-apr parameter is incorrect. >> It must specify an install prefix, a build directory, or an apr-config file. >> configure: error: ./configure.gnu failed for libs/apr-util >> >> I?m running the following command to configure: ./configure >> -prefix=/usr/src/freeswitch/ --enable-core-odbc-support >> >> Also, a lot of directories inside the libs require the configure script to be >> executable. I have to set this 1 by 1. >> >> Is there a way to disable apr? Or is it required by FS? >> >> Regards, >> >> Grant >> >> _________________________________________________________________________ >> 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150123/f96998ab/attachment.html From mike at jerris.com Fri Jan 23 19:53:25 2015 From: mike at jerris.com (Michael Jerris) Date: Fri, 23 Jan 2015 11:53:25 -0500 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 103, Issue 247 In-Reply-To: References: Message-ID: <11BF4A15-817B-4FE1-AAA5-80CE31C59E73@jerris.com> You have some limits on your system causing thread create failure. Look at your ulimits or if you are running virtualized, all the related limits there. > On Jan 22, 2015, at 7:13 PM, Ahmed Habiba wrote: > > Thank you really Michael, > > I go through logs and I find the below, the question why could lead to that, the machine is 16 core?? > > 2015-01-23 02:06:59.593497 [CRIT] switch_core_session.c:1780 Thread Failure! > 2015-01-23 02:06:59.593497 [CRIT] switch_core_session.c:1713 LUKE: I'm hit, but not bad. > 2015-01-23 02:06:59.593497 [CRIT] switch_core_session.c:1714 LUKE'S VOICE: Artoo, see what you can do with it. Hang on back there.... > Green laserfire moves past the beeping little robot as his head turns. After a few beeps and a twist of his mechanical arm, > Artoo reduces the max sessions to 346 thus, saving the switch from certain doom. > > > From: Michael Jerris > > To: FreeSWITCH Users Help > > Date: January 23, 2015 at 2:51:18 AM GMT+3 > Reply-To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Session Limit > > > look at the logs, if you extend beyond your resources and freeswitch can't start threads among other things, it will lower your session limit. > >> On Jan 22, 2015, at 6:47 PM, Ahmed Habiba > wrote: >> >> Dears, >> >> kindly Im facing some issue with concurrent session limit, below is the status showing the maximum sessions of 355, however I set maximum sessions to 10000, you kind help to identify why freeswitch use lower limit than the configured one will be appreciated. >> >> /etc/freeswitch# fs_cli -x "status" >> UP 0 years, 0 days, 0 hours, 19 minutes, 2 seconds, 490 milliseconds, 264 microseconds >> FreeSWITCH (Version 1.5.8b+git git 35f2bcc 2014-02-12 23:33:16Z 64bit) is ready >> 345 session(s) since startup >> 345 session(s) - peak 345, last 5min 345 >> 0 session(s) per Sec out of max 1000, peak 66, last 5min 0 >> 335 session(s) max >> min idle cpu 0.00/62.00 >> Current Stack Size/Max 240K/8192K >> >> switch.conf.xml: >> >> >> >> ;tag=gFX13g0aS2Z8D. To: . Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696564 INVITE. Contact: . Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE. Supported: path. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 286. P-Asserted-Identity: "DISPLAY NAME" . . v=0. o=WSC_PBX 1422043530 1422043531 IN IP4 A.A.A.A. s=SIP Media Capabilities. c=IN IP4 A.A.A.A. t=0 0. m=audio 21726 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # U B.B.B.B:5060 -> A.A.A.A:5080 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP A.A.A.A:5080;rport=5080;branch=z9hG4bKjp8H9Q9vUvmZp. From: "DISPLAY NAME" ;tag=gFX13g0aS2Z8D. To: ;tag=45e7e4a19f9e2adbb8c0c696f8c31422.8a84. Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696564 INVITE. Proxy-Authenticate: Digest realm="A.A.A.A", nonce="VMRPz1TC/k8QLkFKz6mNTenFQ4hT8yIrSb4tyoA=", qop="auth". Content-Length: 0. . # U A.A.A.A:5080 -> B.B.B.B:5060 ACK sip:1NPANXX2222 at B.B.B.B SIP/2.0. Via: SIP/2.0/UDP A.A.A.A:5080;rport;branch=z9hG4bKjp8H9Q9vUvmZp. Max-Forwards: 69. From: "DISPLAY NAME" ;tag=gFX13g0aS2Z8D. To: ;tag=45e7e4a19f9e2adbb8c0c696f8c31422.8a84. Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696564 ACK. Content-Length: 0. . # U A.A.A.A:5080 -> B.B.B.B:5060 INVITE sip:1NPANXX2222 at B.B.B.B SIP/2.0. Via: SIP/2.0/UDP A.A.A.A:5080;rport;branch=z9hG4bKKZ1aBKt0r5ajj. Max-Forwards: 69. From: "DISPLAY NAME" ;tag=gFX13g0aS2Z8D. To: . Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696565 INVITE. Contact: . Expires: 90. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE. Supported: path. Proxy-Authorization: Digest username="8729701", realm="A.A.A.A", nonce="VMRPz1TC/k8QLkFKz6mNTenFQ4hT8yIrSb4tyoA=", cnonce="m4QpfR4QEjOeoQAYURJGhg", algorithm=MD5, uri="sip:1NPANXX2222 at B.B.B.B", response="744aa6cda6803838bcf8c87d1810c6ed", qop=auth, nc=00000001. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 286. P-Asserted-Identity: "DISPLAY NAME" . . v=0. o=WSC_PBX 1422043530 1422043531 IN IP4 A.A.A.A. s=SIP Media Capabilities. c=IN IP4 A.A.A.A. t=0 0. m=audio 21726 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/1642547f/attachment.html From alhakeem at gmail.com Sat Jan 24 06:33:57 2015 From: alhakeem at gmail.com (Abdul Hakeem) Date: Sat, 24 Jan 2015 03:33:57 -0000 Subject: [Freeswitch-users] Fork( ) and Exec ( ) functions In-Reply-To: References: <6053000.JvnxKzh3j9@sos> <1C00AF43-07EF-4A86-8CB0-286C49617B5F@jerris.com> Message-ID: Thanks guys. I?ll do battle with the makefile next week & see how it goes. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, January 19, 2015 5:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fork( ) and Exec ( ) functions We use threads by default in the switch system. There is a setting to use fork instead but it's disabled by default. The system syscall uses fork and exec on its own so maybe you just need to make sure ifdefs properly disable any such functionality. On Monday, January 19, 2015, Michael Jerris wrote: You can also look at how we have addressed this issue for windows. The places I know of that use these functions. switch_system (used for executing external commands and sending emails), and daemonizing FreeSWITCH. > On Jan 18, 2015, at 3:48 PM, Sergey Okhapkin > wrote: > > Usually fork-less OSes provide spawn() family of syscalls to execute a new > process. > > On Sunday 18 January 2015 20:32:02 Abdul Hakeem wrote: >> Hello again, >> The reason I ask is because I am trying to port FS to OSv. >> Unfortunately, Fork ( ) is unsupported in OSv. >> Regards, >> Abdul Hakeem >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Moises >> Silva Sent: Tuesday, January 13, 2015 5:11 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Fork( ) and Exec ( ) functions >> >> On Tue, Jan 13, 2015 at 8:11 AM, Abdul Hakeem > wrote: >> Hello, >> >> I understand FS makes system calls for sending mails, voicemail and fax. >> Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on >> system calls & also, a list of functions which require FS to make system >> calls ? >> >> Not making much sense here. Everything in FS relies heavily on system calls >> as it's a multi-threaded-I/O-driven system for the most part. I think >> you're asking the wrong question. _________________________________________________________________________ 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 -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+ ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/151c9e70/attachment-0001.html From john.nash778 at gmail.com Sat Jan 24 07:52:06 2015 From: john.nash778 at gmail.com (John Nash) Date: Sat, 24 Jan 2015 10:22:06 +0530 Subject: [Freeswitch-users] Playing file using lua before bridge Message-ID: I am trying to play some native files (Pre encoded) using lua script just after call is setup using Bridge (But before any answer). I need to use "Proxy_Media" (Want media to go through freeswitch but no codec translation). I tried using "bridge_pre_execute_aleg_app" but I do not hear sound files being played (Do not see any error message also). I have already tested lua script in some other test extension and it works fine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/862ee73f/attachment.html From john.nash778 at gmail.com Sat Jan 24 08:33:32 2015 From: john.nash778 at gmail.com (John Nash) Date: Sat, 24 Jan 2015 11:03:32 +0530 Subject: [Freeswitch-users] Play native file in proxy_media true mode Message-ID: Is it possible to play a native sound file in proxy_media mode? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/2a9b6dd0/attachment.html From brian at freeswitch.org Sat Jan 24 09:15:49 2015 From: brian at freeswitch.org (Brian West) Date: Sat, 24 Jan 2015 00:15:49 -0600 Subject: [Freeswitch-users] Play native file in proxy_media true mode In-Reply-To: References: Message-ID: No On Friday, January 23, 2015, John Nash wrote: > Is it possible to play a native sound file in proxy_media mode? > -- *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/20150124/393c8e2c/attachment.html From john.nash778 at gmail.com Sat Jan 24 09:50:25 2015 From: john.nash778 at gmail.com (John Nash) Date: Sat, 24 Jan 2015 12:20:25 +0530 Subject: [Freeswitch-users] Play native file in proxy_media true mode In-Reply-To: References: Message-ID: Ok thank you. One more quick question, in Bypass_media=true settings is there any way to change SDP manually?..for example may be "s" or "o" field? On Sat, Jan 24, 2015 at 11:45 AM, Brian West wrote: > No > > On Friday, January 23, 2015, John Nash wrote: > >> Is it possible to play a native sound file in proxy_media mode? >> > > > -- > > *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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/4721ebc2/attachment.html From babak.freeswitch at gmail.com Sat Jan 24 11:26:43 2015 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Sat, 24 Jan 2015 11:56:43 +0330 Subject: [Freeswitch-users] spandsp_start_dtmf detecting first digit 2 times Message-ID: Hi I upgraded from 1.2.12 to 1.4.15+git~20150106T122408Z~f5145540b3~64bit and now with no change to configs ,after spandsp_start_dtmf in playAndGetDigits I see the first digit is detected twice: 2015-01-24 11:51:05.083629 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [3] 2015-01-24 11:51:05.083629 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:1747 done playing file /usr/local/freeswitch/sounds/fa/ir/callie/applications/2/yazd_2030_long.wav 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:410 digits '3' 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:561 IVR action on menu 'ivr_10' matched '3' param 'lua lua/applications/call_script.lua 48 ' 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:565 switch_ivr_menu_execute todo=[2] EXECUTE sofia/internal/87654321 at 172.16.90.3:5060 lua(lua/applications/call_script.lua 48 ) 2015-01-24 11:51:05.083629 [DEBUG] freeswitch_lua.cpp:360 DBH handle 0x7f36b80d0630 Connected. EXECUTE sofia/internal/87654321 at 172.16.90.3:5060 lua(/usr/local/freeswitch/scripts/gui/lua/yazd/yzd_phone_activation.lua ) 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [fa] 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[applications/11/yzd_act_cid_notice.wav] (fa:fa) 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16 at 8000hz 1 channels 20ms 2015-01-24 11:51:05.123638 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [3], duration = 38 ms 2015-01-24 11:51:05.163633 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [3] 2015-01-24 11:51:05.163633 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 2015-01-24 11:51:05.343664 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [3], duration = 178 ms 2015-01-24 11:51:05.863613 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [5] 2015-01-24 11:51:05.863613 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 2015-01-24 11:51:06.123594 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [5], duration = 255 ms 2015-01-24 11:51:06.403613 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [6] 2015-01-24 11:51:06.403613 [DEBUG] switch_channel.c:488 RECV DTMF 6:2000 2015-01-24 11:51:06.643596 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [6], duration = 242 ms 2015-01-24 11:51:07.083609 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [9] 2015-01-24 11:51:07.083609 [DEBUG] switch_channel.c:488 RECV DTMF 9:2000 .... the digit 3 is detected twice! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/91d3be4f/attachment.html From babak.freeswitch at gmail.com Sat Jan 24 13:34:04 2015 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Sat, 24 Jan 2015 14:04:04 +0330 Subject: [Freeswitch-users] spandsp_start_dtmf detecting first digit 2 times In-Reply-To: References: Message-ID: it seems problem is not related to spandsp, because the same happens when I use start_dtmf! On Sat, Jan 24, 2015 at 11:56 AM, Babak Yakhchali < babak.freeswitch at gmail.com> wrote: > Hi > I upgraded from 1.2.12 to 1.4.15+git~20150106T122408Z~f5145540b3~64bit and > now with no change to configs ,after spandsp_start_dtmf in > playAndGetDigits I see the first digit is detected twice: > 2015-01-24 11:51:05.083629 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [3] > 2015-01-24 11:51:05.083629 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 > 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:1747 done playing > file > /usr/local/freeswitch/sounds/fa/ir/callie/applications/2/yazd_2030_long.wav > 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:410 digits '3' > 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:561 IVR action on > menu 'ivr_10' matched '3' param 'lua lua/applications/call_script.lua 48 ' > 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:565 > switch_ivr_menu_execute todo=[2] > EXECUTE sofia/internal/87654321 at 172.16.90.3:5060 > lua(lua/applications/call_script.lua 48 ) > 2015-01-24 11:51:05.083629 [DEBUG] freeswitch_lua.cpp:360 DBH handle > 0x7f36b80d0630 Connected. > EXECUTE sofia/internal/87654321 at 172.16.90.3:5060 > lua(/usr/local/freeswitch/scripts/gui/lua/yazd/yzd_phone_activation.lua ) > 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:70 No language > specified - Using [fa] > 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:250 Handle > play-file:[applications/11/yzd_act_cid_notice.wav] (fa:fa) > 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:1314 Codec > Activated L16 at 8000hz 1 channels 20ms > 2015-01-24 11:51:05.123638 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [3], duration = 38 ms > 2015-01-24 11:51:05.163633 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [3] > 2015-01-24 11:51:05.163633 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 > 2015-01-24 11:51:05.343664 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [3], duration = 178 ms > 2015-01-24 11:51:05.863613 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [5] > 2015-01-24 11:51:05.863613 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 > 2015-01-24 11:51:06.123594 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [5], duration = 255 ms > 2015-01-24 11:51:06.403613 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [6] > 2015-01-24 11:51:06.403613 [DEBUG] switch_channel.c:488 RECV DTMF 6:2000 > 2015-01-24 11:51:06.643596 [DEBUG] mod_spandsp_dsp.c:385 DTMF END > DETECTED: [6], duration = 242 ms > 2015-01-24 11:51:07.083609 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN > DETECTED: [9] > 2015-01-24 11:51:07.083609 [DEBUG] switch_channel.c:488 RECV DTMF 9:2000 > .... > the digit 3 is detected twice! > Thanks > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/817727a3/attachment-0001.html From ahabiba at gmail.com Sat Jan 24 17:05:51 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Sat, 24 Jan 2015 17:05:51 +0300 Subject: [Freeswitch-users] mod_snmp not loading In-Reply-To: References: Message-ID: <785DEB27-8C3D-4582-9B93-0EA973DD07A2@gmail.com> Dears, kindly I tried to use mod_snmp I compile it using make && make install, however when I tried to load it I got the below message:, your kind usual support will be appreciated: 2015-01-24 15:05:58.673046 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_snmp.so **/usr/lib/libnetsnmpagent.so.15: undefined symbol: netsnmp_register_null_context** Thanks, Ahmed Habiba. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/49ce76c2/attachment.html From msc at freeswitch.org Sat Jan 24 22:30:37 2015 From: msc at freeswitch.org (Michael Collins) Date: Sat, 24 Jan 2015 11:30:37 -0800 Subject: [Freeswitch-users] any handset recommendations that operate like polycom+cisco? In-Reply-To: <54B82BA0.20201@mst.edu> References: <54B82BA0.20201@mst.edu> Message-ID: Hi Nate, Sorry for the late reply. Just curious - what is the application here? I'm curious what the person using this phone would be doing with dozens of SIP accounts. Alternatively, would something like Voice Operator Panel be a better choice, even though it's basically a soft-phone and not a hard phone? Thanks, Michael On Thu, Jan 15, 2015 at 1:05 PM, Nathan Neulinger wrote: > Looking for any handset/desk phone recommendations that have the "every > line key can be it's own SIP account, even on > expansion modules" behavior. > > Yealink units for example only allow a set number of lines, and the rest > don't operate as line keys, so even though you > can have a 30 + button phone, it's not usable for a receptionist or > secretary that may handle calls for 15 different > people with their own direct lines. > > With the polycom and cisco (sccp) phones, you can define N different SIP > accounts. > > Are there any other phones out there that operate this way that anyone can > recommend? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/79cd7295/attachment.html From spencer at whiteskycommunications.com Sat Jan 24 22:40:11 2015 From: spencer at whiteskycommunications.com (Spencer Thomason) Date: Sat, 24 Jan 2015 19:40:11 +0000 Subject: [Freeswitch-users] Expires in INVITE after challenge In-Reply-To: <7ACD3382-5247-4118-A9F9-B13B58F62D0E@whiteskycommunications.com> References: <7ACD3382-5247-4118-A9F9-B13B58F62D0E@whiteskycommunications.com> Message-ID: Just to follow up, I did a little homework and I?ve moved this to Jira FS-7192. Thanks, Spencer On Jan 23, 2015, at 6:14 PM, Spencer Thomason > wrote: Hello all, I?ve been reviewing a bunch of pcaps today during interop with a new ULC and I noticed that our FreeSWITCH instances send an Expires header in the INVITE that wasn?t present before the challenge. I?ve got ;tag=gFX13g0aS2Z8D. To: . Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696564 INVITE. Contact: . Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE. Supported: path. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 286. P-Asserted-Identity: "DISPLAY NAME" . . v=0. o=WSC_PBX 1422043530 1422043531 IN IP4 A.A.A.A. s=SIP Media Capabilities. c=IN IP4 A.A.A.A. t=0 0. m=audio 21726 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # U B.B.B.B:5060 -> A.A.A.A:5080 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP A.A.A.A:5080;rport=5080;branch=z9hG4bKjp8H9Q9vUvmZp. From: "DISPLAY NAME" ;tag=gFX13g0aS2Z8D. To: ;tag=45e7e4a19f9e2adbb8c0c696f8c31422.8a84. Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696564 INVITE. Proxy-Authenticate: Digest realm="A.A.A.A", nonce="VMRPz1TC/k8QLkFKz6mNTenFQ4hT8yIrSb4tyoA=", qop="auth". Content-Length: 0. . # U A.A.A.A:5080 -> B.B.B.B:5060 ACK sip:1NPANXX2222 at B.B.B.B SIP/2.0. Via: SIP/2.0/UDP A.A.A.A:5080;rport;branch=z9hG4bKjp8H9Q9vUvmZp. Max-Forwards: 69. From: "DISPLAY NAME" ;tag=gFX13g0aS2Z8D. To: ;tag=45e7e4a19f9e2adbb8c0c696f8c31422.8a84. Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696564 ACK. Content-Length: 0. . # U A.A.A.A:5080 -> B.B.B.B:5060 INVITE sip:1NPANXX2222 at B.B.B.B SIP/2.0. Via: SIP/2.0/UDP A.A.A.A:5080;rport;branch=z9hG4bKKZ1aBKt0r5ajj. Max-Forwards: 69. From: "DISPLAY NAME" ;tag=gFX13g0aS2Z8D. To: . Call-ID: 9b82c095-1e10-1233-a19e-001851124686. CSeq: 70696565 INVITE. Contact: . Expires: 90. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE. Supported: path. Proxy-Authorization: Digest username="8729701", realm="A.A.A.A", nonce="VMRPz1TC/k8QLkFKz6mNTenFQ4hT8yIrSb4tyoA=", cnonce="m4QpfR4QEjOeoQAYURJGhg", algorithm=MD5, uri="sip:1NPANXX2222 at B.B.B.B", response="744aa6cda6803838bcf8c87d1810c6ed", qop=auth, nc=00000001. Privacy: none. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 286. P-Asserted-Identity: "DISPLAY NAME" . . v=0. o=WSC_PBX 1422043530 1422043531 IN IP4 A.A.A.A. s=SIP Media Capabilities. c=IN IP4 A.A.A.A. t=0 0. m=audio 21726 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/c13b8254/attachment.html From msc at freeswitch.org Sat Jan 24 22:43:31 2015 From: msc at freeswitch.org (Michael Collins) Date: Sat, 24 Jan 2015 11:43:31 -0800 Subject: [Freeswitch-users] voicemail silence_detection is always detecting silence In-Reply-To: References: Message-ID: Hi Raphael, I looked at your pastebin and this is just a guess but it appears that the DTMF digit 1 being received (at line 235/236) may be getting interpreted by mod_voicemail as being the user pressing a digit to stop the recording. Since the recording is only started less than one second earlier (I'm guessing around line 234) it doesn't meet the minimum message length of 3 seconds. One way you could test this is to execute flush_dtmf prior to sending the channel to voicemail. -MC On Fri, Jan 16, 2015 at 4:24 AM, Raphael Lechner wrote: > I missed to write the used FreeSWITCH version. I tried with 1.4.14 and 1.4.15+git~20141229T185951Z~507a0f22c5~64bit > (git 507a0f2 2014-12-29 18:59:51Z 64bit) > > Thank you > > On 16 Jan 2015, at 11:09, Raphael Lechner > wrote: > > Hi, > > I configured an extension that first call for some seconds a phone and if > nobody is picking up, the caller is hearing a playback and can press 1 for > leaving a voicemail and 2 to get redirected to a mobile phone. > The Problem is that in my test environment with a SIP Provider everything > works fine after tuning the voicemail.conf.xml to > > > minimum record length: 3? on the calling phone. > After that I can record a message and that works as expected. > Is there a way do disable the silence_detection or any hint what I can > change? > > I tried changing the silence-threshold to 1,50 and silence-hits to > 300,30000 but nothing has changed > > Debug Log > https://pastebin.freeswitch.org/23851 > > The called python script: > def handler(session, args): > voicemail = args.split(' ')[0] > dtmf_pressed = args.split(' ')[1] > forward_number = args.split(' ')[2] > callerid = session.getVariable("caller_id_number") > callername = session.getVariable("caller_id_name").lstrip() > > if dtmf_pressed == '1': > send_sms('377XXXXXXX?,?New Voicemail from %s %s' % (callername, > callerid)) > session.execute("export", "skip_greeting=true") > session.execute("export", "skip_instructions=true") > session.execute("answer") > session.execute("voicemail", "default 192.168.17.252 10?) > #session.execute("bridge", "loopback/app=voicemail:default %s %s" > % (conf['network']['ip'],voicemail)) > elif dtmf_pressed == '2': > consoleLog( "info", "Call is forwarded to %s\n" % forward_number) > session.transfer(forward_number, "XML", "default") > else: > consoleLog( "info", "DTMF received is %s and not 1 or 2.Hangup > Call\n" % (dtmf_pressed)) > session.hangup() > > Thank you, > Raphael > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/c5353967/attachment-0001.html From msc at freeswitch.org Sat Jan 24 22:53:08 2015 From: msc at freeswitch.org (Michael Collins) Date: Sat, 24 Jan 2015 11:53:08 -0800 Subject: [Freeswitch-users] fail_on_single_reject and sequential bridge fail-over? In-Reply-To: References: <54BEC5BF.3060300@gmail.com> <0000014b0c1f55b5-b3e9af8d-e244-487a-89f5-35919aaa602b-000000@email.amazonses.com> <54BFC6EC.1080306@gmail.com> Message-ID: On Wed, Jan 21, 2015 at 4:26 PM, Steven Ayre wrote: > My understanding is that today there is no channel variable in FS that >> lets control reject causes to stop bridge iterations over | separated >> endpoints. > > Am I right or am I wrong? > > > fail_on_single_reject should be working for exactly that purpose. For > example I use the following: > > "continue_on_fail=GATEWAY_DOWN,INVALID_GATEWAY"/> > "fail_on_single_reject=!^^:GATEWAY_DOWN:INVALID_GATEWAY"/> > > > > This passes routes via multiple carriers. If the gateway is > down/non-existent it will try the next sequential destination(s), but any > other failure cause causes the bridge to end without trying the next (fail_on_single_reject). > If all the gateways are down it returns to dialplan (continue_on_fail) > otherwise it hangs up with the reject cause from the bridge. > > Hi Steven, I believe your explanation is spot on and makes a lot of sense. From a documentation standpoint I can see why some are confused. Both the wiki and the source code explicitly mention the AND (i.e. comma) condition being affected by fail_on_single_reject but not the OR (i.e. pipe) condition. If there are no objections I'd like to submit this tidbit to be included in the fail_on_single_reject wiki entry. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/e6b51528/attachment.html From nneul at mst.edu Sun Jan 25 00:01:29 2015 From: nneul at mst.edu (Nathan Neulinger) Date: Sat, 24 Jan 2015 15:01:29 -0600 Subject: [Freeswitch-users] any handset recommendations that operate like polycom+cisco? In-Reply-To: References: <54B82BA0.20201@mst.edu> Message-ID: <54C40829.9060501@mst.edu> Personally I think it's nuts... but we have a number of secretary/admin/receptionist users with Cisco expansion modules that have the shared lines of all of the people in their department on them. (i.e. a 7940/7960 plus the module). Usually with some portion of the other lines set to just flash and not audibly ring. While I'd expect that in most cases they really would be sufficient with busy lamp, sometimes they do use it to answer arbitrary calls for faculty that are out of the office/etc. With the transitioned cisco phones on FS/mod_skinny - it works the same way, however we're wanting to position ourselves with suitable replacements, particularly for any departments that want more than bare bones functionality. With the polycom phones, it appears to also work that way where you can have a sip account for every line key if you want - even including the expansion modules. However, on the Yealink phones (got looking at them cause of the T46G I won at ClueCon) we found the number of accounts very limited. It turns out that with the latest firmware (73.x) on the Yealink units the count is increased on a number of the models (to 16 on the T46 for example). The problem is that with the middle tier ones that you'd add an expansion module to - it doesn't really get you anything. If your base phone is limited to 6 accounts, adding the expansion module ONLY gets you busy-lamp or speed dials. We're working on getting the users "converted" to not using full lines wherever possible, but still want options open. -- Nathan On 01/24/2015 01:30 PM, Michael Collins wrote: > Hi Nate, > > Sorry for the late reply. Just curious - what is the application here? I'm curious what the person using this phone > would be doing with dozens of SIP accounts. Alternatively, would something like Voice Operator Panel be a better choice, > even though it's basically a soft-phone and not a hard phone? > > Thanks, > Michael > > > On Thu, Jan 15, 2015 at 1:05 PM, Nathan Neulinger > wrote: > > Looking for any handset/desk phone recommendations that have the "every line key can be it's own SIP account, even on > expansion modules" behavior. > > Yealink units for example only allow a set number of lines, and the rest don't operate as line keys, so even though you > can have a 30 + button phone, it's not usable for a receptionist or secretary that may handle calls for 15 different > people with their own direct lines. > > With the polycom and cisco (sccp) phones, you can define N different SIP accounts. > > Are there any other phones out there that operate this way that anyone can recommend? > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > 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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From babak.freeswitch at gmail.com Sun Jan 25 02:14:20 2015 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Sun, 25 Jan 2015 02:44:20 +0330 Subject: [Freeswitch-users] spandsp_start_dtmf detecting first digit 2 times In-Reply-To: References: Message-ID: setting min_dup_digit_spacing_ms=100 solved my problem On Sat, Jan 24, 2015 at 2:04 PM, Babak Yakhchali wrote: > it seems problem is not related to spandsp, because the same happens when > I use start_dtmf! > > On Sat, Jan 24, 2015 at 11:56 AM, Babak Yakhchali < > babak.freeswitch at gmail.com> wrote: > >> Hi >> I upgraded from 1.2.12 to 1.4.15+git~20150106T122408Z~f5145540b3~64bit >> and now with no change to configs ,after spandsp_start_dtmf in >> playAndGetDigits I see the first digit is detected twice: >> 2015-01-24 11:51:05.083629 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> DETECTED: [3] >> 2015-01-24 11:51:05.083629 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 >> 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:1747 done >> playing file >> /usr/local/freeswitch/sounds/fa/ir/callie/applications/2/yazd_2030_long.wav >> 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:410 digits '3' >> 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:561 IVR action on >> menu 'ivr_10' matched '3' param 'lua lua/applications/call_script.lua 48 ' >> 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_menu.c:565 >> switch_ivr_menu_execute todo=[2] >> EXECUTE sofia/internal/87654321 at 172.16.90.3:5060 >> lua(lua/applications/call_script.lua 48 ) >> 2015-01-24 11:51:05.083629 [DEBUG] freeswitch_lua.cpp:360 DBH handle >> 0x7f36b80d0630 Connected. >> EXECUTE sofia/internal/87654321 at 172.16.90.3:5060 >> lua(/usr/local/freeswitch/scripts/gui/lua/yazd/yzd_phone_activation.lua ) >> 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:70 No language >> specified - Using [fa] >> 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:250 Handle >> play-file:[applications/11/yzd_act_cid_notice.wav] (fa:fa) >> 2015-01-24 11:51:05.083629 [DEBUG] switch_ivr_play_say.c:1314 Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2015-01-24 11:51:05.123638 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> DETECTED: [3], duration = 38 ms >> 2015-01-24 11:51:05.163633 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> DETECTED: [3] >> 2015-01-24 11:51:05.163633 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 >> 2015-01-24 11:51:05.343664 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> DETECTED: [3], duration = 178 ms >> 2015-01-24 11:51:05.863613 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> DETECTED: [5] >> 2015-01-24 11:51:05.863613 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 >> 2015-01-24 11:51:06.123594 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> DETECTED: [5], duration = 255 ms >> 2015-01-24 11:51:06.403613 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> DETECTED: [6] >> 2015-01-24 11:51:06.403613 [DEBUG] switch_channel.c:488 RECV DTMF 6:2000 >> 2015-01-24 11:51:06.643596 [DEBUG] mod_spandsp_dsp.c:385 DTMF END >> DETECTED: [6], duration = 242 ms >> 2015-01-24 11:51:07.083609 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN >> DETECTED: [9] >> 2015-01-24 11:51:07.083609 [DEBUG] switch_channel.c:488 RECV DTMF 9:2000 >> .... >> the digit 3 is detected twice! >> Thanks >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150125/b14a3f94/attachment.html From bdfoster at davri.com Sun Jan 25 05:53:15 2015 From: bdfoster at davri.com (Brian Foster) Date: Sat, 24 Jan 2015 21:53:15 -0500 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: A WiFi card set to "monitor mode" comes to mind... A friend and I are experimenting with using a Raspberry Pi with a WiFi dongle attached, sniffing for MAC addresses. AFAIK, the user would not need to hook up to a specific network. So you end up having a solution that doesn't require much user intervention for it to work. Uses less battery life than bluetooth, and likely the user's WiFi is probably already switched on (if they have a phone with that ability). Something to think about... If you're interested in something like that, send me an email. Maybe we could work on this together. Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana On Tue, Jan 20, 2015 at 11:22 PM, Guillermo Ruiz Camauer < grcamauer at gmail.com> wrote: > Good observation regarding Bluetooth. I was looking for a solution that > did not require a cell phone (just a Bluetooth fob). Any another indoor > tracking technology come to mind? > > Thanks, > > Guillermo > > Sent from my iPhone > > > On 20/1/2015, at 22:32, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > > > > Guillermo, > > > > I dont think bluethood will work on your need. Reason is that Bt has 1 > > to 1 relationship. This means, you need as many BT radios as many > > people are in the room. Putting 1 BT radio in a PC will work with > > first hit, but second person wont be able to be detected. > > > > I think best option is to install a softphone in your smartphone and > > allow multiple registration. When dialing to your extension, it will > > ring on your extension and on your smartphone as well. > > > > > > 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 > > > > > > 2013-04-10 17:49 GMT-04:00 Guillermo Ruiz Camauer : > >> I don't want to hijack your thread, but since the subject matter is > >> FollowMe, I would like to add that I am looking for something along the > >> lines of: > >> > >> A FollowMe that somehow knows what room you are in within a building and > >> rings the nearest extension with a special ringtone which is assigned to > >> each user. > >> The "somehow" could be through a Bluetooth dongle attached to a PC in > the > >> room that detects the User's cell phone and updates a DB that FS has > access > >> to, or an Mobile Phone App that triangulates on WiFi Access Points and > >> updates a DB, etc. > >> > >> Has anyone heard of such a system? Experiences? > >> > >> Thank you, > >> > >> Guillermo Ruiz Camauer > >> > >> > >>> On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster > wrote: > >>> > >>> I've been given an assignment. It's a little rough, and honestly I've > been > >>> working on other projects and at the same time loosing my > freeswitch-fu. So, > >>> here it goes. > >>> > >>> Company owner wants to be able to implement a follow me function. He's > >>> asking for the deskphones to begin ringing, then have cell phones ring > N > >>> seconds later WHILE the deskphones continue to ring. The function has > to be > >>> able to work using a couple different ways of dialing (we've got call > groups > >>> implemented, > >>> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). > When the > >>> mobile phone is answered, we need to be able to get some feedback from > the > >>> callee to figure out if they're human. We'll use AVMD to kill the call > if it > >>> detects a voicemail beep. > >>> > >>> I've looked at several different examples on the wiki and mailing list, > >>> and the only way I can figure out how to do it while keeping the > >>> requirements in mind is to at some point resort to using Loopback > (something > >>> i didnt want to do). > >>> > >>> Requirements are: > >>> 1. Use a custom IVR/menu/something to get a confirmation from the > callee > >>> that they are human (while also keeping it available for > customization he's > >>> wanting a way to blacklist numbers on that same menu). So that rules > out > >>> group_confirm_file, etc. > >>> 2. Use AVMD to kill the call if we detect the call was picked up by > >>> voicemail. > >>> 3. The custom IVR/menu/something isn't used on the deskphones > >>> 4. Deskphones need to continue to ring after the external number leg is > >>> started. I can't timeout the call on the deskphone then call the cell > phone, > >>> or call the deskphone, time it out, then call the deskphone and cell > phone. > >>> 5. It has to work on any type of calling method (so basically, if the > >>> deskphone rings then eventually the cell phone needs to ring to if it's > >>> assigned). > >>> > >>> Has anyone done something similar, and if so, how did you do it? > >>> > >>> Thank you, > >>> > >>> Brian Foster > >>> Project Manager/Owner's Representative > >>> Davri Investments, Incorporated > >>> P: +1-317-787-2686 > >>> M: +1-317-600-9753 > >>> Indianapolis, Indiana > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >>> http://www.cudatel.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.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 > >> > >> > >> > >> -- > >> Guillermo Ruiz Camauer > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >> http://www.cudatel.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/fd9eb292/attachment-0001.html From bdfoster at davri.com Sun Jan 25 05:54:03 2015 From: bdfoster at davri.com (Brian Foster) Date: Sat, 24 Jan 2015 21:54:03 -0500 Subject: [Freeswitch-users] Follow me implementation In-Reply-To: References: Message-ID: Sorry, I just realized you said "without a cell phone"... I got nothing lol Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana On Sat, Jan 24, 2015 at 9:53 PM, Brian Foster wrote: > A WiFi card set to "monitor mode" comes to mind... > > A friend and I are experimenting with using a Raspberry Pi with a WiFi > dongle attached, sniffing for MAC addresses. AFAIK, the user would not need > to hook up to a specific network. So you end up having a solution that > doesn't require much user intervention for it to work. Uses less battery > life than bluetooth, and likely the user's WiFi is probably already > switched on (if they have a phone with that ability). > > Something to think about... If you're interested in something like that, > send me an email. Maybe we could work on this together. > > Thank you, > > Brian Foster > Project Manager/Owner's Representative > Davri Investments, Incorporated > P: +1-317-787-2686 > M: +1-317-600-9753 > Indianapolis, Indiana > > On Tue, Jan 20, 2015 at 11:22 PM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> Good observation regarding Bluetooth. I was looking for a solution that >> did not require a cell phone (just a Bluetooth fob). Any another indoor >> tracking technology come to mind? >> >> Thanks, >> >> Guillermo >> >> Sent from my iPhone >> >> > On 20/1/2015, at 22:32, Luis Daniel Lucio Quiroz < >> luis.daniel.lucio at gmail.com> wrote: >> > >> > Guillermo, >> > >> > I dont think bluethood will work on your need. Reason is that Bt has 1 >> > to 1 relationship. This means, you need as many BT radios as many >> > people are in the room. Putting 1 BT radio in a PC will work with >> > first hit, but second person wont be able to be detected. >> > >> > I think best option is to install a softphone in your smartphone and >> > allow multiple registration. When dialing to your extension, it will >> > ring on your extension and on your smartphone as well. >> > >> > >> > 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 >> > >> > >> > 2013-04-10 17:49 GMT-04:00 Guillermo Ruiz Camauer > >: >> >> I don't want to hijack your thread, but since the subject matter is >> >> FollowMe, I would like to add that I am looking for something along the >> >> lines of: >> >> >> >> A FollowMe that somehow knows what room you are in within a building >> and >> >> rings the nearest extension with a special ringtone which is assigned >> to >> >> each user. >> >> The "somehow" could be through a Bluetooth dongle attached to a PC in >> the >> >> room that detects the User's cell phone and updates a DB that FS has >> access >> >> to, or an Mobile Phone App that triangulates on WiFi Access Points and >> >> updates a DB, etc. >> >> >> >> Has anyone heard of such a system? Experiences? >> >> >> >> Thank you, >> >> >> >> Guillermo Ruiz Camauer >> >> >> >> >> >>> On Wed, Apr 10, 2013 at 6:32 PM, Brian Foster >> wrote: >> >>> >> >>> I've been given an assignment. It's a little rough, and honestly I've >> been >> >>> working on other projects and at the same time loosing my >> freeswitch-fu. So, >> >>> here it goes. >> >>> >> >>> Company owner wants to be able to implement a follow me function. He's >> >>> asking for the deskphones to begin ringing, then have cell phones >> ring N >> >>> seconds later WHILE the deskphones continue to ring. The function has >> to be >> >>> able to work using a couple different ways of dialing (we've got call >> groups >> >>> implemented, >> >>> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups). >> When the >> >>> mobile phone is answered, we need to be able to get some feedback >> from the >> >>> callee to figure out if they're human. We'll use AVMD to kill the >> call if it >> >>> detects a voicemail beep. >> >>> >> >>> I've looked at several different examples on the wiki and mailing >> list, >> >>> and the only way I can figure out how to do it while keeping the >> >>> requirements in mind is to at some point resort to using Loopback >> (something >> >>> i didnt want to do). >> >>> >> >>> Requirements are: >> >>> 1. Use a custom IVR/menu/something to get a confirmation from the >> callee >> >>> that they are human (while also keeping it available for >> customization he's >> >>> wanting a way to blacklist numbers on that same menu). So that rules >> out >> >>> group_confirm_file, etc. >> >>> 2. Use AVMD to kill the call if we detect the call was picked up by >> >>> voicemail. >> >>> 3. The custom IVR/menu/something isn't used on the deskphones >> >>> 4. Deskphones need to continue to ring after the external number leg >> is >> >>> started. I can't timeout the call on the deskphone then call the cell >> phone, >> >>> or call the deskphone, time it out, then call the deskphone and cell >> phone. >> >>> 5. It has to work on any type of calling method (so basically, if the >> >>> deskphone rings then eventually the cell phone needs to ring to if >> it's >> >>> assigned). >> >>> >> >>> Has anyone done something similar, and if so, how did you do it? >> >>> >> >>> Thank you, >> >>> >> >>> Brian Foster >> >>> Project Manager/Owner's Representative >> >>> Davri Investments, Incorporated >> >>> P: +1-317-787-2686 >> >>> M: +1-317-600-9753 >> >>> Indianapolis, Indiana >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> >>> http://www.cudatel.com >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.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 >> >> >> >> >> >> >> >> -- >> >> Guillermo Ruiz Camauer >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> >> http://www.cudatel.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.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 >> >> _________________________________________________________________________ >> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150124/4a1688aa/attachment.html From babak.freeswitch at gmail.com Sun Jan 25 09:47:52 2015 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Sun, 25 Jan 2015 10:17:52 +0330 Subject: [Freeswitch-users] voicemail silence_detection is always detecting silence In-Reply-To: References: Message-ID: It seems you are getting dtmf in two different methods. try to remove start_dtmf. On Sat, Jan 24, 2015 at 11:13 PM, Michael Collins wrote: > Hi Raphael, > > I looked at your pastebin and this is just a guess but it appears that the > DTMF digit 1 being received (at line 235/236) may be getting interpreted by > mod_voicemail as being the user pressing a digit to stop the recording. > Since the recording is only started less than one second earlier (I'm > guessing around line 234) it doesn't meet the minimum message length of 3 > seconds. > > One way you could test this is to execute flush_dtmf > > prior to sending the channel to voicemail. > > -MC > > On Fri, Jan 16, 2015 at 4:24 AM, Raphael Lechner < > raphael.lechner at gmail.com> wrote: > >> I missed to write the used FreeSWITCH version. I tried with 1.4.14 and 1.4.15+git~20141229T185951Z~507a0f22c5~64bit >> (git 507a0f2 2014-12-29 18:59:51Z 64bit) >> >> Thank you >> >> On 16 Jan 2015, at 11:09, Raphael Lechner >> wrote: >> >> Hi, >> >> I configured an extension that first call for some seconds a phone and if >> nobody is picking up, the caller is hearing a playback and can press 1 for >> leaving a voicemail and 2 to get redirected to a mobile phone. >> The Problem is that in my test environment with a SIP Provider everything >> works fine after tuning the voicemail.conf.xml to >> >> >> > minimum record length: 3? on the calling phone. >> After that I can record a message and that works as expected. >> Is there a way do disable the silence_detection or any hint what I can >> change? >> >> I tried changing the silence-threshold to 1,50 and silence-hits to >> 300,30000 but nothing has changed >> >> Debug Log >> https://pastebin.freeswitch.org/23851 >> >> The called python script: >> def handler(session, args): >> voicemail = args.split(' ')[0] >> dtmf_pressed = args.split(' ')[1] >> forward_number = args.split(' ')[2] >> callerid = session.getVariable("caller_id_number") >> callername = session.getVariable("caller_id_name").lstrip() >> >> if dtmf_pressed == '1': >> send_sms('377XXXXXXX?,?New Voicemail from %s %s' % (callername, >> callerid)) >> session.execute("export", "skip_greeting=true") >> session.execute("export", "skip_instructions=true") >> session.execute("answer") >> session.execute("voicemail", "default 192.168.17.252 10?) >> #session.execute("bridge", "loopback/app=voicemail:default %s >> %s" % (conf['network']['ip'],voicemail)) >> elif dtmf_pressed == '2': >> consoleLog( "info", "Call is forwarded to %s\n" % forward_number) >> session.transfer(forward_number, "XML", "default") >> else: >> consoleLog( "info", "DTMF received is %s and not 1 or 2.Hangup >> Call\n" % (dtmf_pressed)) >> session.hangup() >> >> Thank you, >> Raphael >> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150125/304bad2d/attachment-0001.html From groysem at gmail.com Sun Jan 25 18:17:33 2015 From: groysem at gmail.com (Shai Perelman) Date: Sun, 25 Jan 2015 17:17:33 +0200 Subject: [Freeswitch-users] can I use Event_Socket_Outbound to only monitor? Message-ID: hi, I am using https://wiki.freeswitch.org/wiki/Event_Socket_Outbound my goal is to monitor what happens on the extension but without affecting the normal behaviour. I have this dialplan. TagTypeDataOrder[image: add] condition username 9998 1[image: edit] [image: delete] action socket 109.65.149.22:9999 async 3[image: edit] [image: delete] I am calling *9664 (music on hold) and the call is blocking while wating for commands from my side. I want it to answer automatticaly like it would do with out the socket module, and just send me the events regarding that extension thanks Shai ITD Communications -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150125/cf0dc23d/attachment.html From msc at freeswitch.org Mon Jan 26 00:59:31 2015 From: msc at freeswitch.org (Michael Collins) Date: Sun, 25 Jan 2015 13:59:31 -0800 Subject: [Freeswitch-users] Getting Redirecting number in Events (ISDN) In-Reply-To: References: Message-ID: I strongly recommend you spin up a FreeSWITCH 1.4 instance for testing. The 1.2 branch has served its purpose and has been honorably discharged as it were. If what you need gets added to the code base it will not be in 1.2. When you get 1.4 up and running be sure to route an ISDN call through the info dialplan app just to make sure it's not already there and available for your use. If it's available in a channel variable then it should be available to add to events as needed. -Michael On Wed, Jan 21, 2015 at 3:58 AM, DP Siddharth < dp.siddharth at eng.knowlarity.com> wrote: > Hi, > > I am using ISDN (PRI) for handling calls. In a use case we need > Redirecting number as part of freeswitch events. I verified various events > but we get only caller/called. I was looking into the code & found for SS7 > this is available but not for ISDN, just need confirmation if any way > possible to get this. I am using 1.2.24 (1.2.stable) freeswitch for testing. > > following is ISDN SETUP message for reference: > > Prot Disc:Q.931/I.451 (0x08) > Call Ref:242E (Originating side) > Type:SETUP (0x5) > Sending complete: > Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64 > Kbit/s(16) L1Prot:G.711 A-Law(3) > Channel Id:No:13 Type:B-chans(3) Preferred/Implicit > Calling Party Number:8860128000(l:10) plan:isdn(1) > type:national(2)scr:network, provided(3) pres:allowed(0) > Called Party Number:01725218700(l:11) plan:isdn(1) type:subscriber(4) > Redirecting Number:53703417(l:8) plan:isdn(1) type:national(2)scr:user, > not screened(0) pres:restricted(1)reason:Call forwarding unconditional(15) > Redirecting Number:53703417(l:8) plan:isdn(1) type:national(2)scr:user, > not screened(0) pres:restricted(1)reason:Call forwarding unconditional(15) > [ 08 02 24 2e 05 a1 04 03 80 90 a3 18 03 a1 83 8d 6c 0c 21 83 38 38 36 > 30 31 32 38 30 30 30 70 0c > c1 30 31 37 32 35 32 31 38 37 30 30 74 0d 21 20 8f 39 36 35 33 37 30 > 33 34 31 37 74 0d 21 20 8f > 39 36 35 33 37 30 33 34 31 37 ] > > > -- > Thanks & Regards, > D P Siddharth > Director (Platform) > Knowlarity Communications > Ph: +919999115231 > dp.siddharth at eng.knowlarity.com > > *"Come together to build a lasting world-class cloud telephony company > that helps businesses grow"* > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150125/da86ce11/attachment.html From msc at freeswitch.org Mon Jan 26 01:13:34 2015 From: msc at freeswitch.org (Michael Collins) Date: Sun, 25 Jan 2015 14:13:34 -0800 Subject: [Freeswitch-users] any handset recommendations that operate like polycom+cisco? In-Reply-To: <54C40829.9060501@mst.edu> References: <54B82BA0.20201@mst.edu> <54C40829.9060501@mst.edu> Message-ID: On Sat, Jan 24, 2015 at 1:01 PM, Nathan Neulinger wrote: > Personally I think it's nuts... but we have a number of > secretary/admin/receptionist users with Cisco expansion modules > that have the shared lines of all of the people in their department on > them. (i.e. a 7940/7960 plus the module). Usually > with some portion of the other lines set to just flash and not audibly > ring. While I'd expect that in most cases they > really would be sufficient with busy lamp, sometimes they do use it to > answer arbitrary calls for faculty that are out > of the office/etc. > > With the transitioned cisco phones on FS/mod_skinny - it works the same > way, however we're wanting to position ourselves > with suitable replacements, particularly for any departments that want > more than bare bones functionality. > > With the polycom phones, it appears to also work that way where you can > have a sip account for every line key if you > want - even including the expansion modules. > > However, on the Yealink phones (got looking at them cause of the T46G I > won at ClueCon) we found the number of accounts > very limited. > > It turns out that with the latest firmware (73.x) on the Yealink units the > count is increased on a number of the models > (to 16 on the T46 for example). The problem is that with the middle tier > ones that you'd add an expansion module to - it > doesn't really get you anything. If your base phone is limited to 6 > accounts, adding the expansion module ONLY gets you > busy-lamp or speed dials. > > We're working on getting the users "converted" to not using full lines > wherever possible, but still want options open. > > -- Nathan > Thanks for the explanation. I share your feelings about the T46. I love that phone but hate the fact that you only get 6 SIP accounts. (Glad to hear that they added more in a recent firmware - I'll test that out at some point...) If you find a solution other than the Cisco one I would be interested in hearing about it. Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150125/43e7d653/attachment.html From phenix at vfemail.net Mon Jan 26 00:43:14 2015 From: phenix at vfemail.net (Tanguy) Date: Sun, 25 Jan 2015 22:43:14 +0100 Subject: [Freeswitch-users] Freeswitch don't send keepalive Message-ID: <54C56372.5070708@vfemail.net> Hello I'm a asterisk user, and i would like to try freeswitch for having a real multi tenant system. I configured FS with fusionpbx. My FS is installed on a VPS ( has a public IP ), my extensions are behind a nat router. After few minutes my phone became unreachable for inbound call. After a outbound call, the phone is again reachable for few minutes. I captured the traffic on server side with tcpdump, unlike my asterisk server, freeswitch never send to the phone a keepalive packet ( SIP OPTIONS ) I added theses options in sip_profiles/internal.xml Please note the strange expiry delay. After being powered off, the phone stays registered. freeswitch at internal> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 24ef-c0a80101-5-1 at 192.168.0.10 User: 7082 at mydomain Contact: "" Agent: THOMSON ST2030 hw5 fw2.76 00-14-7F-00-4C-1A Status: Registered(UDP-NAT)(unknown) EXP(1907-10-11 18:28:34) EXPSECS(909113102) Ping-Status: Reachable Host: vpsxxxxxx.ovh.net IP: 86.201.x.x Port: 5090 Auth-User: 7082 Auth-Realm: mydomain MWI-Account: 7082 at mydomain Any idea ? Best regards From ahabiba at gmail.com Mon Jan 26 01:55:43 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Mon, 26 Jan 2015 01:55:43 +0300 Subject: [Freeswitch-users] Snmp Module is not loading In-Reply-To: <785DEB27-8C3D-4582-9B93-0EA973DD07A2@gmail.com> References: <785DEB27-8C3D-4582-9B93-0EA973DD07A2@gmail.com> Message-ID: <7390F618-6FAA-4BCE-BDE8-FE61D2A9AFCF@gmail.com> Dears, kindly I tried to use mod_snmp I compile it using make && make install, however when I tried to load it I got the below message:, your kind usual support will be appreciated: 2015-01-24 15:05:58.673046 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_snmp.so **/usr/lib/libnetsnmpagent.so.15: undefined symbol: netsnmp_register_null_context** Thanks, Ahmed Habiba. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/e87fa97e/attachment.html From msc at freeswitch.org Mon Jan 26 02:19:32 2015 From: msc at freeswitch.org (Michael Collins) Date: Sun, 25 Jan 2015 15:19:32 -0800 Subject: [Freeswitch-users] can I use Event_Socket_Outbound to only monitor? In-Reply-To: References: Message-ID: In event socket outbound the listening server controls the call. In your case you just want to get the events associated with that call? I've never actually tried it but maybe you could do something like: And then after the socket app answers have it issue the linger command, then transfer or execute_extension to send the call off to do something. Watch your socket connection for events and see happens. If you get it to work please let us know. :) -Michael On Sun, Jan 25, 2015 at 7:17 AM, Shai Perelman wrote: > > hi, I am using > https://wiki.freeswitch.org/wiki/Event_Socket_Outbound > my goal is to monitor what happens on the extension but without affecting > the normal behaviour. > > I have this dialplan. > > TagTypeDataOrder[image: add] > > condition username 9998 1[image: edit] > > [image: delete] > > action socket 109.65.149.22:9999 async 3[image: edit] > > [image: delete] > > > > I am calling *9664 (music on hold) > and the call is blocking while wating for commands from my side. > > I want it to answer automatticaly like it would do with out the socket > module, and just send me the events regarding that extension > > thanks > Shai > ITD Communications > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150125/2cd3da6f/attachment.html From msc at freeswitch.org Mon Jan 26 02:27:19 2015 From: msc at freeswitch.org (Michael Collins) Date: Sun, 25 Jan 2015 15:27:19 -0800 Subject: [Freeswitch-users] Playing file using lua before bridge In-Reply-To: References: Message-ID: Recommend you pastebin a full debug log of a call that exhibits this behavior. -MC On Fri, Jan 23, 2015 at 8:52 PM, John Nash wrote: > I am trying to play some native files (Pre encoded) using lua script just > after call is setup using Bridge (But before any answer). I need to use > "Proxy_Media" (Want media to go through freeswitch but no codec > translation). I tried using > "bridge_pre_execute_aleg_app" but I do not hear sound files being played > (Do not see any error message also). I have already tested lua script in > some other test extension and it works fine. > > > > > > data="bridge_pre_execute_aleg_app=lua"/> > data="bridge_pre_execute_aleg_data=tt.lua ${credit_time}"/> > > data="execute_on_answer=sched_hangup +${credit_time}" /> > data="sofia/$${domain}/${regex(${sip_req_uri}|^kb-(.+)$|%1)};fs_path=sip:${sip_received_ip}:${sip_received_port}"/> > > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150125/f9643b42/attachment.html From jdayola at spingine.com Mon Jan 26 06:10:42 2015 From: jdayola at spingine.com (jdayola at spingine.com) Date: Mon, 26 Jan 2015 11:10:42 +0800 Subject: [Freeswitch-users] callcenter mod rtmp cant receive calls... Message-ID: Hi! Im using mod callcenter. I have 3 agents on queue 600 and assigned them extensions 601,602 and 603. Everything was working fine when I tested it using xlite and zoiper. But on production we have our own softphone based on flash so we need to use rtmp. I already enabled rtmp but I don't receive calls on my extensions using rtmp. Like if I assign 601 and 602 to xlite and zoiper and 603 on our flash client only 601 and 602 can receive calls. Do I need to make a new dial plan if I use RTMP? Im still new to voip and Im using fusionpbx as my freeswitch gui. Regards, John D. From gb at cm.nl Mon Jan 26 09:51:46 2015 From: gb at cm.nl (Grant Bagdasarian) Date: Mon, 26 Jan 2015 06:51:46 +0000 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: References: Message-ID: Hello, I just extracted the tarball, installed a few missing dependencies and ran the configure command: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Friday, January 23, 2015 4:44 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring I'm inclined to believe you have something broken on your box here. I just grabbed the tarball and built it right on my Wheezy box... How exactly are you trying to build it from the tarball? On 1/23/15, 8:49 AM, "Grant Bagdasarian" wrote: lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 7.8 (wheezy) Release: 7.8 Codename: wheezy uname -a Linux HOSTNAME 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Friday, January 23, 2015 3:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring On what platform are you getting this error? Sent from my iPhone On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian wrote: Nevermind. I used the 1.4.15 tarball which contained this error. Now I've used the git clone master which works fine. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian Sent: Friday, January 23, 2015 11:51 AM To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) Subject: [Freeswitch-users] APR issue when configuring Hello, I'm getting the following error during configure. checking for APR... configure: error: the --with-apr parameter is incorrect. It must specify an install prefix, a build directory, or an apr-config file. configure: error: ./configure.gnu failed for libs/apr-util I'm running the following command to configure: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support Also, a lot of directories inside the libs require the configure script to be executable. I have to set this 1 by 1. Is there a way to disable apr? Or is it required by FS? Regards, Grant _________________________________________________________________________ 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/3103e144/attachment.html From bordmi at rarus.ru Mon Jan 26 10:40:32 2015 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Mon, 26 Jan 2015 11:40:32 +0400 Subject: [Freeswitch-users] mod_event_multicast bind interface question Message-ID: I have freeswitch installed on server with two etehrnet interfaces. My profiles are started on eth1, but multicast must be sent from eth0. Is it possible? And how? -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/dd46c4af/attachment.html From denis at ringme.ru Mon Jan 26 11:55:51 2015 From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=) Date: Mon, 26 Jan 2015 11:55:51 +0300 Subject: [Freeswitch-users] lua + aes|md5 Message-ID: <54C60117.7040204@ringme.ru> Hello. I need module md5 or aes for checksums, but modules from luarocks not work. Who can make luarocks version for freeswitch with 5.2 version? Or system RPM with 5.2 My OS is CentOS 6. From vipkilla at gmail.com Mon Jan 26 16:14:35 2015 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 26 Jan 2015 08:14:35 -0500 Subject: [Freeswitch-users] mod_snmp not loading In-Reply-To: <785DEB27-8C3D-4582-9B93-0EA973DD07A2@gmail.com> References: <785DEB27-8C3D-4582-9B93-0EA973DD07A2@gmail.com> Message-ID: What distro of linux are you using? Last I checked, mod_snmp had issues on centos because of a dependencie On Sat, Jan 24, 2015 at 9:05 AM, Ahmed Habiba wrote: > Dears, > > kindly I tried to use mod_snmp I compile it using make && make install, > however when I tried to load it I got the below message:, your kind usual > support will be appreciated: > > 2015-01-24 15:05:58.673046 [CRIT] switch_loadable_module.c:1447 Error > Loading module /usr/local/freeswitch/mod/mod_snmp.so > **/usr/lib/libnetsnmpagent.so.15: undefined symbol: > netsnmp_register_null_context** > > Thanks, > > Ahmed Habiba. > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/caabae3a/attachment.html From hkalyoncu at gmail.com Mon Jan 26 16:26:09 2015 From: hkalyoncu at gmail.com (huseyin kalyoncu) Date: Mon, 26 Jan 2015 15:26:09 +0200 Subject: [Freeswitch-users] inband dtmf Message-ID: hello first i want to thank the developers and contributors of this amazing product. we have been using freeswitch for almost 4 years without a major problem. i have a question regarding inband dtmf. we have receiving calls from telco using dtmf rfc2833. most of outgoing calls also dtmf rfc2833. but we have a new outgoing profile which is behind a firewall. we did not make a successful call with transport UDP. so we set the transport to TCP and now we have successful calls but the only problem is with dtmf. when we set the dtmf to rfc2833 for this profile, we saw that dtmf packets do not arrive correctly to outgoing destination. when we dig up the problem we realized that there is always a time skew on dtmf packets. for this reason we tried to set dtmf inband for this particular outgoing profile. to accomplish this we used start_dtmf_generate just before the bridge action. but this time no dtmf package arrive at destination. is this dtmf conversion (from rfc2833 to inband) even possible? what should be the correct configuration to achieve this? thanks huseyin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/cbbcb885/attachment-0001.html From dp.siddharth at eng.knowlarity.com Mon Jan 26 16:37:35 2015 From: dp.siddharth at eng.knowlarity.com (DP Siddharth) Date: Mon, 26 Jan 2015 19:07:35 +0530 Subject: [Freeswitch-users] Getting Redirecting number in Events (ISDN) In-Reply-To: References: Message-ID: Thanks , I will test it with 1.4. On Mon, Jan 26, 2015 at 3:29 AM, Michael Collins wrote: > I strongly recommend you spin up a FreeSWITCH 1.4 instance for testing. > The 1.2 branch has served its purpose and has been honorably discharged as > it were. If what you need gets added to the code base it will not be in 1.2. > > When you get 1.4 up and running be sure to route an ISDN call through the > info dialplan app just to make sure it's not already there and available > for your use. If it's available in a channel variable then it should be > available to add to events as needed. > > -Michael > > On Wed, Jan 21, 2015 at 3:58 AM, DP Siddharth < > dp.siddharth at eng.knowlarity.com> wrote: > >> Hi, >> >> I am using ISDN (PRI) for handling calls. In a use case we need >> Redirecting number as part of freeswitch events. I verified various events >> but we get only caller/called. I was looking into the code & found for SS7 >> this is available but not for ISDN, just need confirmation if any way >> possible to get this. I am using 1.2.24 (1.2.stable) freeswitch for testing. >> >> following is ISDN SETUP message for reference: >> >> Prot Disc:Q.931/I.451 (0x08) >> Call Ref:242E (Originating side) >> Type:SETUP (0x5) >> Sending complete: >> Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64 >> Kbit/s(16) L1Prot:G.711 A-Law(3) >> Channel Id:No:13 Type:B-chans(3) Preferred/Implicit >> Calling Party Number:8860128000(l:10) plan:isdn(1) >> type:national(2)scr:network, provided(3) pres:allowed(0) >> Called Party Number:01725218700(l:11) plan:isdn(1) type:subscriber(4) >> Redirecting Number:53703417(l:8) plan:isdn(1) >> type:national(2)scr:user, not screened(0) pres:restricted(1)reason:Call >> forwarding unconditional(15) >> Redirecting Number:53703417(l:8) plan:isdn(1) type:national(2)scr:user, >> not screened(0) pres:restricted(1)reason:Call forwarding unconditional(15) >> [ 08 02 24 2e 05 a1 04 03 80 90 a3 18 03 a1 83 8d 6c 0c 21 83 38 38 36 >> 30 31 32 38 30 30 30 70 0c >> c1 30 31 37 32 35 32 31 38 37 30 30 74 0d 21 20 8f 39 36 35 33 37 30 >> 33 34 31 37 74 0d 21 20 8f >> 39 36 35 33 37 30 33 34 31 37 ] >> >> >> -- >> Thanks & Regards, >> D P Siddharth >> Director (Platform) >> Knowlarity Communications >> Ph: +919999115231 >> dp.siddharth at eng.knowlarity.com >> >> *"Come together to build a lasting world-class cloud telephony company >> that helps businesses grow"* >> >> _________________________________________________________________________ >> 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 > -- Thanks & Regards, D P Siddharth Director (Platform) Knowlarity Communications Ph: +919999115231 dp.siddharth at eng.knowlarity.com *"Come together to build a lasting world-class cloud telephony company that helps businesses grow"* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/cc615ae4/attachment.html From asilva at wirelessmundi.com Mon Jan 26 17:02:02 2015 From: asilva at wirelessmundi.com (Antonio Silva) Date: Mon, 26 Jan 2015 15:02:02 +0100 Subject: [Freeswitch-users] lua + aes|md5 In-Reply-To: <54C60117.7040204@ringme.ru> References: <54C60117.7040204@ringme.ru> Message-ID: <54C648DA.8070804@wirelessmundi.com> Are you using esl? I make it work compiling lua esl module adding to makefile the option LOCAL_CFLAGS=-I/usr/include/lua5.1 On 01/26/2015 09:55 AM, ????? wrote: > Hello. > > I need module md5 or aes for checksums, but modules from luarocks not work. > Who can make luarocks version for freeswitch with 5.2 version? Or system > RPM with 5.2 > > My OS is CentOS 6. > > _________________________________________________________________________ > 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 -- --- Ant?nio Silva From mike at jerris.com Mon Jan 26 18:21:03 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 26 Jan 2015 10:21:03 -0500 Subject: [Freeswitch-users] Fork( ) and Exec ( ) functions In-Reply-To: References: <6053000.JvnxKzh3j9@sos> <1C00AF43-07EF-4A86-8CB0-286C49617B5F@jerris.com> Message-ID: most likely this is going to require you to make code changes, not just makefile changes. > On Jan 23, 2015, at 10:33 PM, Abdul Hakeem wrote: > > Thanks guys. > I?ll do battle with the makefile next week & see how it goes. > ? <> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Sent: Monday, January 19, 2015 5:12 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Fork( ) and Exec ( ) functions > > We use threads by default in the switch system. There is a setting to use fork instead but it's disabled by default. The system syscall uses fork and exec on its own so maybe you just need to make sure ifdefs properly disable any such functionality. > > On Monday, January 19, 2015, Michael Jerris > wrote: > You can also look at how we have addressed this issue for windows. The places I know of that use these functions. switch_system (used for executing external commands and sending emails), and daemonizing FreeSWITCH. > > > > On Jan 18, 2015, at 3:48 PM, Sergey Okhapkin > wrote: > > > > Usually fork-less OSes provide spawn() family of syscalls to execute a new > > process. > > > > On Sunday 18 January 2015 20:32:02 Abdul Hakeem wrote: > >> Hello again, > >> The reason I ask is because I am trying to port FS to OSv. > >> Unfortunately, Fork ( ) is unsupported in OSv. > >> Regards, > >> Abdul Hakeem > >> > >> From: freeswitch-users-bounces at lists.freeswitch.org <> > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org <>] On Behalf Of Moises > >> Silva Sent: Tuesday, January 13, 2015 5:11 PM > >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] Fork( ) and Exec ( ) functions > >> > >> On Tue, Jan 13, 2015 at 8:11 AM, Abdul Hakeem > wrote: > >> Hello, > >> > >> I understand FS makes system calls for sending mails, voicemail and fax. > >> Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on > >> system calls & also, a list of functions which require FS to make system > >> calls ? > >> > >> Not making much sense here. Everything in FS relies heavily on system calls > >> as it's a multi-threaded-I/O-driven system for the most part. I think > >> you're asking the wrong question. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/9b09fc8b/attachment.html From 0x6e6562 at gmail.com Mon Jan 26 17:01:20 2015 From: 0x6e6562 at gmail.com (Ben Hood) Date: Mon, 26 Jan 2015 14:01:20 +0000 Subject: [Freeswitch-users] Configuring request parameters with mod_xml_curl Message-ID: Hi, I'm using mod_xml_curl with FS 1.4.15 -1 and I was wondering if there is a way to configure the request parameters in FS to only send a subset of all available request parameters. My use case is that my web service responding to FS only uses two out of the many possible parameters that FS sends. Over time this is filling up the debug logs with lots of noise, so I was hoping to trim this down. Does anybody know of a way to do this? Thanks, Ben From ahabiba at gmail.com Mon Jan 26 18:30:50 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Mon, 26 Jan 2015 18:30:50 +0300 Subject: [Freeswitch-users] mod_snmp not loading In-Reply-To: References: Message-ID: Thank you really Vik here is below my linux version: Distributor ID: Ubuntu Description: Ubuntu 12.04.5 LTS Release: 12.04 Codename: precise From: Vik Killa > To: FreeSWITCH Users Help > Date: January 26, 2015 at 4:14:35 PM GMT+3 Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_snmp not loading What distro of linux are you using? Last I checked, mod_snmp had issues on centos because of a dependencie On Sat, Jan 24, 2015 at 9:05 AM, Ahmed Habiba > wrote: Dears, kindly I tried to use mod_snmp I compile it using make && make install, however when I tried to load it I got the below message:, your kind usual support will be appreciated: 2015-01-24 15:05:58.673046 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_snmp.so **/usr/lib/libnetsnmpagent.so.15: undefined symbol: netsnmp_register_null_context** Thanks, Ahmed Habiba. _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/5e5d4a2c/attachment.html From vipkilla at gmail.com Mon Jan 26 18:31:34 2015 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 26 Jan 2015 10:31:34 -0500 Subject: [Freeswitch-users] Configuring request parameters with mod_xml_curl In-Reply-To: References: Message-ID: I don't think you can control what FS sends as request params. Figure out how to trim your logs better. On Mon, Jan 26, 2015 at 9:01 AM, Ben Hood <0x6e6562 at gmail.com> wrote: > Hi, > > I'm using mod_xml_curl with FS 1.4.15 -1 and I was wondering if there > is a way to configure the request parameters in FS to only send a > subset of all available request parameters. > > My use case is that my web service responding to FS only uses two out > of the many possible parameters that FS sends. > > Over time this is filling up the debug logs with lots of noise, so I > was hoping to trim this down. > > Does anybody know of a way to do this? > > Thanks, > > Ben > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/99fdf367/attachment.html From adrottenberg at gmail.com Mon Jan 26 18:57:58 2015 From: adrottenberg at gmail.com (Duvid Rottenberg) Date: Mon, 26 Jan 2015 10:57:58 -0500 Subject: [Freeswitch-users] can I use Event_Socket_Outbound to only monitor? In-Reply-To: References: Message-ID: I am pretty sure that with outbound socket as long as you don't send any commands back to Freeswitch, it will just execute whatever is in the xml dialplan. Thanks, Duvid Rottenberg On Sun, Jan 25, 2015 at 6:19 PM, Michael Collins wrote: > In event socket outbound the listening server controls the call. In your > case you just want to get the events associated with that call? I've never > actually tried it but maybe you could do something like: > > > > > And then after the socket app answers have it issue the linger command, > then transfer or execute_extension to send the call off to do something. > Watch your socket connection for events and see happens. If you get it to > work please let us know. :) > > -Michael > > > > On Sun, Jan 25, 2015 at 7:17 AM, Shai Perelman wrote: > >> >> hi, I am using >> https://wiki.freeswitch.org/wiki/Event_Socket_Outbound >> my goal is to monitor what happens on the extension but without affecting >> the normal behaviour. >> >> I have this dialplan. >> >> TagTypeDataOrder[image: add] >> >> condition username 9998 1[image: edit] >> >> [image: delete] >> >> action socket 109.65.149.22:9999 async 3[image: edit] >> >> [image: delete] >> >> >> >> I am calling *9664 (music on hold) >> and the call is blocking while wating for commands from my side. >> >> I want it to answer automatticaly like it would do with out the socket >> module, and just send me the events regarding that extension >> >> thanks >> Shai >> ITD Communications >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/07bd0f28/attachment-0001.html From 0x6e6562 at gmail.com Mon Jan 26 20:09:01 2015 From: 0x6e6562 at gmail.com (Ben Hood) Date: Mon, 26 Jan 2015 17:09:01 +0000 Subject: [Freeswitch-users] Configuring request parameters with mod_xml_curl In-Reply-To: References: Message-ID: Thanks for the heads up. On Mon, Jan 26, 2015 at 3:31 PM, Vik Killa wrote: > I don't think you can control what FS sends as request params. > Figure out how to trim your logs better. From nbhatti at gmail.com Mon Jan 26 21:49:09 2015 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Mon, 26 Jan 2015 21:49:09 +0300 Subject: [Freeswitch-users] Configuring request parameters with mod_xml_curl In-Reply-To: References: Message-ID: ?Try enable-post-var in xml_curl, it will send only the variable you define there one per line. ? Thanks, Muhammad Naseer Bhatti From:?Ben Hood <0x6e6562 at gmail.com> Reply:?FreeSWITCH Users Help > Date:?January 26, 2015 at 8:10:24 PM To:?FreeSWITCH Users Help > Subject:? Re: [Freeswitch-users] Configuring request parameters with mod_xml_curl Thanks for the heads up. On Mon, Jan 26, 2015 at 3:31 PM, Vik Killa wrote: > I don't think you can control what FS sends as request params. > Figure out how to trim your logs better. _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/15850343/attachment.html From 0x6e6562 at gmail.com Mon Jan 26 23:00:04 2015 From: 0x6e6562 at gmail.com (Ben Hood) Date: Mon, 26 Jan 2015 20:00:04 +0000 Subject: [Freeswitch-users] Configuring request parameters with mod_xml_curl In-Reply-To: References: Message-ID: Aha - does enable-post-var also work when you're using the GET method, or do you also have to use POST to enable this? On Mon, Jan 26, 2015 at 6:49 PM, Muhammad Naseer Bhatti wrote: > > Try enable-post-var in xml_curl, it will send only the variable you define > there one per line. From mike at jerris.com Mon Jan 26 23:07:43 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 26 Jan 2015 14:07:43 -0600 Subject: [Freeswitch-users] lua + aes|md5 In-Reply-To: <54C648DA.8070804@wirelessmundi.com> References: <54C60117.7040204@ringme.ru> <54C648DA.8070804@wirelessmundi.com> Message-ID: In master I just added the --enable-system-lua configure flag that makes mod_lua and esl lua both use the system lib, and fixed a number of issues using different versions of lua. This may help you out. Please note, in 1.6 we will remove the need for the configure flag, and this will always use system libs. Mike > On Jan 26, 2015, at 8:02 AM, Antonio Silva wrote: > > Are you using esl? > > I make it work compiling lua esl module adding to makefile the option > LOCAL_CFLAGS=-I/usr/include/lua5.1 > > > > On 01/26/2015 09:55 AM, ????? wrote: >> Hello. >> >> I need module md5 or aes for checksums, but modules from luarocks not work. >> Who can make luarocks version for freeswitch with 5.2 version? Or system >> RPM with 5.2 >> >> My OS is CentOS 6. >> >> _________________________________________________________________________ >> 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 > > -- > --- > Ant?nio Silva > > > _________________________________________________________________________ > 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 From mike at jerris.com Mon Jan 26 23:21:42 2015 From: mike at jerris.com (Michael Jerris) Date: Mon, 26 Jan 2015 14:21:42 -0600 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: References: Message-ID: <31690C3C-FF53-4BD3-B738-352E71396CCA@jerris.com> is /usr/src/freeswitch your source directory too? Its a bit weird to be specifying that as prefix, maybe its an issue when specifying a prefix as the same as your source dir? > On Jan 26, 2015, at 12:51 AM, Grant Bagdasarian wrote: > > Hello, > > I just extracted the tarball, installed a few missing dependencies and ran the configure command: > ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice > Sent: Friday, January 23, 2015 4:44 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] APR issue when configuring > > I?m inclined to believe you have something broken on your box here. I just grabbed the tarball and built it right on my Wheezy box... How exactly are you trying to build it from the tarball? > > > On 1/23/15, 8:49 AM, "Grant Bagdasarian" > wrote: > > lsb_release -a > No LSB modules are available. > Distributor ID: Debian > Description: Debian GNU/Linux 7.8 (wheezy) > Release: 7.8 > Codename: wheezy > > uname -a > Linux HOSTNAME 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Ken Rice > Sent: Friday, January 23, 2015 3:04 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] APR issue when configuring > > > On what platform are you getting this error? > > > > > Sent from my iPhone > > > On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian > wrote: > > Nevermind. > I used the 1.4.15 tarball which contained this error. > > Now I?ve used the git clone master which works fine. > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Grant Bagdasarian > Sent: Friday, January 23, 2015 11:51 AM > To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org ) > Subject: [Freeswitch-users] APR issue when configuring > > Hello, > > I?m getting the following error during configure. > > checking for APR... configure: error: the --with-apr parameter is incorrect. It must specify an install prefix, a build directory, or an apr-config file. > configure: error: ./configure.gnu failed for libs/apr-util > > I?m running the following command to configure: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support > > Also, a lot of directories inside the libs require the configure script to be executable. I have to set this 1 by 1. > > Is there a way to disable apr? Or is it required by FS? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/259d0d8b/attachment-0001.html From steveayre at gmail.com Mon Jan 26 23:23:56 2015 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 26 Jan 2015 20:23:56 +0000 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: <31690C3C-FF53-4BD3-B738-352E71396CCA@jerris.com> References: <31690C3C-FF53-4BD3-B738-352E71396CCA@jerris.com> Message-ID: Also is that -prefix or --prefix you're using? On 26 January 2015 at 20:21, Michael Jerris wrote: > is /usr/src/freeswitch your source directory too? Its a bit weird to be > specifying that as prefix, maybe its an issue when specifying a prefix as > the same as your source dir? > > On Jan 26, 2015, at 12:51 AM, Grant Bagdasarian wrote: > > Hello, > > I just extracted the tarball, installed a few missing dependencies and ran > the configure command: > ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Ken Rice > *Sent:* Friday, January 23, 2015 4:44 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] APR issue when configuring > > > I?m inclined to believe you have something broken on your box here. I just > grabbed the tarball and built it right on my Wheezy box... How exactly are > you trying to build it from the tarball? > > > On 1/23/15, 8:49 AM, "Grant Bagdasarian" wrote: > > *lsb_release -a*No LSB modules are available. > Distributor ID: Debian > Description: Debian GNU/Linux 7.8 (wheezy) > Release: 7.8 > Codename: wheezy > > > *uname -a*Linux HOSTNAME 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 > GNU/Linux > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Ken Rice > *Sent:* Friday, January 23, 2015 3:04 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] APR issue when configuring > > > On what platform are you getting this error? > > > > > Sent from my iPhone > > > On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian wrote: > > Nevermind. > I used the 1.4.15 tarball which contained this error. > > Now I?ve used the git clone master which works fine. > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Grant > Bagdasarian > *Sent:* Friday, January 23, 2015 11:51 AM > *To:* FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) > *Subject:* [Freeswitch-users] APR issue when configuring > > Hello, > > I?m getting the following error during configure. > > checking for APR... configure: error: the --with-apr parameter is > incorrect. It must specify an install prefix, a build directory, or an > apr-config file. > configure: error: ./configure.gnu failed for libs/apr-util > > I?m running the following command to configure: ./configure > -prefix=/usr/src/freeswitch/ --enable-core-odbc-support > > Also, a lot of directories inside the libs require the configure script to > be executable. I have to set this 1 by 1. > > Is there a way to disable apr? Or is it required by FS? > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/e7123fb7/attachment.html From krice at freeswitch.org Mon Jan 26 23:54:25 2015 From: krice at freeswitch.org (Ken Rice) Date: Mon, 26 Jan 2015 20:54:25 +0000 Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) January 17th-23rd Message-ID: <54c6a9818defc_e779c3532088927@ip-10-5-130-87.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1thCTTe FreeSWITCH Week in Review (Master Branch) January 17th-23rd Hello, again. This week in the FreeSWITCH master branch we had 21 commits. There were quite a few commits this week geared toward new features, including: adding switch_cache_db_create_schema() to test for SCF_AUTO_SCHEMAS flag, enable nat mode for verto when ext-rtp-ip is set, auto urlencode user portion of sip uri, uuid_media support for SRTP Call, and adding ?enable-sytem-lua configure arg to allow building mod_lua against system lua and allow mod_lua to build against lua 5.1 or 5.2. These lua changes mean enabling the usage of system lua in preparation for removing the lua library from tree. New features that were added: 07d09b7 FS-7180 Add ?enable-sytem-lua configure arg to allow building mod_lua against system lua and allow mod_lua to build against lua 5.1 or 5.2 [Jira: http://ift.tt/1z2ZB0V] 749ced5 FS-7180 More work toward add ?enable-sytem-lua configure arg to allow building mod_lua against system lua and allow mod_lua to build against lua 5.1 or 5.2 [Jira: http://ift.tt/1z2ZB0V] 1d361b6 FS-7180 Let esl lua module build against lua 5.1 or 5.2 (requires newer swig) [Jira: http://ift.tt/1z2ZB0V] 861961b FS-7180 When using system lua, properly link against reanmed library versions on debian for mod_lua [Jira: http://ift.tt/1z2ZB0V] 8fc19a6 FS-7180 Properly build esl luamod when not using the ?enable-system-lua configure arg [Jira: http://ift.tt/1z2ZB0V] 01dcb74 FS-7187 Add switch_cache_db_create_schema() to test for SCF_AUTO_SCHEMAS flag [Jira: http://ift.tt/1thCUXs] 1710214 Enable nat mode for verto when ext-rtp-ip is set 76370f4 Auto urlencode user portion of sip uri 83dd941 FS-7166 Uuid_media support for SRTP Calls [Jira: http://ift.tt/1z2ZDWC] 95a8efb Up the ice failover val to 3 sec Improvements in cross platform build supports: 90ab1d1 Fix cent5 build The following bugs were squashed: b744484 FS-7173 Work toward fixing a recording issues after an external call is transferred to an internal extension [Jira: http://ift.tt/1thCUXu] 15a7ff2 Fix hash dump gdb function f770b31 FS-7106 Fix a concurrency issue in caching of files from http urls [Jira: http://ift.tt/1BEamXd] 062ddcf FS-7174 Make sure not to leave any sessions readlocked in bridge_early_media=true in case one in the middle of the list is abandoned [Jira: http://ift.tt/1thCVdK] 90d3cb6 Fix media reload on verto and sip re-invites fc93895 FS-7173 More work toward fixing recording issues after an external call is transferred to an internal extension [Jira: http://ift.tt/1thCUXu] b37d071 FS-7186 Fix for enterprise bridging/originate issue [Jira: http://ift.tt/1z2ZDWF] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/578edfb9/attachment.html From phenix at vfemail.net Mon Jan 26 23:54:37 2015 From: phenix at vfemail.net (Tanguy) Date: Mon, 26 Jan 2015 21:54:37 +0100 Subject: [Freeswitch-users] Freeswitch don't send keepalive In-Reply-To: <54C56372.5070708@vfemail.net> References: <54C56372.5070708@vfemail.net> Message-ID: <54C6A98D.5010708@vfemail.net> Hello, after a reboot my phones are registering properly, the " sofia status profile internal reg " command display a correct expiry time and freeswitch send "SIP options" every 30 secs. But why did i need to reboot ? NB: Before rebooting, I did not try to restart the freeswitch service only, i only used reloadxml Regards On -10/01/-28163 20:59, Tanguy wrote: > Hello > > I'm a asterisk user, and i would like to try freeswitch for having a > real multi tenant system. I configured FS with fusionpbx. My FS is > installed on a VPS ( has a public IP ), my extensions are behind a nat > router. > > After few minutes my phone became unreachable for inbound call. After > a outbound call, the phone is again reachable for few minutes. > > I captured the traffic on server side with tcpdump, unlike my asterisk > server, freeswitch never send to the phone a keepalive packet ( SIP > OPTIONS ) > > I added theses options in sip_profiles/internal.xml > > > > > > > > > > > > > Please note the strange expiry delay. After being powered off, the > phone stays registered. > > freeswitch at internal> sofia status profile internal reg > > Registrations: > ================================================================================================= > > Call-ID: 24ef-c0a80101-5-1 at 192.168.0.10 > User: 7082 at mydomain > Contact: "" > > Agent: THOMSON ST2030 hw5 fw2.76 00-14-7F-00-4C-1A > Status: Registered(UDP-NAT)(unknown) EXP(1907-10-11 18:28:34) > EXPSECS(909113102) > Ping-Status: Reachable > Host: vpsxxxxxx.ovh.net > IP: 86.201.x.x > Port: 5090 > Auth-User: 7082 > Auth-Realm: mydomain > MWI-Account: 7082 at mydomain > > Any idea ? > > Best regards > > > From aqsyounas at gmail.com Tue Jan 27 00:45:26 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 27 Jan 2015 02:45:26 +0500 Subject: [Freeswitch-users] how to transfer all conference members to an extension on stop-talking event. Message-ID: Hi, list. I need to transfer all conference members to some other extension whenever there is any stop-talking event. But don't know how can i do so? I am using python script as inbound connection to listening to events. Whenever there is stop-talking event I need to transfer all conference members to another dialplan extension. Your help would be much appreciated. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/ad334e98/attachment-0001.html From 0x6e6562 at gmail.com Tue Jan 27 00:51:29 2015 From: 0x6e6562 at gmail.com (Ben Hood) Date: Mon, 26 Jan 2015 21:51:29 +0000 Subject: [Freeswitch-users] Configuring request parameters with mod_xml_curl In-Reply-To: References: Message-ID: On Mon, Jan 26, 2015 at 8:00 PM, Ben Hood <0x6e6562 at gmail.com> wrote: > Aha - does enable-post-var also work when you're using the GET method, > or do you also have to use POST to enable this? I found out by trial and error that by specifying a request parameter per enable-post-var entry, I could effectively cherry pick the request even using the GET method. For example, adding these two lines: Results in the following request: data: [hostname=x§ion=dialplan&tag_name=&key_name=&key_value=&variable_sip_from_user_stripped=1234567890&variable_sip_to_user=0987654321] So thank you very much for the tip off. Do you happen to know how I could even get rid of the empty tag_name, key_name and key_value request parameters? From max at nysolutions.com Tue Jan 27 01:04:49 2015 From: max at nysolutions.com (Moishe Grunstein) Date: Mon, 26 Jan 2015 22:04:49 +0000 Subject: [Freeswitch-users] Freeswitch don't send keepalive In-Reply-To: <54C6A98D.5010708@vfemail.net> References: <54C56372.5070708@vfemail.net> <54C6A98D.5010708@vfemail.net> Message-ID: Might be a bug in the phone's firmware. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tanguy Sent: Monday, January 26, 2015 3:55 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch don't send keepalive Hello, after a reboot my phones are registering properly, the " sofia status profile internal reg " command display a correct expiry time and freeswitch send "SIP options" every 30 secs. But why did i need to reboot ? NB: Before rebooting, I did not try to restart the freeswitch service only, i only used reloadxml Regards On -10/01/-28163 20:59, Tanguy wrote: > Hello > > I'm a asterisk user, and i would like to try freeswitch for having a > real multi tenant system. I configured FS with fusionpbx. My FS is > installed on a VPS ( has a public IP ), my extensions are behind a nat > router. > > After few minutes my phone became unreachable for inbound call. After > a outbound call, the phone is again reachable for few minutes. > > I captured the traffic on server side with tcpdump, unlike my asterisk > server, freeswitch never send to the phone a keepalive packet ( SIP > OPTIONS ) > > I added theses options in sip_profiles/internal.xml > > name="enable-timer" value="false"/> name="NDLB-received-in-nat-reg-contact" value="true"/> name="NDLB-force-rport" value="true"/> name="NDLB-broken-auth-hash" value="true"/> name="sip-force-expires" value="240"/> name="all-reg-options-ping" value="true"/> name="registration-thread-frequency" value="30"/> name="unregister-on-options-fail" value="true"/> name="nat-options-ping" value="true"/> > > Please note the strange expiry delay. After being powered off, the > phone stays registered. > > freeswitch at internal> sofia status profile internal reg > > Registrations: > ====================================================================== > =========================== > > Call-ID: 24ef-c0a80101-5-1 at 192.168.0.10 > User: 7082 at mydomain > Contact: "" > > Agent: THOMSON ST2030 hw5 fw2.76 00-14-7F-00-4C-1A > Status: Registered(UDP-NAT)(unknown) EXP(1907-10-11 18:28:34) > EXPSECS(909113102) > Ping-Status: Reachable > Host: vpsxxxxxx.ovh.net > IP: 86.201.x.x > Port: 5090 > Auth-User: 7082 > Auth-Realm: mydomain > MWI-Account: 7082 at mydomain > > Any idea ? > > Best regards > > > _________________________________________________________________________ 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 From jpyle at fidelityvoice.com Tue Jan 27 03:04:29 2015 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 26 Jan 2015 19:04:29 -0500 Subject: [Freeswitch-users] Port changes in multiple 183s cause no audio after 200 Message-ID: Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit). The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with. The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example: - Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183). In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome! - Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/ffa90dc3/attachment.html From kamil.nigmatullin at gmail.com Tue Jan 27 06:43:38 2015 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Tue, 27 Jan 2015 09:43:38 +0600 Subject: [Freeswitch-users] can I use Event_Socket_Outbound to only monitor? In-Reply-To: References: Message-ID: Yes if you want monitor and catch events you need to use inbound socket 26 ???. 2015 ?. 21:58 ???????????? "Duvid Rottenberg" < adrottenberg at gmail.com> ???????: > I am pretty sure that with outbound socket as long as you don't send any > commands back to Freeswitch, it will just execute whatever is in the xml > dialplan. > > Thanks, > Duvid Rottenberg > > On Sun, Jan 25, 2015 at 6:19 PM, Michael Collins > wrote: > >> In event socket outbound the listening server controls the call. In your >> case you just want to get the events associated with that call? I've never >> actually tried it but maybe you could do something like: >> >> >> >> >> And then after the socket app answers have it issue the linger command, >> then transfer or execute_extension to send the call off to do something. >> Watch your socket connection for events and see happens. If you get it to >> work please let us know. :) >> >> -Michael >> >> >> >> On Sun, Jan 25, 2015 at 7:17 AM, Shai Perelman wrote: >> >>> >>> hi, I am using >>> https://wiki.freeswitch.org/wiki/Event_Socket_Outbound >>> my goal is to monitor what happens on the extension but without >>> affecting the normal behaviour. >>> >>> I have this dialplan. >>> >>> TagTypeDataOrder[image: add] >>> >>> condition username 9998 1[image: edit] >>> >>> [image: delete] >>> >>> action socket 109.65.149.22:9999 async 3[image: edit] >>> >>> [image: delete] >>> >>> >>> >>> I am calling *9664 (music on hold) >>> and the call is blocking while wating for commands from my side. >>> >>> I want it to answer automatticaly like it would do with out the socket >>> module, and just send me the events regarding that extension >>> >>> thanks >>> Shai >>> ITD Communications >>> >>> _________________________________________________________________________ >>> 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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/39641768/attachment-0001.html From msc at freeswitch.org Tue Jan 27 06:58:20 2015 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Jan 2015 19:58:20 -0800 Subject: [Freeswitch-users] how to transfer all conference members to an extension on stop-talking event. In-Reply-To: References: Message-ID: You can use the "conference list" API to get a list of all uuids in the conference and then use uuid_transfer on each of them. -MC On Mon, Jan 26, 2015 at 1:45 PM, Aqs Younas wrote: > Hi, list. > > I need to transfer all conference members to some other extension whenever > there is any stop-talking event. But don't know how can i do so? > > I am using python script as inbound connection to listening to events. > Whenever there is stop-talking event I need to transfer all conference > members to another dialplan extension. > > Your help would be much appreciated. > > Thanks. > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/2d29dac0/attachment.html From msc at freeswitch.org Tue Jan 27 06:59:40 2015 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Jan 2015 19:59:40 -0800 Subject: [Freeswitch-users] can I use Event_Socket_Outbound to only monitor? In-Reply-To: References: Message-ID: Using the socket app with the "async" option will cause that channel's events to be sent. -MC On Mon, Jan 26, 2015 at 7:43 PM, Kamil Nigmatullin < kamil.nigmatullin at gmail.com> wrote: > Yes if you want monitor and catch events you need to use inbound socket > 26 ???. 2015 ?. 21:58 ???????????? "Duvid Rottenberg" < > adrottenberg at gmail.com> ???????: > >> I am pretty sure that with outbound socket as long as you don't send any >> commands back to Freeswitch, it will just execute whatever is in the xml >> dialplan. >> >> Thanks, >> Duvid Rottenberg >> >> On Sun, Jan 25, 2015 at 6:19 PM, Michael Collins >> wrote: >> >>> In event socket outbound the listening server controls the call. In your >>> case you just want to get the events associated with that call? I've never >>> actually tried it but maybe you could do something like: >>> >>> >>> >>> >>> And then after the socket app answers have it issue the linger command, >>> then transfer or execute_extension to send the call off to do something. >>> Watch your socket connection for events and see happens. If you get it to >>> work please let us know. :) >>> >>> -Michael >>> >>> >>> >>> On Sun, Jan 25, 2015 at 7:17 AM, Shai Perelman >>> wrote: >>> >>>> >>>> hi, I am using >>>> https://wiki.freeswitch.org/wiki/Event_Socket_Outbound >>>> my goal is to monitor what happens on the extension but without >>>> affecting the normal behaviour. >>>> >>>> I have this dialplan. >>>> >>>> TagTypeDataOrder[image: add] >>>> >>>> condition username 9998 1[image: edit] >>>> >>>> [image: delete] >>>> >>>> action socket 109.65.149.22:9999 async 3[image: edit] >>>> >>>> [image: delete] >>>> >>>> >>>> >>>> I am calling *9664 (music on hold) >>>> and the call is blocking while wating for commands from my side. >>>> >>>> I want it to answer automatticaly like it would do with out the socket >>>> module, and just send me the events regarding that extension >>>> >>>> thanks >>>> Shai >>>> ITD Communications >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150126/33e2f8c9/attachment.html From gb at cm.nl Tue Jan 27 10:14:20 2015 From: gb at cm.nl (Grant Bagdasarian) Date: Tue, 27 Jan 2015 07:14:20 +0000 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: References: <31690C3C-FF53-4BD3-B738-352E71396CCA@jerris.com> Message-ID: Its ?prefix with a single -. Yes, the directory is /usr/src/freeswitch. I always use this parameter to install all the files in the same directory, else it installs in /usr/local/freeswitch I believe. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Monday, January 26, 2015 9:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring Also is that -prefix or --prefix you're using? On 26 January 2015 at 20:21, Michael Jerris > wrote: is /usr/src/freeswitch your source directory too? Its a bit weird to be specifying that as prefix, maybe its an issue when specifying a prefix as the same as your source dir? On Jan 26, 2015, at 12:51 AM, Grant Bagdasarian > wrote: Hello, I just extracted the tarball, installed a few missing dependencies and ran the configure command: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Friday, January 23, 2015 4:44 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring I?m inclined to believe you have something broken on your box here. I just grabbed the tarball and built it right on my Wheezy box... How exactly are you trying to build it from the tarball? On 1/23/15, 8:49 AM, "Grant Bagdasarian" > wrote: lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 7.8 (wheezy) Release: 7.8 Codename: wheezy uname -a Linux HOSTNAME 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Friday, January 23, 2015 3:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring On what platform are you getting this error? Sent from my iPhone On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian > wrote: Nevermind. I used the 1.4.15 tarball which contained this error. Now I?ve used the git clone master which works fine. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian Sent: Friday, January 23, 2015 11:51 AM To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) Subject: [Freeswitch-users] APR issue when configuring Hello, I?m getting the following error during configure. checking for APR... configure: error: the --with-apr parameter is incorrect. It must specify an install prefix, a build directory, or an apr-config file. configure: error: ./configure.gnu failed for libs/apr-util I?m running the following command to configure: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support Also, a lot of directories inside the libs require the configure script to be executable. I have to set this 1 by 1. Is there a way to disable apr? Or is it required by FS? _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/96cdd540/attachment-0001.html From krice at freeswitch.org Tue Jan 27 11:11:25 2015 From: krice at freeswitch.org (Ken Rice) Date: Tue, 27 Jan 2015 02:11:25 -0600 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: Message-ID: That should be --prefix (2 - in front of prefix) ... -prefix doesn?t really do anything On 1/27/15, 1:14 AM, "Grant Bagdasarian" wrote: > Its ?prefix with a single -. > > Yes, the directory is /usr/src/freeswitch. I always use this parameter to > install all the files in the same directory, else it installs in > /usr/local/freeswitch I believe. > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven > Ayre > Sent: Monday, January 26, 2015 9:24 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] APR issue when configuring > > > Also is that -prefix or --prefix you're using? > > > > On 26 January 2015 at 20:21, Michael Jerris wrote: > > is /usr/src/freeswitch your source directory too? Its a bit weird to be > specifying that as prefix, maybe its an issue when specifying a prefix as the > same as your source dir? > > >> >> On Jan 26, 2015, at 12:51 AM, Grant Bagdasarian wrote: >> >> >> Hello, >> >> >> >> I just extracted the tarball, installed a few missing dependencies and ran >> the configure command: >> >> ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support >> >> >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice >> Sent: Friday, January 23, 2015 4:44 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] APR issue when configuring >> >> >> I?m inclined to believe you have something broken on your box here. I just >> grabbed the tarball and built it right on my Wheezy box... How exactly are >> you trying to build it from the tarball? >> >> >> On 1/23/15, 8:49 AM, "Grant Bagdasarian" wrote: >> >> lsb_release -a >> No LSB modules are available. >> Distributor ID: Debian >> Description: Debian GNU/Linux 7.8 (wheezy) >> Release: 7.8 >> Codename: wheezy >> >> uname -a >> Linux HOSTNAME 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Ken >> Rice >> Sent: Friday, January 23, 2015 3:04 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] APR issue when configuring >> >> >> On what platform are you getting this error? >> >> >> >> >> Sent from my iPhone >> >> >> On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian wrote: >> >> >> Nevermind. >> I used the 1.4.15 tarball which contained this error. >> >> Now I?ve used the git clone master which works fine. >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Grant >> Bagdasarian >> Sent: Friday, January 23, 2015 11:51 AM >> To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) >> Subject: [Freeswitch-users] APR issue when configuring >> >> Hello, >> >> I?m getting the following error during configure. >> >> checking for APR... configure: error: the --with-apr parameter is incorrect. >> It must specify an install prefix, a build directory, or an apr-config file. >> configure: error: ./configure.gnu failed for libs/apr-util >> >> I?m running the following command to configure: ./configure >> -prefix=/usr/src/freeswitch/ --enable-core-odbc-support >> >> Also, a lot of directories inside the libs require the configure script to be >> executable. I have to set this 1 by 1. >> >> Is there a way to disable apr? Or is it required by FS? > > > _________________________________________________________________________ > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/51908843/attachment.html From nbhatti at gmail.com Tue Jan 27 11:55:21 2015 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 27 Jan 2015 11:55:21 +0300 Subject: [Freeswitch-users] Configuring request parameters with mod_xml_curl In-Reply-To: References: Message-ID: <5A3AE751-C752-4B34-8E7E-6F593B4A0FA2@gmail.com> No I don't think you can do that. Sent from my iPhone > On Jan 27, 2015, at 00:51, Ben Hood <0x6e6562 at gmail.com> wrote: > >> On Mon, Jan 26, 2015 at 8:00 PM, Ben Hood <0x6e6562 at gmail.com> wrote: >> Aha - does enable-post-var also work when you're using the GET method, >> or do you also have to use POST to enable this? > > I found out by trial and error that by specifying a request parameter > per enable-post-var entry, I could effectively cherry pick the request > even using the GET method. For example, adding these two lines: > > > > > Results in the following request: > > data: [hostname=x§ion=dialplan&tag_name=&key_name=&key_value=&variable_sip_from_user_stripped=1234567890&variable_sip_to_user=0987654321] > > So thank you very much for the tip off. > > Do you happen to know how I could even get rid of the empty tag_name, > key_name and key_value request parameters? From denis at ringme.ru Tue Jan 27 12:03:32 2015 From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=) Date: Tue, 27 Jan 2015 12:03:32 +0300 Subject: [Freeswitch-users] lua + aes|md5 In-Reply-To: <54C648DA.8070804@wirelessmundi.com> References: <54C60117.7040204@ringme.ru> <54C648DA.8070804@wirelessmundi.com> Message-ID: <54C75464.9000305@ringme.ru> > Are you using esl? No, mod_lua. From 0x6e6562 at gmail.com Tue Jan 27 12:05:45 2015 From: 0x6e6562 at gmail.com (Ben Hood) Date: Tue, 27 Jan 2015 09:05:45 +0000 Subject: [Freeswitch-users] Configuring request parameters with mod_xml_curl In-Reply-To: <5A3AE751-C752-4B34-8E7E-6F593B4A0FA2@gmail.com> References: <5A3AE751-C752-4B34-8E7E-6F593B4A0FA2@gmail.com> Message-ID: On Tue, Jan 27, 2015 at 8:55 AM, Muhammad Naseer Bhatti wrote: > No I don't think you can do that. OK, thanks for letting me know. At some stage I could look at the source to see how/whether this is controlled, but for now, your tip about enable-post-var has improved things significantly. From denis at ringme.ru Tue Jan 27 12:07:33 2015 From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=) Date: Tue, 27 Jan 2015 12:07:33 +0300 Subject: [Freeswitch-users] lua + aes|md5 In-Reply-To: References: <54C60117.7040204@ringme.ru> <54C648DA.8070804@wirelessmundi.com> Message-ID: <54C75555.3050704@ringme.ru> On 26.01.2015 23:07, Michael Jerris wrote: > In master I just added the --enable-system-lua configure flag that makes mod_lua and esl lua both use the system lib, and fixed a number of issues using different versions of lua. This may help you out. Please note, in 1.6 we will remove the need for the configure flag, and this will always use system libs. > You have legacy mod_lua with 5.1, why you can't make another module like mod_lua_legacy? Why 5.1 now deprecated and in files.freeswitch.org your lua-5.2 packages missing? Please, add lua (5.2) for system. From dp.siddharth at eng.knowlarity.com Tue Jan 27 14:36:00 2015 From: dp.siddharth at eng.knowlarity.com (DP Siddharth) Date: Tue, 27 Jan 2015 17:06:00 +0530 Subject: [Freeswitch-users] Getting Redirecting number in Events (ISDN) In-Reply-To: References: Message-ID: So I tested it again with libpri in place of sangoma_isdn & seems working. Actually I saw ftmod_sangoma_isdn code & found problem in getting from library itself (libsng_isdn). So thought of testing first with libpri before moving to 1.4. I will discuss with Sangoma team for this. thanks for valuable inputs. On Mon, Jan 26, 2015 at 7:07 PM, DP Siddharth < dp.siddharth at eng.knowlarity.com> wrote: > Thanks , I will test it with 1.4. > > > On Mon, Jan 26, 2015 at 3:29 AM, Michael Collins > wrote: > >> I strongly recommend you spin up a FreeSWITCH 1.4 instance for testing. >> The 1.2 branch has served its purpose and has been honorably discharged as >> it were. If what you need gets added to the code base it will not be in 1.2. >> >> When you get 1.4 up and running be sure to route an ISDN call through the >> info dialplan app just to make sure it's not already there and available >> for your use. If it's available in a channel variable then it should be >> available to add to events as needed. >> >> -Michael >> >> On Wed, Jan 21, 2015 at 3:58 AM, DP Siddharth < >> dp.siddharth at eng.knowlarity.com> wrote: >> >>> Hi, >>> >>> I am using ISDN (PRI) for handling calls. In a use case we need >>> Redirecting number as part of freeswitch events. I verified various events >>> but we get only caller/called. I was looking into the code & found for SS7 >>> this is available but not for ISDN, just need confirmation if any way >>> possible to get this. I am using 1.2.24 (1.2.stable) freeswitch for testing. >>> >>> following is ISDN SETUP message for reference: >>> >>> Prot Disc:Q.931/I.451 (0x08) >>> Call Ref:242E (Originating side) >>> Type:SETUP (0x5) >>> Sending complete: >>> Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) >>> TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3) >>> Channel Id:No:13 Type:B-chans(3) Preferred/Implicit >>> Calling Party Number:8860128000(l:10) plan:isdn(1) >>> type:national(2)scr:network, provided(3) pres:allowed(0) >>> Called Party Number:01725218700(l:11) plan:isdn(1) type:subscriber(4) >>> Redirecting Number:53703417(l:8) plan:isdn(1) >>> type:national(2)scr:user, not screened(0) pres:restricted(1)reason:Call >>> forwarding unconditional(15) >>> Redirecting Number:53703417(l:8) plan:isdn(1) >>> type:national(2)scr:user, not screened(0) pres:restricted(1)reason:Call >>> forwarding unconditional(15) >>> [ 08 02 24 2e 05 a1 04 03 80 90 a3 18 03 a1 83 8d 6c 0c 21 83 38 38 36 >>> 30 31 32 38 30 30 30 70 0c >>> c1 30 31 37 32 35 32 31 38 37 30 30 74 0d 21 20 8f 39 36 35 33 37 30 >>> 33 34 31 37 74 0d 21 20 8f >>> 39 36 35 33 37 30 33 34 31 37 ] >>> >>> >>> -- >>> Thanks & Regards, >>> D P Siddharth >>> Director (Platform) >>> Knowlarity Communications >>> Ph: +919999115231 >>> dp.siddharth at eng.knowlarity.com >>> >>> *"Come together to build a lasting world-class cloud telephony company >>> that helps businesses grow"* >>> >>> _________________________________________________________________________ >>> 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 >> > > > > -- > Thanks & Regards, > D P Siddharth > Director (Platform) > Knowlarity Communications > Ph: +919999115231 > dp.siddharth at eng.knowlarity.com > > *"Come together to build a lasting world-class cloud telephony company > that helps businesses grow"* > -- Thanks & Regards, D P Siddharth Director (Platform) Knowlarity Communications Ph: +919999115231 dp.siddharth at eng.knowlarity.com *"Come together to build a lasting world-class cloud telephony company that helps businesses grow"* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/14c0e3dc/attachment.html From steveayre at gmail.com Tue Jan 27 14:53:21 2015 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 27 Jan 2015 11:53:21 +0000 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: References: <31690C3C-FF53-4BD3-B738-352E71396CCA@jerris.com> Message-ID: It'd be better to keep source and build separate. You could build in /usr/src/freeswitch and install to /opt/freeswitch for example. /usr/src shouldn't really be used for running a daemon. Since you're using Debian are you aware of the APT repository on files.freeswitch.org? That installs it to standard directories you would expect (/usr/bin/ /usr/sbin/ /etc/freeswitch and so on). On 27 January 2015 at 07:14, Grant Bagdasarian wrote: > Its ?prefix with a single -. > > > > Yes, the directory is /usr/src/freeswitch. I always use this parameter to > install all the files in the same directory, else it installs in > /usr/local/freeswitch I believe. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* Monday, January 26, 2015 9:24 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] APR issue when configuring > > > > Also is that -prefix or --prefix you're using? > > > > On 26 January 2015 at 20:21, Michael Jerris wrote: > > is /usr/src/freeswitch your source directory too? Its a bit weird to be > specifying that as prefix, maybe its an issue when specifying a prefix as > the same as your source dir? > > > > On Jan 26, 2015, at 12:51 AM, Grant Bagdasarian wrote: > > > > Hello, > > > > I just extracted the tarball, installed a few missing dependencies and ran > the configure command: > > ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Ken Rice > *Sent:* Friday, January 23, 2015 4:44 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] APR issue when configuring > > > > I?m inclined to believe you have something broken on your box here. I just > grabbed the tarball and built it right on my Wheezy box... How exactly are > you trying to build it from the tarball? > > > On 1/23/15, 8:49 AM, "Grant Bagdasarian" wrote: > > > *lsb_release -a *No LSB modules are available. > Distributor ID: Debian > Description: Debian GNU/Linux 7.8 (wheezy) > Release: 7.8 > Codename: wheezy > > > *uname -a *Linux HOSTNAME 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 > GNU/Linux > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Ken Rice > *Sent:* Friday, January 23, 2015 3:04 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] APR issue when configuring > > > On what platform are you getting this error? > > > > > Sent from my iPhone > > > On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian wrote: > > > Nevermind. > I used the 1.4.15 tarball which contained this error. > > Now I?ve used the git clone master which works fine. > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Grant > Bagdasarian > *Sent:* Friday, January 23, 2015 11:51 AM > *To:* FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) > *Subject:* [Freeswitch-users] APR issue when configuring > > Hello, > > I?m getting the following error during configure. > > checking for APR... configure: error: the --with-apr parameter is > incorrect. It must specify an install prefix, a build directory, or an > apr-config file. > configure: error: ./configure.gnu failed for libs/apr-util > > I?m running the following command to configure: ./configure > -prefix=/usr/src/freeswitch/ --enable-core-odbc-support > > Also, a lot of directories inside the libs require the configure script to > be executable. I have to set this 1 by 1. > > Is there a way to disable apr? Or is it required by FS? > > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/2d0ce37c/attachment-0001.html From steveayre at gmail.com Tue Jan 27 14:54:52 2015 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 27 Jan 2015 11:54:52 +0000 Subject: [Freeswitch-users] Port changes in multiple 183s cause no audio after 200 In-Reply-To: References: Message-ID: This sounds like it belongs on Jira so the issue can be tracked. On 27 January 2015 at 00:04, Jeff Pyle wrote: > Hello, > > The following is on FreeSWITCH Version > 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 > 21:57:33Z 64bit). > > The design goal for this configuration is that of a simple transcoding > SBC. SIP calls arrive with various supported codecs, SIP calls bridge out > on PCMU. No users, no auth, etc. Overall, it seems to work but there is > one call flow I'm struggling with. > > The B-leg of calls are bridged to a PSTN gateway. If the gateway's > signaling follows 100, 183, 200, all is well. But if the gateway sends > multiple 183s with different RTP ports, there is no audio when the call > goes to 200. See the following example: > > - Gateway signals 183 with SDP indicating audio on port 16384. > - Gateway signals 183 with SDP indicating audio on port 16386. > - Gateway signals 200 with SDP indicating audio on port 16384 (same as > original 183). > > In the debug I see where it detects the port change from 183 #1 to 183 > #2. As such, I hear early media from both until the 200 OK. When the call > connects, I see no such port change in the debug, and since it's still > listening on the wrong port (from 183 #2), there is no audio. > > I've seen some older posts where Anthony seemed against even the port > change from 183 #1 to 183 #2, yet that seems to work okay today. It just > doesn't sense the port change from 183 --> 200. I don't know if this is a > feature, bug, or misconfigured option. Thoughts are welcome! > > > - Jeff > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/2d300a2a/attachment.html From groysem at gmail.com Tue Jan 27 16:06:32 2015 From: groysem at gmail.com (Shai Perelman) Date: Tue, 27 Jan 2015 15:06:32 +0200 Subject: [Freeswitch-users] can I use Event_Socket_Outbound to only monitor? In-Reply-To: References: Message-ID: Hi, thanks, as far as I can see even with the async option it still waits for commands from my side and the call is not transfered to music on hold which is what I am dialing, the moment I remove the socket action it returns to normal behaviour. so I am trying this from the other end as Kamil suggested, with inbound event socket. it works and the call proceeds while I am getting the events but I need to filter the events I am getting to only include the events with the extension(s) that I want to monitor involved. can somebody help me with what filter command(s) do I need to send to see only the events that have for example extension 9999 involved? thanks Shai On Tue, Jan 27, 2015 at 5:59 AM, Michael Collins wrote: > Using the socket app with the "async" option will cause that channel's > events to be sent. > -MC > > On Mon, Jan 26, 2015 at 7:43 PM, Kamil Nigmatullin < > kamil.nigmatullin at gmail.com> wrote: > >> Yes if you want monitor and catch events you need to use inbound socket >> 26 ???. 2015 ?. 21:58 ???????????? "Duvid Rottenberg" < >> adrottenberg at gmail.com> ???????: >> >>> I am pretty sure that with outbound socket as long as you don't send any >>> commands back to Freeswitch, it will just execute whatever is in the xml >>> dialplan. >>> >>> Thanks, >>> Duvid Rottenberg >>> >>> On Sun, Jan 25, 2015 at 6:19 PM, Michael Collins >>> wrote: >>> >>>> In event socket outbound the listening server controls the call. In >>>> your case you just want to get the events associated with that call? I've >>>> never actually tried it but maybe you could do something like: >>>> >>>> >>>> >>>> >>>> And then after the socket app answers have it issue the linger command, >>>> then transfer or execute_extension to send the call off to do something. >>>> Watch your socket connection for events and see happens. If you get it to >>>> work please let us know. :) >>>> >>>> -Michael >>>> >>>> >>>> >>>> On Sun, Jan 25, 2015 at 7:17 AM, Shai Perelman >>>> wrote: >>>> >>>>> >>>>> hi, I am using >>>>> https://wiki.freeswitch.org/wiki/Event_Socket_Outbound >>>>> my goal is to monitor what happens on the extension but without >>>>> affecting the normal behaviour. >>>>> >>>>> I have this dialplan. >>>>> >>>>> TagTypeDataOrder[image: add] >>>>> >>>>> condition username 9998 1[image: edit] >>>>> >>>>> [image: delete] >>>>> >>>>> action socket 109.65.149.22:9999 async 3[image: edit] >>>>> >>>>> [image: delete] >>>>> >>>>> >>>>> >>>>> I am calling *9664 (music on hold) >>>>> and the call is blocking while wating for commands from my side. >>>>> >>>>> I want it to answer automatticaly like it would do with out the socket >>>>> module, and just send me the events regarding that extension >>>>> >>>>> thanks >>>>> Shai >>>>> ITD Communications >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>> >>> >>> >>> _________________________________________________________________________ >>> 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 >> > > > _________________________________________________________________________ > 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 > -- www.groyse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/15a73cdb/attachment-0001.html From aqsyounas at gmail.com Tue Jan 27 16:50:05 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 27 Jan 2015 18:50:05 +0500 Subject: [Freeswitch-users] how to transfer all conference members to an extension on stop-talking event. In-Reply-To: References: Message-ID: Thank you guys. Happy to find a very helping Freeswitch Community. On 27 January 2015 at 08:58, Michael Collins wrote: > You can use the "conference list" API to get a list of all uuids in the > conference and then use uuid_transfer on each of them. > > -MC > > On Mon, Jan 26, 2015 at 1:45 PM, Aqs Younas wrote: > >> Hi, list. >> >> I need to transfer all conference members to some other extension >> whenever there is any stop-talking event. But don't know how can i do so? >> >> I am using python script as inbound connection to listening to events. >> Whenever there is stop-talking event I need to transfer all conference >> members to another dialplan extension. >> >> Your help would be much appreciated. >> >> Thanks. >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/d039d190/attachment.html From phenix at vfemail.net Tue Jan 27 14:14:13 2015 From: phenix at vfemail.net (tanguy) Date: Tue, 27 Jan 2015 05:14:13 -0600 Subject: [Freeswitch-users] Freeswitch don't send keepalive In-Reply-To: References: <54C56372.5070708@vfemail.net> <54C6A98D.5010708@vfemail.net> Message-ID: <20150127051413.Horde.zZcJgMVVwhgFJiOpxa70cg2@www.vfemail.net> Hello Sorry, my message was not realy clear. Rebooting the phone was useless, i have rebooted the server himself. Regards Quoting Moishe Grunstein : > Might be a bug in the phone's firmware. > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice ------------------------------------------------- VFEmail.net - http://www.vfemail.net ONLY AT VFEmail! - Use our Metadata Mitigator to keep your email out of the NSA's hands! $24.95 ONETIME Lifetime accounts with Privacy Features! 15GB disk! No bandwidth quotas! Commercial and Bulk Mail Options! From mike at jerris.com Tue Jan 27 17:53:59 2015 From: mike at jerris.com (Michael Jerris) Date: Tue, 27 Jan 2015 08:53:59 -0600 Subject: [Freeswitch-users] lua + aes|md5 In-Reply-To: <54C75555.3050704@ringme.ru> References: <54C60117.7040204@ringme.ru> <54C648DA.8070804@wirelessmundi.com> <54C75555.3050704@ringme.ru> Message-ID: <5E0553FD-ED2E-4F2E-ADD1-00EBB7B1E3B6@jerris.com> > On Jan 27, 2015, at 3:07 AM, ????? wrote: > > > On 26.01.2015 23:07, Michael Jerris wrote: >> In master I just added the --enable-system-lua configure flag that makes mod_lua and esl lua both use the system lib, and fixed a number of issues using different versions of lua. This may help you out. Please note, in 1.6 we will remove the need for the configure flag, and this will always use system libs. >> > You have legacy mod_lua with 5.1, why you can't make another module like > mod_lua_legacy? > Why 5.1 now deprecated and in files.freeswitch.org your lua-5.2 packages > missing? > Please, add lua (5.2) for system. legacy mod_lua is no longer needed, as a just described above. 5.1 is not deprecated. From olegstolyar at gmail.com Tue Jan 27 18:04:52 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Tue, 27 Jan 2015 07:04:52 -0800 Subject: [Freeswitch-users] Connecting to FS from a mobile app Message-ID: Guys, Has anyone done that? I currently have WebRTC clients connecting to FS from Chrome and/or Firefox but I now need to do it from my mobile apps (Android and iOS). Any recommendation on how to do it? I use JsSip in the browsers. Any way to make a JS library like that to work from an app? If not, any other suggestions? WebRTC vs. SIP? WebRTC with SIP vs WebRTC with Verto? I apologize if this topic is not about Freeswitch's own functionality but I thought people would b interested and wanted to pick this group's brain. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/02f286e7/attachment.html From krice at freeswitch.org Tue Jan 27 18:19:28 2015 From: krice at freeswitch.org (Ken Rice) Date: Tue, 27 Jan 2015 09:19:28 -0600 Subject: [Freeswitch-users] Connecting to FS from a mobile app In-Reply-To: Message-ID: Verto already works in Chrome on Android.... Was using it that way just recently On 1/27/15, 9:04 AM, "Oleg Stolyar" wrote: > Guys, > > Has anyone done that?? I currently have WebRTC clients connecting to FS from > Chrome and/or Firefox but I now need to do it from my mobile apps (Android and > iOS). > > Any recommendation on how to do it?? I use JsSip in the browsers.? Any way to > make a JS library like that to work from an app?? If not, any other > suggestions? ? > > WebRTC vs. SIP? > WebRTC with SIP vs WebRTC with Verto? > > I apologize if this topic is not about Freeswitch's own functionality but I > thought people would b interested and wanted to pick this group's brain. > > > > > _________________________________________________________________________ > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/0ebfc5d3/attachment.html From bordmi at rarus.ru Tue Jan 27 18:21:08 2015 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=) Date: Tue, 27 Jan 2015 19:21:08 +0400 Subject: [Freeswitch-users] Connecting to FS from a mobile app In-Reply-To: References: Message-ID: Why not Android SIP client??? 2015-01-27 18:04 GMT+03:00 Oleg Stolyar : > Guys, > > Has anyone done that? I currently have WebRTC clients connecting to FS > from Chrome and/or Firefox but I now need to do it from my mobile apps > (Android and iOS). > > Any recommendation on how to do it? I use JsSip in the browsers. Any way > to make a JS library like that to work from an app? If not, any other > suggestions? > > WebRTC vs. SIP? > WebRTC with SIP vs WebRTC with Verto? > > I apologize if this topic is not about Freeswitch's own functionality but > I thought people would b interested and wanted to pick this group's brain. > > > > _________________________________________________________________________ > 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 > -- with best regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/25eb1f78/attachment-0001.html From sdame at 207me.com Tue Jan 27 18:21:38 2015 From: sdame at 207me.com (Stephen Dame) Date: Tue, 27 Jan 2015 10:21:38 -0500 Subject: [Freeswitch-users] Connecting to FS from a mobile app In-Reply-To: References: Message-ID: <043601d03a44$f2fd2d50$d8f787f0$@207me.com> The standard sip.js library with dozen lines of html5 works well on desktops (PC/MAC), and ANDRIOD using the chrome browser on my android Galaxy3 phone, and Galaxy Tab with no modifications. We have it enabled for stereo opus/48000/2, and have also been able to connect listen only without requiring permissions to grab the mic. Look out icecast, great quality and no delay. Have not looked at how to get it running in IOS yet. Regards, Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Oleg Stolyar Sent: Tuesday, January 27, 2015 10:05 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Connecting to FS from a mobile app Guys, Has anyone done that? I currently have WebRTC clients connecting to FS from Chrome and/or Firefox but I now need to do it from my mobile apps (Android and iOS). Any recommendation on how to do it? I use JsSip in the browsers. Any way to make a JS library like that to work from an app? If not, any other suggestions? WebRTC vs. SIP? WebRTC with SIP vs WebRTC with Verto? I apologize if this topic is not about Freeswitch's own functionality but I thought people would b interested and wanted to pick this group's brain. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/451b9a67/attachment.html From olegstolyar at gmail.com Tue Jan 27 18:33:52 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Tue, 27 Jan 2015 07:33:52 -0800 Subject: [Freeswitch-users] Connecting to FS from a mobile app In-Reply-To: References: Message-ID: Thanks guys, Ken, Stephen, I also have it running in Chrome on Android but I need it to work from my app. Not sure if there is a seamless way to open a hidden Chrome tab from an app. I am exploring it among other things. Dmitriy, Android SIP client is a possibility. I prefer to try to make it work with WebRTC first though because of the built-in security (with SIP I'd need to implement TLS) and for a couple other internal reasons. If I can't make it work with WebRTC, I'll start looking into SIP - I know those are more standard. On Tue, Jan 27, 2015 at 7:21 AM, ???????, ??????? / Dmitriy Borisov < bordmi at rarus.ru> wrote: > Why not Android SIP client??? > > 2015-01-27 18:04 GMT+03:00 Oleg Stolyar : > >> Guys, >> >> Has anyone done that? I currently have WebRTC clients connecting to FS >> from Chrome and/or Firefox but I now need to do it from my mobile apps >> (Android and iOS). >> >> Any recommendation on how to do it? I use JsSip in the browsers. Any >> way to make a JS library like that to work from an app? If not, any other >> suggestions? >> >> WebRTC vs. SIP? >> WebRTC with SIP vs WebRTC with Verto? >> >> I apologize if this topic is not about Freeswitch's own functionality but >> I thought people would b interested and wanted to pick this group's brain. >> >> >> >> _________________________________________________________________________ >> 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 >> > > > > -- > with best regards, > Dmitriy Borisov > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/3446fb33/attachment.html From brian at freeswitch.org Tue Jan 27 19:01:10 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 27 Jan 2015 10:01:10 -0600 Subject: [Freeswitch-users] Bug Marshalls Needed! Message-ID: FreeSWITCHers, We're currently in need of bug marshalls, The job is basically to triage JIRA's make sure the data is correct, try to replicate the issue if the description is clear enough or ask for more info about the issue. If you're interested in helping out please email krice at freeswitch.org or myself. Thanks, -- *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/20150127/ffc61e89/attachment.html From lists at kavun.ch Tue Jan 27 19:01:38 2015 From: lists at kavun.ch (Emrah) Date: Tue, 27 Jan 2015 17:01:38 +0100 Subject: [Freeswitch-users] Cannot contact registered endpoint behind NAT since latest make current Message-ID: <19B3EC9F-DD9A-45C7-99D0-0A2676D5D6C2@kavun.ch> Hi all, Since my make current of this week, I cannot bridge to registered endpoints that are seen by my FS as devices behind NAT. I tried bridging both with user/aurora605 at domain as well as using the full sofia_contact value. Here is the output I get: Cannot locate registered user aurora605 at 192.168.2.191;received=1.2.3.4:34146;fs_nat=yes;fs_path=sip at 3Aaurora605%401.2.3.4%3A34146 this is the output of sofia_contact sofia/internal/sip:aurora605 at 192.168.2.191;received=1.2.3.4:1029;fs_nat=yes;fs_path=sip%3Aaurora605%401.2.3.4%3A1029 What are your suggestions to fix this issue? Best, Emrah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/fa33431d/attachment.html From ing.antonyam at gmail.com Tue Jan 27 19:38:55 2015 From: ing.antonyam at gmail.com (Antony Aguirre Morales) Date: Tue, 27 Jan 2015 10:38:55 -0600 Subject: [Freeswitch-users] Error whith xml_curl Message-ID: I set my freeswitch with the directory database, now by registering my extensions are connected correctly, but when making a call of the extension extention 1 to 2 me throws the following error "01/27/2015 10: 22: 35.158536 [ERR] mod_dptools.c: 3977 No dial-string available, Please check your user directory." and directly sends the voice mailbox of the extension. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/d07b0738/attachment-0001.html From blasterjr at gmail.com Tue Jan 27 20:37:20 2015 From: blasterjr at gmail.com (Chris Tunbridge) Date: Tue, 27 Jan 2015 10:37:20 -0700 Subject: [Freeswitch-users] Connecting to FS from a mobile app In-Reply-To: References: Message-ID: Just wanted to give my experience, if you want to make an app using WebRTC/WebSockets you can use SIP.js + PhoneRTC + Cordova for Android & iOS, be warned that the garbage collector and pausing will cause you to drop registration without a background service which gets much more complicated. Cordova: http://cordova.apache.org/ PhoneRTC: https://github.com/alongubkin/phonertc SIP.js: http://sipjs.com/ SIP.js PhoneRTC Media Handler: https://github.com/joseph-onsip/PhoneRTCMediaHandler As for the person who mentioned Android SIP Client, its terrible and not available on all devices, out of 40 android devices we tested only 15 of them could actually use the Android SIP client, and out of those 15, only 5 could reliably make and receive calls without either crashing, dropping audio, or having massive delays in the audio (WiFi, 3g, 4g and LTE tested). On Tue, Jan 27, 2015 at 8:33 AM, Oleg Stolyar wrote: > Thanks guys, > > Ken, Stephen, > > I also have it running in Chrome on Android but I need it to work from my > app. Not sure if there is a seamless way to open a hidden Chrome tab from > an app. I am exploring it among other things. > > Dmitriy, > > Android SIP client is a possibility. I prefer to try to make it work with > WebRTC first though because of the built-in security (with SIP I'd need to > implement TLS) and for a couple other internal reasons. > > If I can't make it work with WebRTC, I'll start looking into SIP - I know > those are more standard. > > On Tue, Jan 27, 2015 at 7:21 AM, ???????, ??????? / Dmitriy Borisov < > bordmi at rarus.ru> wrote: > >> Why not Android SIP client??? >> >> 2015-01-27 18:04 GMT+03:00 Oleg Stolyar : >> >>> Guys, >>> >>> Has anyone done that? I currently have WebRTC clients connecting to FS >>> from Chrome and/or Firefox but I now need to do it from my mobile apps >>> (Android and iOS). >>> >>> Any recommendation on how to do it? I use JsSip in the browsers. Any >>> way to make a JS library like that to work from an app? If not, any other >>> suggestions? >>> >>> WebRTC vs. SIP? >>> WebRTC with SIP vs WebRTC with Verto? >>> >>> I apologize if this topic is not about Freeswitch's own functionality >>> but I thought people would b interested and wanted to pick this group's >>> brain. >>> >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> -- >> with best regards, >> Dmitriy Borisov >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/daedd71b/attachment.html From blasterjr at gmail.com Tue Jan 27 20:40:13 2015 From: blasterjr at gmail.com (Chris Tunbridge) Date: Tue, 27 Jan 2015 10:40:13 -0700 Subject: [Freeswitch-users] Connecting to FS from a mobile app In-Reply-To: References: Message-ID: Also if you want to target Android only there's also the Crosswalk platform https://crosswalk-project.org/ This provides a Chrome Based web view for use similar to Cordova. On Tue, Jan 27, 2015 at 10:37 AM, Chris Tunbridge wrote: > Just wanted to give my experience, if you want to make an app using > WebRTC/WebSockets you can use SIP.js + PhoneRTC + Cordova for Android & > iOS, be warned that the garbage collector and pausing will cause you to > drop registration without a background service which gets much more > complicated. > > Cordova: http://cordova.apache.org/ > PhoneRTC: https://github.com/alongubkin/phonertc > SIP.js: http://sipjs.com/ > SIP.js PhoneRTC Media Handler: > https://github.com/joseph-onsip/PhoneRTCMediaHandler > > As for the person who mentioned Android SIP Client, its terrible and not > available on all devices, out of 40 android devices we tested only 15 of > them could actually use the Android SIP client, and out of those 15, only 5 > could reliably make and receive calls without either crashing, dropping > audio, or having massive delays in the audio (WiFi, 3g, 4g and LTE tested). > > On Tue, Jan 27, 2015 at 8:33 AM, Oleg Stolyar > wrote: > >> Thanks guys, >> >> Ken, Stephen, >> >> I also have it running in Chrome on Android but I need it to work from my >> app. Not sure if there is a seamless way to open a hidden Chrome tab from >> an app. I am exploring it among other things. >> >> Dmitriy, >> >> Android SIP client is a possibility. I prefer to try to make it work >> with WebRTC first though because of the built-in security (with SIP I'd >> need to implement TLS) and for a couple other internal reasons. >> >> If I can't make it work with WebRTC, I'll start looking into SIP - I know >> those are more standard. >> >> On Tue, Jan 27, 2015 at 7:21 AM, ???????, ??????? / Dmitriy Borisov < >> bordmi at rarus.ru> wrote: >> >>> Why not Android SIP client??? >>> >>> 2015-01-27 18:04 GMT+03:00 Oleg Stolyar : >>> >>>> Guys, >>>> >>>> Has anyone done that? I currently have WebRTC clients connecting to FS >>>> from Chrome and/or Firefox but I now need to do it from my mobile apps >>>> (Android and iOS). >>>> >>>> Any recommendation on how to do it? I use JsSip in the browsers. Any >>>> way to make a JS library like that to work from an app? If not, any other >>>> suggestions? >>>> >>>> WebRTC vs. SIP? >>>> WebRTC with SIP vs WebRTC with Verto? >>>> >>>> I apologize if this topic is not about Freeswitch's own functionality >>>> but I thought people would b interested and wanted to pick this group's >>>> brain. >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>> >>> >>> >>> -- >>> with best regards, >>> Dmitriy Borisov >>> >>> _________________________________________________________________________ >>> 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 >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/5c814f5b/attachment.html From krice at freeswitch.org Tue Jan 27 20:56:36 2015 From: krice at freeswitch.org (Ken Rice) Date: Tue, 27 Jan 2015 11:56:36 -0600 Subject: [Freeswitch-users] Buy the FreeSWITCH Dev's Dinner! Message-ID: Hey Guys, The core FreeSWITCH Team is all together for the Annual Engineering and Planning Meeting! Nows your chance to buy them dinner in appreciation for all their hard work! Visit https://freeswitch.org/ and hit the donate button and let them know this is for dinner! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/1856d78c/attachment-0001.html From alipey at gmail.com Tue Jan 27 21:21:05 2015 From: alipey at gmail.com (Ali Pey) Date: Tue, 27 Jan 2015 13:21:05 -0500 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:2620 Socket Error! Message-ID: Hello, What should I be looking at if I get this error time to time? [CRIT] mod_event_socket.c:2620 Socket Error! It seems to be traffic related and happens around 370 channels utilized. Sometimes, it kills the FS server. I have the ulimit settings for the freeswitch process as followings: ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 240 ulimit -l unlimited Any suggestions? Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/4babbbf2/attachment.html From olegstolyar at gmail.com Tue Jan 27 21:23:58 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Tue, 27 Jan 2015 10:23:58 -0800 Subject: [Freeswitch-users] Connecting to FS from a mobile app In-Reply-To: References: Message-ID: Thank you Chris, this is amazingly useful feedback! On Tue, Jan 27, 2015 at 9:40 AM, Chris Tunbridge wrote: > Also if you want to target Android only there's also the Crosswalk platform > > https://crosswalk-project.org/ > > This provides a Chrome Based web view for use similar to Cordova. > > > > On Tue, Jan 27, 2015 at 10:37 AM, Chris Tunbridge > wrote: > >> Just wanted to give my experience, if you want to make an app using >> WebRTC/WebSockets you can use SIP.js + PhoneRTC + Cordova for Android & >> iOS, be warned that the garbage collector and pausing will cause you to >> drop registration without a background service which gets much more >> complicated. >> >> Cordova: http://cordova.apache.org/ >> PhoneRTC: https://github.com/alongubkin/phonertc >> SIP.js: http://sipjs.com/ >> SIP.js PhoneRTC Media Handler: >> https://github.com/joseph-onsip/PhoneRTCMediaHandler >> >> As for the person who mentioned Android SIP Client, its terrible and not >> available on all devices, out of 40 android devices we tested only 15 of >> them could actually use the Android SIP client, and out of those 15, only 5 >> could reliably make and receive calls without either crashing, dropping >> audio, or having massive delays in the audio (WiFi, 3g, 4g and LTE tested). >> >> On Tue, Jan 27, 2015 at 8:33 AM, Oleg Stolyar >> wrote: >> >>> Thanks guys, >>> >>> Ken, Stephen, >>> >>> I also have it running in Chrome on Android but I need it to work from >>> my app. Not sure if there is a seamless way to open a hidden Chrome tab >>> from an app. I am exploring it among other things. >>> >>> Dmitriy, >>> >>> Android SIP client is a possibility. I prefer to try to make it work >>> with WebRTC first though because of the built-in security (with SIP I'd >>> need to implement TLS) and for a couple other internal reasons. >>> >>> If I can't make it work with WebRTC, I'll start looking into SIP - I >>> know those are more standard. >>> >>> On Tue, Jan 27, 2015 at 7:21 AM, ???????, ??????? / Dmitriy Borisov < >>> bordmi at rarus.ru> wrote: >>> >>>> Why not Android SIP client??? >>>> >>>> 2015-01-27 18:04 GMT+03:00 Oleg Stolyar : >>>> >>>>> Guys, >>>>> >>>>> Has anyone done that? I currently have WebRTC clients connecting to >>>>> FS from Chrome and/or Firefox but I now need to do it from my mobile apps >>>>> (Android and iOS). >>>>> >>>>> Any recommendation on how to do it? I use JsSip in the browsers. Any >>>>> way to make a JS library like that to work from an app? If not, any other >>>>> suggestions? >>>>> >>>>> WebRTC vs. SIP? >>>>> WebRTC with SIP vs WebRTC with Verto? >>>>> >>>>> I apologize if this topic is not about Freeswitch's own functionality >>>>> but I thought people would b interested and wanted to pick this group's >>>>> brain. >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>> >>>> >>>> >>>> -- >>>> with best regards, >>>> Dmitriy Borisov >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/b08ed590/attachment.html From italorossib at gmail.com Tue Jan 27 22:40:20 2015 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 27 Jan 2015 16:40:20 -0300 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:2620 Socket Error! In-Reply-To: References: Message-ID: Try to get a backtrace. This error is raised when FS looses connection with an inbound/outbound socket, check your application logs. On Tue, Jan 27, 2015 at 3:21 PM, Ali Pey wrote: > Hello, > > What should I be looking at if I get this error time to time? > > [CRIT] mod_event_socket.c:2620 Socket Error! > > > It seems to be traffic related and happens around 370 channels utilized. > Sometimes, it kills the FS server. > > I have the ulimit settings for the freeswitch process as followings: > > ulimit -c unlimited > ulimit -d unlimited > ulimit -f unlimited > ulimit -i unlimited > ulimit -n 999999 > ulimit -q unlimited > ulimit -u unlimited > ulimit -v unlimited > ulimit -x unlimited > ulimit -s 240 > ulimit -l unlimited > > Any suggestions? > > Thanks, > Ali Pey > > > _________________________________________________________________________ > 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 > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/8f47e075/attachment.html From brian at freeswitch.org Tue Jan 27 22:41:02 2015 From: brian at freeswitch.org (Brian West) Date: Tue, 27 Jan 2015 13:41:02 -0600 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:2620 Socket Error! In-Reply-To: References: Message-ID: How are you using event socket? Is this the outbound event socket stuff? On Tue, Jan 27, 2015 at 12:21 PM, Ali Pey wrote: > Hello, > > What should I be looking at if I get this error time to time? > > [CRIT] mod_event_socket.c:2620 Socket Error! > > > It seems to be traffic related and happens around 370 channels utilized. > Sometimes, it kills the FS server. > > I have the ulimit settings for the freeswitch process as followings: > > ulimit -c unlimited > ulimit -d unlimited > ulimit -f unlimited > ulimit -i unlimited > ulimit -n 999999 > ulimit -q unlimited > ulimit -u unlimited > ulimit -v unlimited > ulimit -x unlimited > ulimit -s 240 > ulimit -l unlimited > > Any suggestions? > > Thanks, > Ali Pey > > > _________________________________________________________________________ > 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/20150127/a595e426/attachment-0001.html From alipey at gmail.com Tue Jan 27 23:14:30 2015 From: alipey at gmail.com (Ali Pey) Date: Tue, 27 Jan 2015 15:14:30 -0500 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:2620 Socket Error! In-Reply-To: References: Message-ID: Yes, we are using ESL socket using fs_ivrd. What should I look for? Thanks. On Tue, Jan 27, 2015 at 2:41 PM, Brian West wrote: > How are you using event socket? Is this the outbound event socket stuff? > > On Tue, Jan 27, 2015 at 12:21 PM, Ali Pey wrote: > >> Hello, >> >> What should I be looking at if I get this error time to time? >> >> [CRIT] mod_event_socket.c:2620 Socket Error! >> >> >> It seems to be traffic related and happens around 370 channels utilized. >> Sometimes, it kills the FS server. >> >> I have the ulimit settings for the freeswitch process as followings: >> >> ulimit -c unlimited >> ulimit -d unlimited >> ulimit -f unlimited >> ulimit -i unlimited >> ulimit -n 999999 >> ulimit -q unlimited >> ulimit -u unlimited >> ulimit -v unlimited >> ulimit -x unlimited >> ulimit -s 240 >> ulimit -l unlimited >> >> Any suggestions? >> >> Thanks, >> Ali Pey >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/c1f62082/attachment.html From steveayre at gmail.com Wed Jan 28 02:38:18 2015 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 27 Jan 2015 23:38:18 +0000 Subject: [Freeswitch-users] Connecting to FS from a mobile app In-Reply-To: References: Message-ID: You can also use libraries like pjsip in android/ios apps. On 27 January 2015 at 18:23, Oleg Stolyar wrote: > Thank you Chris, this is amazingly useful feedback! > > On Tue, Jan 27, 2015 at 9:40 AM, Chris Tunbridge > wrote: > >> Also if you want to target Android only there's also the Crosswalk >> platform >> >> https://crosswalk-project.org/ >> >> This provides a Chrome Based web view for use similar to Cordova. >> >> >> >> On Tue, Jan 27, 2015 at 10:37 AM, Chris Tunbridge >> wrote: >> >>> Just wanted to give my experience, if you want to make an app using >>> WebRTC/WebSockets you can use SIP.js + PhoneRTC + Cordova for Android & >>> iOS, be warned that the garbage collector and pausing will cause you to >>> drop registration without a background service which gets much more >>> complicated. >>> >>> Cordova: http://cordova.apache.org/ >>> PhoneRTC: https://github.com/alongubkin/phonertc >>> SIP.js: http://sipjs.com/ >>> SIP.js PhoneRTC Media Handler: >>> https://github.com/joseph-onsip/PhoneRTCMediaHandler >>> >>> As for the person who mentioned Android SIP Client, its terrible and not >>> available on all devices, out of 40 android devices we tested only 15 of >>> them could actually use the Android SIP client, and out of those 15, only 5 >>> could reliably make and receive calls without either crashing, dropping >>> audio, or having massive delays in the audio (WiFi, 3g, 4g and LTE tested). >>> >>> On Tue, Jan 27, 2015 at 8:33 AM, Oleg Stolyar >>> wrote: >>> >>>> Thanks guys, >>>> >>>> Ken, Stephen, >>>> >>>> I also have it running in Chrome on Android but I need it to work from >>>> my app. Not sure if there is a seamless way to open a hidden Chrome tab >>>> from an app. I am exploring it among other things. >>>> >>>> Dmitriy, >>>> >>>> Android SIP client is a possibility. I prefer to try to make it work >>>> with WebRTC first though because of the built-in security (with SIP I'd >>>> need to implement TLS) and for a couple other internal reasons. >>>> >>>> If I can't make it work with WebRTC, I'll start looking into SIP - I >>>> know those are more standard. >>>> >>>> On Tue, Jan 27, 2015 at 7:21 AM, ???????, ??????? / Dmitriy Borisov < >>>> bordmi at rarus.ru> wrote: >>>> >>>>> Why not Android SIP client??? >>>>> >>>>> 2015-01-27 18:04 GMT+03:00 Oleg Stolyar : >>>>> >>>>>> Guys, >>>>>> >>>>>> Has anyone done that? I currently have WebRTC clients connecting to >>>>>> FS from Chrome and/or Firefox but I now need to do it from my mobile apps >>>>>> (Android and iOS). >>>>>> >>>>>> Any recommendation on how to do it? I use JsSip in the browsers. >>>>>> Any way to make a JS library like that to work from an app? If not, any >>>>>> other suggestions? >>>>>> >>>>>> WebRTC vs. SIP? >>>>>> WebRTC with SIP vs WebRTC with Verto? >>>>>> >>>>>> I apologize if this topic is not about Freeswitch's own functionality >>>>>> but I thought people would b interested and wanted to pick this group's >>>>>> brain. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> with best regards, >>>>> Dmitriy Borisov >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>> >>> >>> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/efdcb505/attachment-0001.html From aademattia at comcast.net Wed Jan 28 03:49:54 2015 From: aademattia at comcast.net (Andrew) Date: Tue, 27 Jan 2015 19:49:54 -0500 Subject: [Freeswitch-users] sip trace Message-ID: <0a1b01d03a94$5991e9d0$0cb5bd70$@comcast.net> Can FreeSWITCH export a sip trace like XML CDR? Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/27422424/attachment.html From olegstolyar at gmail.com Wed Jan 28 03:56:23 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Tue, 27 Jan 2015 16:56:23 -0800 Subject: [Freeswitch-users] Connecting to FS from a mobile app In-Reply-To: References: Message-ID: Thanks Steven! Do you know if someone successfully used pjsip for production level mobile apps? On Tue, Jan 27, 2015 at 3:38 PM, Steven Ayre wrote: > You can also use libraries like pjsip in android/ios apps. > > On 27 January 2015 at 18:23, Oleg Stolyar wrote: > >> Thank you Chris, this is amazingly useful feedback! >> >> On Tue, Jan 27, 2015 at 9:40 AM, Chris Tunbridge >> wrote: >> >>> Also if you want to target Android only there's also the Crosswalk >>> platform >>> >>> https://crosswalk-project.org/ >>> >>> This provides a Chrome Based web view for use similar to Cordova. >>> >>> >>> >>> On Tue, Jan 27, 2015 at 10:37 AM, Chris Tunbridge >>> wrote: >>> >>>> Just wanted to give my experience, if you want to make an app using >>>> WebRTC/WebSockets you can use SIP.js + PhoneRTC + Cordova for Android & >>>> iOS, be warned that the garbage collector and pausing will cause you to >>>> drop registration without a background service which gets much more >>>> complicated. >>>> >>>> Cordova: http://cordova.apache.org/ >>>> PhoneRTC: https://github.com/alongubkin/phonertc >>>> SIP.js: http://sipjs.com/ >>>> SIP.js PhoneRTC Media Handler: >>>> https://github.com/joseph-onsip/PhoneRTCMediaHandler >>>> >>>> As for the person who mentioned Android SIP Client, its terrible and >>>> not available on all devices, out of 40 android devices we tested only 15 >>>> of them could actually use the Android SIP client, and out of those 15, >>>> only 5 could reliably make and receive calls without either crashing, >>>> dropping audio, or having massive delays in the audio (WiFi, 3g, 4g and LTE >>>> tested). >>>> >>>> On Tue, Jan 27, 2015 at 8:33 AM, Oleg Stolyar >>>> wrote: >>>> >>>>> Thanks guys, >>>>> >>>>> Ken, Stephen, >>>>> >>>>> I also have it running in Chrome on Android but I need it to work from >>>>> my app. Not sure if there is a seamless way to open a hidden Chrome tab >>>>> from an app. I am exploring it among other things. >>>>> >>>>> Dmitriy, >>>>> >>>>> Android SIP client is a possibility. I prefer to try to make it work >>>>> with WebRTC first though because of the built-in security (with SIP I'd >>>>> need to implement TLS) and for a couple other internal reasons. >>>>> >>>>> If I can't make it work with WebRTC, I'll start looking into SIP - I >>>>> know those are more standard. >>>>> >>>>> On Tue, Jan 27, 2015 at 7:21 AM, ???????, ??????? / Dmitriy Borisov < >>>>> bordmi at rarus.ru> wrote: >>>>> >>>>>> Why not Android SIP client??? >>>>>> >>>>>> 2015-01-27 18:04 GMT+03:00 Oleg Stolyar : >>>>>> >>>>>>> Guys, >>>>>>> >>>>>>> Has anyone done that? I currently have WebRTC clients connecting to >>>>>>> FS from Chrome and/or Firefox but I now need to do it from my mobile apps >>>>>>> (Android and iOS). >>>>>>> >>>>>>> Any recommendation on how to do it? I use JsSip in the browsers. >>>>>>> Any way to make a JS library like that to work from an app? If not, any >>>>>>> other suggestions? >>>>>>> >>>>>>> WebRTC vs. SIP? >>>>>>> WebRTC with SIP vs WebRTC with Verto? >>>>>>> >>>>>>> I apologize if this topic is not about Freeswitch's own >>>>>>> functionality but I thought people would b interested and wanted to pick >>>>>>> this group's brain. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> 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 >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> with best regards, >>>>>> Dmitriy Borisov >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>> >>>> >>>> >>> >>> _________________________________________________________________________ >>> 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 >> > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/3b4b9c0d/attachment.html From t.mahe at b-and-c.net Wed Jan 28 04:08:22 2015 From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=) Date: Tue, 27 Jan 2015 17:08:22 -0800 Subject: [Freeswitch-users] sip trace In-Reply-To: <0a1b01d03a94$5991e9d0$0cb5bd70$@comcast.net> References: <0a1b01d03a94$5991e9d0$0cb5bd70$@comcast.net> Message-ID: <54C83686.5060509@b-and-c.net> Hi Andrew, You should look at homer and it's modus operandi. It'll point you to the right direction. Le 27/01/2015 16:49, Andrew a ?crit : > > Can FreeSWITCH export a sip trace like XML CDR? > > > > Andrew > > > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/06f647d6/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150127/06f647d6/attachment-0001.bin From ravenox at gmail.com Wed Jan 28 01:40:52 2015 From: ravenox at gmail.com (=?UTF-8?B?0JDRgNGC0YPRgCDQmtGA0LDQtdCy?=) Date: Wed, 28 Jan 2015 01:40:52 +0300 Subject: [Freeswitch-users] Limit on b-leg Message-ID: Hello everyone! Could you help please how-to realize limits on b-leg? I already saw a wiki page, and way to outbound limit on gateways but, there's some problems to use this solution in my cases. Case 1: [provider sip] -> [fs] -> [user 1] In this case, I'm need to set call limit for user 1 (it can handle no more than concurrent 1 call, connected device is not managed by me) . If I will set it on a-leg in dialplan, the limit will bound to a-leg's UUID and stay here even if b-leg (which is user 1) will be destroyed, for example if user 1 transferred (of att-xferred) call to another user. Case 2: [provider sip] -> [fs] -> [user 1], [user 2], [user 3] (enterprise dial) In this case, I'm need to set own call limit for each user's channel, only for matching b-leg, not for all. This cases, may be realized with loopback channel (which is not elegant, because produces addition problems with cdrs...) but, what will be if one of this users is fax, and we're need to make T.38 pass-trough between a-leg and b-leg? At this moment there's no way to pass-trough t.38 with loopback channels. Ok, after all of this, I'm tried to some sort of googling and found this issue in FS Jira: https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-1792/FS-1792.html I was tried to use uuid_limit for setting limit on b-leg, but: - I cannot use uuid-limit on a-leg because, there's no uuid for b-leg before bridge called. I can generate uuid by api function, and set for b-leg as channel variable in bridge, but now I also cannot run uuid_limit on b-leg because there's no b-leg channel before bridge called. - If I'm write a script which handles channel init event, checking some variable, and after this, executes uuid_limit on it, there's some other problem - events working in async way, so user's phone will ring before uuid_limit action will be executed (after that user will have +1 missed call, which is not must be here). So the last way to implement this cases is to write module for FS (C or .Net) which will handle execute_on_init, variable, and executes action when channel initializing. But, maybe there any other way to implement this functionality. Or may be execute_on_init shall be added to FS core? -- With best regards, Arthur Kraev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150128/997c37e7/attachment.html From hkalyoncu at gmail.com Wed Jan 28 11:53:30 2015 From: hkalyoncu at gmail.com (huseyin kalyoncu) Date: Wed, 28 Jan 2015 10:53:30 +0200 Subject: [Freeswitch-users] inband dtmf In-Reply-To: References: Message-ID: i found the following statement on freeswitch wiki: "*DTMF intercept w/ DTMF detection, removal and regeneration* Detect DTMF using Goertzel and drop samples identified as containing DTMF tones. Regenerate the detected DTMF tones on the opposite leg. *This AFAIK is the only DTMF intercept mode supported by FreeSWITCH ATM.**"* according to this can we convert outband(rfc2833) dtmf to inband dtmf? On Mon, Jan 26, 2015 at 3:26 PM, huseyin kalyoncu wrote: > hello > > first i want to thank the developers and contributors of this amazing > product. > we have been using freeswitch for almost 4 years without a major problem. > > i have a question regarding inband dtmf. > we have receiving calls from telco using dtmf rfc2833. most of outgoing > calls also > dtmf rfc2833. but we have a new outgoing profile which is behind a > firewall. we did > not make a successful call with transport UDP. so we set the transport to > TCP and > now we have successful calls but the only problem is with dtmf. when we > set the dtmf to > rfc2833 for this profile, we saw that dtmf packets do not arrive correctly > to outgoing > destination. when we dig up the problem we realized that there is always a > time skew > on dtmf packets. for this reason we tried to set dtmf inband for this > particular outgoing > profile. to accomplish this we used start_dtmf_generate just before the > bridge action. > but this time no dtmf package arrive at destination. is this dtmf > conversion (from rfc2833 > to inband) even possible? > what should be the correct configuration to achieve this? > > thanks > huseyin > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150128/6732c1fe/attachment.html From jpyle at fidelityvoice.com Wed Jan 28 16:29:08 2015 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Wed, 28 Jan 2015 08:29:08 -0500 Subject: [Freeswitch-users] inband dtmf In-Reply-To: References: Message-ID: I've been working on a configuration to solve the same problem. In my case it's a simple in-and-out bridging configuration on a single sofia profile, where the a-leg can be inband or RFC2833 an any number of codecs, but the b-leg must be inband on G.711u. I have the profile configured with dtmf-mode=rfc2833 as a default. Then, in the dialplan: The first extension seems to do a decent job at detecting whether or not RFC2833-style DTMF is present in the SDP ("telephone-event"), and if so: - start_dtmf_generate: This will cause b-leg inband tones when DTMF is detected on the a-leg. This solves the 2833-to-inband (a to b) direction. - export nolocal:execute_on_answer=start_dtmf: This enables DSP detection of inband tones on the b-leg, causing them to relay back to the a-leg as RFC2833 events. - set ringback: Turning on this DTMF stuff causes the a-leg to see a 183 Session Progress, probably due to the media processing. Without setting the ringback, a 180 Ringing from the b-leg doesn't indicate at all on the a-leg. With setting, it's effectively converted into a 183 Session Progress with inband ringback. If the a-leg does not have RFC2833 indicated, I assume it's inband because I don't support DTMF over SIP INFO. In this case the detect-2833 extension's anti-action sets the dtmf-mode appropriately, overriding the 2833 default in the profile. This causes the inband tones to pass straight through FS without any detection or manipulation, which is just fine for my case. The good news is this configuration seems to do everything I've described. The bad news is that it does not mute the inband tones as they travel from b-leg to a-leg. I haven't figured out that piece yet. Suggestions welcome! - Jeff On Wed, Jan 28, 2015 at 3:53 AM, huseyin kalyoncu wrote: > i found the following statement on freeswitch wiki: > > "*DTMF intercept w/ DTMF detection, removal and regeneration* > Detect DTMF using Goertzel and drop samples identified as containing DTMF > tones. Regenerate the detected DTMF tones on the opposite leg. *This > AFAIK is the only DTMF intercept mode supported by FreeSWITCH ATM.**"* > > according to this can we convert outband(rfc2833) dtmf to inband dtmf? > > > On Mon, Jan 26, 2015 at 3:26 PM, huseyin kalyoncu > wrote: > >> hello >> >> first i want to thank the developers and contributors of this amazing >> product. >> we have been using freeswitch for almost 4 years without a major problem. >> >> i have a question regarding inband dtmf. >> we have receiving calls from telco using dtmf rfc2833. most of outgoing >> calls also >> dtmf rfc2833. but we have a new outgoing profile which is behind a >> firewall. we did >> not make a successful call with transport UDP. so we set the transport to >> TCP and >> now we have successful calls but the only problem is with dtmf. when we >> set the dtmf to >> rfc2833 for this profile, we saw that dtmf packets do not arrive >> correctly to outgoing >> destination. when we dig up the problem we realized that there is always >> a time skew >> on dtmf packets. for this reason we tried to set dtmf inband for this >> particular outgoing >> profile. to accomplish this we used start_dtmf_generate just before the >> bridge action. >> but this time no dtmf package arrive at destination. is this dtmf >> conversion (from rfc2833 >> to inband) even possible? >> what should be the correct configuration to achieve this? >> >> thanks >> huseyin >> >> > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150128/77ed51f1/attachment-0001.html From brian at freeswitch.org Wed Jan 28 17:26:50 2015 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jan 2015 08:26:50 -0600 Subject: [Freeswitch-users] inband dtmf In-Reply-To: References: Message-ID: You'll want Either in the {} or set as a profile flag, that should fix the issue with time skew. On Wed, Jan 28, 2015 at 7:29 AM, Jeff Pyle wrote: > I've been working on a configuration to solve the same problem. In my > case it's a simple in-and-out bridging configuration on a single sofia > profile, where the a-leg can be inband or RFC2833 an any number of codecs, > but the b-leg must be inband on G.711u. I have the profile configured with > dtmf-mode=rfc2833 as a default. Then, in the dialplan: > > > > > > > > data="nolocal:execute_on_answer=start_dtmf"/> > > > > > > > > > > > > > > The first extension seems to do a decent job at detecting whether or not > RFC2833-style DTMF is present in the SDP ("telephone-event"), and if so: > > - start_dtmf_generate: This will cause b-leg inband tones when DTMF is > detected on the a-leg. This solves the 2833-to-inband (a to b) direction. > > - export nolocal:execute_on_answer=start_dtmf: This enables DSP > detection of inband tones on the b-leg, causing them to relay back to the > a-leg as RFC2833 events. > > - set ringback: Turning on this DTMF stuff causes the a-leg to see a > 183 Session Progress, probably due to the media processing. Without > setting the ringback, a 180 Ringing from the b-leg doesn't indicate at all > on the a-leg. With setting, it's effectively converted into a 183 Session > Progress with inband ringback. > > If the a-leg does not have RFC2833 indicated, I assume it's inband because > I don't support DTMF over SIP INFO. In this case the detect-2833 > extension's anti-action sets the dtmf-mode appropriately, overriding the > 2833 default in the profile. This causes the inband tones to pass straight > through FS without any detection or manipulation, which is just fine for my > case. > > The good news is this configuration seems to do everything I've > described. The bad news is that it does not mute the inband tones as they > travel from b-leg to a-leg. I haven't figured out that piece yet. > Suggestions welcome! > > > - Jeff > > > On Wed, Jan 28, 2015 at 3:53 AM, huseyin kalyoncu > wrote: > >> i found the following statement on freeswitch wiki: >> >> "*DTMF intercept w/ DTMF detection, removal and regeneration* >> Detect DTMF using Goertzel and drop samples identified as containing DTMF >> tones. Regenerate the detected DTMF tones on the opposite leg. *This >> AFAIK is the only DTMF intercept mode supported by FreeSWITCH ATM.**"* >> >> according to this can we convert outband(rfc2833) dtmf to inband dtmf? >> >> >> On Mon, Jan 26, 2015 at 3:26 PM, huseyin kalyoncu >> wrote: >> >>> hello >>> >>> first i want to thank the developers and contributors of this amazing >>> product. >>> we have been using freeswitch for almost 4 years without a major problem. >>> >>> i have a question regarding inband dtmf. >>> we have receiving calls from telco using dtmf rfc2833. most of outgoing >>> calls also >>> dtmf rfc2833. but we have a new outgoing profile which is behind a >>> firewall. we did >>> not make a successful call with transport UDP. so we set the transport >>> to TCP and >>> now we have successful calls but the only problem is with dtmf. when we >>> set the dtmf to >>> rfc2833 for this profile, we saw that dtmf packets do not arrive >>> correctly to outgoing >>> destination. when we dig up the problem we realized that there is always >>> a time skew >>> on dtmf packets. for this reason we tried to set dtmf inband for this >>> particular outgoing >>> profile. to accomplish this we used start_dtmf_generate just before the >>> bridge action. >>> but this time no dtmf package arrive at destination. is this dtmf >>> conversion (from rfc2833 >>> to inband) even possible? >>> what should be the correct configuration to achieve this? >>> >>> thanks >>> huseyin >>> >>> >> >> _________________________________________________________________________ >> 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/20150128/6fc50f21/attachment.html From hkalyoncu at gmail.com Wed Jan 28 17:45:56 2015 From: hkalyoncu at gmail.com (huseyin kalyoncu) Date: Wed, 28 Jan 2015 16:45:56 +0200 Subject: [Freeswitch-users] inband dtmf In-Reply-To: References: Message-ID: thanks for your answers. Jeff, i will try your suggestion. Brian, we tried that already but unfortunately did no effect. also we tried bunch of some other rtp parameters from freeswitch wiki but they also did not solve the problem. On Wed, Jan 28, 2015 at 4:26 PM, Brian West wrote: > You'll want > > > > Either in the {} or set as a profile flag, that should fix the issue with > time skew. > > > > On Wed, Jan 28, 2015 at 7:29 AM, Jeff Pyle > wrote: > >> I've been working on a configuration to solve the same problem. In my >> case it's a simple in-and-out bridging configuration on a single sofia >> profile, where the a-leg can be inband or RFC2833 an any number of codecs, >> but the b-leg must be inband on G.711u. I have the profile configured with >> dtmf-mode=rfc2833 as a default. Then, in the dialplan: >> >> >> >> >> >> >> >> > data="nolocal:execute_on_answer=start_dtmf"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> The first extension seems to do a decent job at detecting whether or not >> RFC2833-style DTMF is present in the SDP ("telephone-event"), and if so: >> >> - start_dtmf_generate: This will cause b-leg inband tones when DTMF is >> detected on the a-leg. This solves the 2833-to-inband (a to b) direction. >> >> - export nolocal:execute_on_answer=start_dtmf: This enables DSP >> detection of inband tones on the b-leg, causing them to relay back to the >> a-leg as RFC2833 events. >> >> - set ringback: Turning on this DTMF stuff causes the a-leg to see a >> 183 Session Progress, probably due to the media processing. Without >> setting the ringback, a 180 Ringing from the b-leg doesn't indicate at all >> on the a-leg. With setting, it's effectively converted into a 183 Session >> Progress with inband ringback. >> >> If the a-leg does not have RFC2833 indicated, I assume it's inband >> because I don't support DTMF over SIP INFO. In this case the detect-2833 >> extension's anti-action sets the dtmf-mode appropriately, overriding the >> 2833 default in the profile. This causes the inband tones to pass straight >> through FS without any detection or manipulation, which is just fine for my >> case. >> >> The good news is this configuration seems to do everything I've >> described. The bad news is that it does not mute the inband tones as they >> travel from b-leg to a-leg. I haven't figured out that piece yet. >> Suggestions welcome! >> >> >> - Jeff >> >> >> On Wed, Jan 28, 2015 at 3:53 AM, huseyin kalyoncu >> wrote: >> >>> i found the following statement on freeswitch wiki: >>> >>> "*DTMF intercept w/ DTMF detection, removal and regeneration* >>> Detect DTMF using Goertzel and drop samples identified as containing >>> DTMF tones. Regenerate the detected DTMF tones on the opposite leg. *This >>> AFAIK is the only DTMF intercept mode supported by FreeSWITCH ATM.**"* >>> >>> according to this can we convert outband(rfc2833) dtmf to inband dtmf? >>> >>> >>> On Mon, Jan 26, 2015 at 3:26 PM, huseyin kalyoncu >>> wrote: >>> >>>> hello >>>> >>>> first i want to thank the developers and contributors of this amazing >>>> product. >>>> we have been using freeswitch for almost 4 years without a major >>>> problem. >>>> >>>> i have a question regarding inband dtmf. >>>> we have receiving calls from telco using dtmf rfc2833. most of >>>> outgoing calls also >>>> dtmf rfc2833. but we have a new outgoing profile which is behind a >>>> firewall. we did >>>> not make a successful call with transport UDP. so we set the transport >>>> to TCP and >>>> now we have successful calls but the only problem is with dtmf. when we >>>> set the dtmf to >>>> rfc2833 for this profile, we saw that dtmf packets do not arrive >>>> correctly to outgoing >>>> destination. when we dig up the problem we realized that there is >>>> always a time skew >>>> on dtmf packets. for this reason we tried to set dtmf inband for this >>>> particular outgoing >>>> profile. to accomplish this we used start_dtmf_generate just before the >>>> bridge action. >>>> but this time no dtmf package arrive at destination. is this dtmf >>>> conversion (from rfc2833 >>>> to inband) even possible? >>>> what should be the correct configuration to achieve this? >>>> >>>> thanks >>>> huseyin >>>> >>>> >>> >>> _________________________________________________________________________ >>> 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 > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150128/8cff328d/attachment-0001.html From treitinger at as-infodienste.de Wed Jan 28 18:55:30 2015 From: treitinger at as-infodienste.de (Melanie Treitinger) Date: Wed, 28 Jan 2015 16:55:30 +0100 Subject: [Freeswitch-users] SRTP with AES 256 Message-ID: <54C90672.2050506@as-infodienste.de> Hi, I'm trying to make sip calls with encryption. Everything works fine with AES 128, but with AES 256 there is no audio but also no error reported. How do I use AES 256? Is it implemented in FreeSWITCH already? Thanks Melanie From brian at freeswitch.org Wed Jan 28 19:01:32 2015 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jan 2015 10:01:32 -0600 Subject: [Freeswitch-users] SRTP with AES 256 In-Reply-To: <54C90672.2050506@as-infodienste.de> References: <54C90672.2050506@as-infodienste.de> Message-ID: Yes, First off what revision of FreeSWITCH are you using? And what endpoints are you working with? On Wed, Jan 28, 2015 at 9:55 AM, Melanie Treitinger < treitinger at as-infodienste.de> wrote: > Hi, I'm trying to make sip calls with encryption. > Everything works fine with AES 128, but with AES 256 there is no audio > but also no error reported. > > How do I use AES 256? Is it implemented in FreeSWITCH already? > > Thanks > Melanie > > _________________________________________________________________________ > 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/20150128/d7eb4e00/attachment.html From auge at virtues.net Wed Jan 28 19:03:23 2015 From: auge at virtues.net (Thomas Auge) Date: Wed, 28 Jan 2015 13:03:23 -0300 Subject: [Freeswitch-users] Audio levels between webRTC and Softphone very different Message-ID: <54C9084B.7090802@virtues.net> We're seeing an issue where the audio level from our webRTC app when connecting to softphones (and some hardware devices) via SIP though freeswitch arrives much higher at the other end than it should be. The other way around the levels from the phone to webRTC are way too low. Does anyone have an idea what could be causing this? From msc at freeswitch.org Wed Jan 28 19:40:59 2015 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Jan 2015 08:40:59 -0800 Subject: [Freeswitch-users] can I use Event_Socket_Outbound to only monitor? In-Reply-To: References: Message-ID: If you want to look at events involving a specific user as opposed to a specific uuid then I recommend that you first turn on all events and see what header(s) they have in common and go from there. You'll need to make good use of the filter command. The wiki has a good intro: https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket#mod_event_socket-filter If you want more details then check out chapter 4 of the FreeSWITCH Cookbook where events and filtering are more thoroughly discussed. -Michael On Tue, Jan 27, 2015 at 5:06 AM, Shai Perelman wrote: > Hi, thanks, as far as I can see even with the async option it still waits > for commands from my side and the call is not transfered to music on hold > which is what I am dialing, the moment I remove the socket action it > returns to normal behaviour. > > so I am trying this from the other end as Kamil suggested, with inbound > event socket. it works and the call proceeds while I am getting the events > but I need > to filter the events I am getting to only include the events with the > extension(s) that I want to monitor involved. can somebody help me with > what filter command(s) do I need to send to see only the events that have > for example extension 9999 involved? > > thanks > Shai > > On Tue, Jan 27, 2015 at 5:59 AM, Michael Collins > wrote: > >> Using the socket app with the "async" option will cause that channel's >> events to be sent. >> -MC >> >> On Mon, Jan 26, 2015 at 7:43 PM, Kamil Nigmatullin < >> kamil.nigmatullin at gmail.com> wrote: >> >>> Yes if you want monitor and catch events you need to use inbound socket >>> 26 ???. 2015 ?. 21:58 ???????????? "Duvid Rottenberg" < >>> adrottenberg at gmail.com> ???????: >>> >>>> I am pretty sure that with outbound socket as long as you don't send >>>> any commands back to Freeswitch, it will just execute whatever is in the >>>> xml dialplan. >>>> >>>> Thanks, >>>> Duvid Rottenberg >>>> >>>> On Sun, Jan 25, 2015 at 6:19 PM, Michael Collins >>>> wrote: >>>> >>>>> In event socket outbound the listening server controls the call. In >>>>> your case you just want to get the events associated with that call? I've >>>>> never actually tried it but maybe you could do something like: >>>>> >>>>> >>>>> >>>>> >>>>> And then after the socket app answers have it issue the linger >>>>> command, then transfer or execute_extension to send the call off to do >>>>> something. Watch your socket connection for events and see happens. If you >>>>> get it to work please let us know. :) >>>>> >>>>> -Michael >>>>> >>>>> >>>>> >>>>> On Sun, Jan 25, 2015 at 7:17 AM, Shai Perelman >>>>> wrote: >>>>> >>>>>> >>>>>> hi, I am using >>>>>> https://wiki.freeswitch.org/wiki/Event_Socket_Outbound >>>>>> my goal is to monitor what happens on the extension but without >>>>>> affecting the normal behaviour. >>>>>> >>>>>> I have this dialplan. >>>>>> >>>>>> TagTypeDataOrder[image: add] >>>>>> >>>>>> condition username 9998 1[image: edit] >>>>>> >>>>>> [image: delete] >>>>>> >>>>>> action socket 109.65.149.22:9999 async 3[image: edit] >>>>>> >>>>>> [image: delete] >>>>>> >>>>>> >>>>>> >>>>>> I am calling *9664 (music on hold) >>>>>> and the call is blocking while wating for commands from my side. >>>>>> >>>>>> I want it to answer automatticaly like it would do with out the >>>>>> socket module, and just send me the events regarding that extension >>>>>> >>>>>> thanks >>>>>> Shai >>>>>> ITD Communications >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >> >> >> _________________________________________________________________________ >> 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 >> > > > > -- > www.groyse.com > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150128/ce1c5aa2/attachment-0001.html From mike at jerris.com Wed Jan 28 19:55:55 2015 From: mike at jerris.com (Michael Jerris) Date: Wed, 28 Jan 2015 10:55:55 -0600 Subject: [Freeswitch-users] Audio levels between webRTC and Softphone very different In-Reply-To: <54C9084B.7090802@virtues.net> References: <54C9084B.7090802@virtues.net> Message-ID: <0BF02702-82E6-45E8-A143-28BB3A322A05@jerris.com> Check AGC settings in your browser app. > On Jan 28, 2015, at 10:03 AM, Thomas Auge wrote: > > We're seeing an issue where the audio level from our webRTC app when connecting to softphones (and some hardware > devices) via SIP though freeswitch arrives much higher at the other end than it should be. The other way around the > levels from the phone to webRTC are way too low. > > Does anyone have an idea what could be causing this? From msc at freeswitch.org Wed Jan 28 20:02:13 2015 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Jan 2015 09:02:13 -0800 Subject: [Freeswitch-users] Cannot contact registered endpoint behind NAT since latest make current In-Reply-To: <19B3EC9F-DD9A-45C7-99D0-0A2676D5D6C2@kavun.ch> References: <19B3EC9F-DD9A-45C7-99D0-0A2676D5D6C2@kavun.ch> Message-ID: Did you do a SIP trace to see what traffic, if any, is sent to the phone? -Michael On Tue, Jan 27, 2015 at 8:01 AM, Emrah wrote: > Hi all, > Since my make current of this week, I cannot bridge to registered > endpoints that are seen by my FS as devices behind NAT. > I tried bridging both with user/aurora605 at domain as well as using the > full sofia_contact value. > Here is the output I get: > Cannot locate registered user aurora605 at 192.168.2.191 > ;received=1.2.3.4:34146;fs_nat=yes;fs_path=sip at 3Aaurora605 > %401.2.3.4%3A34146 > this is the output of sofia_contact > sofia/internal/sip:aurora605 at 192.168.2.191;received=1.2.3.4:1029 > ;fs_nat=yes;fs_path=sip%3Aaurora605%401.2.3.4%3A1029 > > What are your suggestions to fix this issue? > > Best, > Emrah > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150128/045f0571/attachment.html From msc at freeswitch.org Wed Jan 28 20:04:36 2015 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Jan 2015 09:04:36 -0800 Subject: [Freeswitch-users] sip trace In-Reply-To: <0a1b01d03a94$5991e9d0$0cb5bd70$@comcast.net> References: <0a1b01d03a94$5991e9d0$0cb5bd70$@comcast.net> Message-ID: Just curious - what's the application here? Debugging? Historical logging? Or ... ? -MC On Tue, Jan 27, 2015 at 4:49 PM, Andrew wrote: > Can FreeSWITCH export a sip trace like XML CDR? > > > > Andrew > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150128/444d9133/attachment.html From auge at virtues.net Wed Jan 28 20:06:08 2015 From: auge at virtues.net (Thomas Auge) Date: Wed, 28 Jan 2015 14:06:08 -0300 Subject: [Freeswitch-users] Audio levels between webRTC and Softphone very different In-Reply-To: <0BF02702-82E6-45E8-A143-28BB3A322A05@jerris.com> References: <54C9084B.7090802@virtues.net> <0BF02702-82E6-45E8-A143-28BB3A322A05@jerris.com> Message-ID: <54C91700.6010902@virtues.net> > Check AGC settings in your browser app. It's all off. In the softphones it's more a "felt" difference. So far we only have an accurate measure from a Comrex Access. Strange thing is, even the miliwatt from freeswitch comes in at 10dBu there. At first I blamed it on the comrex, but then I remembered levels being annoyingly high on softphones, too. Really just fishing for ideas here ... :-) From aqsyounas at gmail.com Wed Jan 28 20:53:28 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 28 Jan 2015 22:53:28 +0500 Subject: [Freeswitch-users] Can't set/get values using mod_redis Message-ID: Hi, list. I have successfully installed redis-server on my system and mod_redis on freeswitch. But when i try to set a value with redis in my default.xml, i see nothing. This is my default.xml file. * * And what I see in my logs is this. 2015-01-28 22:48:19.542161 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/14048002020 at 1003] has been answered 2015-01-28 22:48:19.542161 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/14048002020 at 1003 [BREAK] 2015-01-28 22:48:19.542161 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/14048002020 at 1003 [BREAK] 2015-01-28 22:48:19.542161 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/14048002020 at 1003 [BREAK] 2015-01-28 22:48:19.562178 [DEBUG] switch_channel.c:3689 (sofia/internal/ 14048002020 at 1003) Callstate Change EARLY -> ACTIVE 2015-01-28 22:48:19.562178 [DEBUG] sofia.c:6614 Channel sofia/internal/ 14048002020 at 1003 entering state [ready][200] EXECUTE sofia/internal/14048002020 at 1003 set(max_forwards=100) 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 sofia/internal/ 14048002020 at 1003 SET [max_forwards]=[100] EXECUTE sofia/internal/14048002020 at 1003 set(dst=19292461002) 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 sofia/internal/ 14048002020 at 1003 SET [dst]=[19292461002] EXECUTE sofia/internal/14048002020 at 1003 log(c63b137d-4553-41ed-a4e8-fff431889c40) 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1628 c63b137d-4553-41ed-a4e8-fff431889c40 EXECUTE sofia/internal/14048002020 at 1003 set(src=14048002020) 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 sofia/internal/14048002020 at 1003 SET [src]=[14048002020] *EXECUTE sofia/internal/14048002020 at 1003 set(ignore=)* 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 sofia/internal/14048002020 at 1003 SET [ignore]=[UNDEF] *EXECUTE sofia/internal/14048002020 at 1003 set(ignore=)* 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 sofia/internal/14048002020 at 1003 SET [ignore]=[UNDEF] *EXECUTE sofia/internal/14048002020 at 1003 log()* 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1628 I see nothing. Neither a value being set or an error. Your help would be much appreciated. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150128/bb274324/attachment.html From william.king at quentustech.com Wed Jan 28 21:06:23 2015 From: william.king at quentustech.com (William King) Date: Wed, 28 Jan 2015 12:06:23 -0600 Subject: [Freeswitch-users] Can't set/get values using mod_redis In-Reply-To: References: Message-ID: <54C9251F.6090903@quentustech.com> The current mod_redis implementation does not support arbitrary redis commands. It was build as a channel limit(with only increment and decrement functions). The functionality it appears you are trying to use would require additional feature development. For instance using the hiredis C library and supporting arbitrary redis commands. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 01/28/2015 11:53 AM, Aqs Younas wrote: > Hi, list. > > I have successfully installed redis-server on my system and mod_redis on > freeswitch. > But when i try to set a value with redis in my default.xml, i see > nothing. This is my default.xml file. > > > > > > > > > > > > * > > * > > > > > > > And what I see in my logs is this. > > 2015-01-28 22:48:19.542161 [NOTICE] mod_dptools.c:1258 Channel > [sofia/internal/14048002020 @1003] has been answered > 2015-01-28 22:48:19.542161 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/14048002020 @1003 [BREAK] > 2015-01-28 22:48:19.542161 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/14048002020 @1003 [BREAK] > 2015-01-28 22:48:19.542161 [DEBUG] switch_core_session.c:1053 Send > signal sofia/internal/14048002020 @1003 [BREAK] > 2015-01-28 22:48:19.562178 [DEBUG] switch_channel.c:3689 > (sofia/internal/14048002020 @1003) Callstate Change > EARLY -> ACTIVE > 2015-01-28 22:48:19.562178 [DEBUG] sofia.c:6614 Channel > sofia/internal/14048002020 @1003 entering state > [ready][200] > EXECUTE sofia/internal/14048002020 @1003 > set(max_forwards=100) > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > sofia/internal/14048002020 @1003 SET [max_forwards]=[100] > EXECUTE sofia/internal/14048002020 @1003 > set(dst=19292461002 ) > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > sofia/internal/14048002020 @1003 SET [dst]=[19292461002 > ] > EXECUTE sofia/internal/14048002020 at 1003 > log(c63b137d-4553-41ed-a4e8-fff431889c40) > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1628 > c63b137d-4553-41ed-a4e8-fff431889c40 > EXECUTE sofia/internal/14048002020 at 1003 set(src=14048002020) > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > sofia/internal/14048002020 at 1003 SET [src]=[14048002020] > *EXECUTE sofia/internal/14048002020 at 1003 set(ignore=)* > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > sofia/internal/14048002020 at 1003 SET [ignore]=[UNDEF] > *EXECUTE sofia/internal/14048002020 at 1003 set(ignore=)* > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > sofia/internal/14048002020 at 1003 SET [ignore]=[UNDEF]* > EXECUTE sofia/internal/14048002020 at 1003 log()* > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1628 > > I see nothing. Neither a value being set or an error. > > Your help would be much appreciated. > Thanks. > > > > > _________________________________________________________________________ > 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 > From ira at connectmevoice.com Wed Jan 28 22:35:30 2015 From: ira at connectmevoice.com (Ira Tessler) Date: Wed, 28 Jan 2015 14:35:30 -0500 Subject: [Freeswitch-users] Cannot contact registered endpoint behind NAT since latest make current In-Reply-To: References: <19B3EC9F-DD9A-45C7-99D0-0A2676D5D6C2@kavun.ch> Message-ID: This seems to be a bug in the latest master build. I opened a Jira FS-7211 Ira Tessler ConnectMe 732-490-9007 x2 > On Jan 28, 2015, at 12:02 PM, Michael Collins wrote: > > Did you do a SIP trace to see what traffic, if any, is sent to the phone? > -Michael > > On Tue, Jan 27, 2015 at 8:01 AM, Emrah > wrote: > Hi all, > Since my make current of this week, I cannot bridge to registered endpoints that are seen by my FS as devices behind NAT. > I tried bridging both with user/aurora605 at domain as well as using the full sofia_contact value. > Here is the output I get: > Cannot locate registered user aurora605 at 192.168.2.191 ;received=1.2.3.4:34146;fs_nat=yes;fs_path=sip at 3Aaurora605%401.2.3.4%3A34146 > this is the output of sofia_contact > sofia/internal/sip:aurora605 at 192.168.2.191 ;received=1.2.3.4:1029;fs_nat=yes;fs_path=sip%3Aaurora605%401.2.3.4%3A1029 > > What are your suggestions to fix this issue? > > Best, > Emrah > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150128/0c678dac/attachment.html From gb at cm.nl Thu Jan 29 10:10:25 2015 From: gb at cm.nl (Grant Bagdasarian) Date: Thu, 29 Jan 2015 07:10:25 +0000 Subject: [Freeswitch-users] APR issue when configuring In-Reply-To: References: <31690C3C-FF53-4BD3-B738-352E71396CCA@jerris.com> Message-ID: Thanks for the heads up! I wasn?t aware of the APT repository, thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Tuesday, January 27, 2015 12:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring It'd be better to keep source and build separate. You could build in /usr/src/freeswitch and install to /opt/freeswitch for example. /usr/src shouldn't really be used for running a daemon. Since you're using Debian are you aware of the APT repository on files.freeswitch.org? That installs it to standard directories you would expect (/usr/bin/ /usr/sbin/ /etc/freeswitch and so on). On 27 January 2015 at 07:14, Grant Bagdasarian > wrote: Its ?prefix with a single -. Yes, the directory is /usr/src/freeswitch. I always use this parameter to install all the files in the same directory, else it installs in /usr/local/freeswitch I believe. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Monday, January 26, 2015 9:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring Also is that -prefix or --prefix you're using? On 26 January 2015 at 20:21, Michael Jerris > wrote: is /usr/src/freeswitch your source directory too? Its a bit weird to be specifying that as prefix, maybe its an issue when specifying a prefix as the same as your source dir? On Jan 26, 2015, at 12:51 AM, Grant Bagdasarian > wrote: Hello, I just extracted the tarball, installed a few missing dependencies and ran the configure command: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Friday, January 23, 2015 4:44 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring I?m inclined to believe you have something broken on your box here. I just grabbed the tarball and built it right on my Wheezy box... How exactly are you trying to build it from the tarball? On 1/23/15, 8:49 AM, "Grant Bagdasarian" > wrote: lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 7.8 (wheezy) Release: 7.8 Codename: wheezy uname -a Linux HOSTNAME 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Friday, January 23, 2015 3:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] APR issue when configuring On what platform are you getting this error? Sent from my iPhone On Jan 23, 2015, at 6:12 AM, Grant Bagdasarian > wrote: Nevermind. I used the 1.4.15 tarball which contained this error. Now I?ve used the git clone master which works fine. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian Sent: Friday, January 23, 2015 11:51 AM To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) Subject: [Freeswitch-users] APR issue when configuring Hello, I?m getting the following error during configure. checking for APR... configure: error: the --with-apr parameter is incorrect. It must specify an install prefix, a build directory, or an apr-config file. configure: error: ./configure.gnu failed for libs/apr-util I?m running the following command to configure: ./configure -prefix=/usr/src/freeswitch/ --enable-core-odbc-support Also, a lot of directories inside the libs require the configure script to be executable. I have to set this 1 by 1. Is there a way to disable apr? Or is it required by FS? _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/05744702/attachment-0001.html From treitinger at as-infodienste.de Thu Jan 29 10:53:54 2015 From: treitinger at as-infodienste.de (Melanie Treitinger) Date: Thu, 29 Jan 2015 08:53:54 +0100 Subject: [Freeswitch-users] SRTP with AES 256 In-Reply-To: References: <54C90672.2050506@as-infodienste.de> Message-ID: <54C9E712.2060107@as-infodienste.de> I'm using Version 1.5.15b git d199060 2015-01-09 00:01:19Z 32bit Endpoints are Innovaphone IP810 telephone system, registered as gateway, and Innovaphone IP241 phone wich is calling e.g. Tetris via this gateway. In my dialplan I have set With AES128, I can hear Tetris. When I change this to "AES_CM_256_HMAC_SHA1_80", which I would prefer, I hear nothing. In the file mime.types I see only "audio/rtp-enc-aescm128". Do you need more information? Am 28.01.2015 um 17:01 schrieb Brian West: > Yes, First off what revision of FreeSWITCH are you using? And what > endpoints are you working with? > > On Wed, Jan 28, 2015 at 9:55 AM, Melanie Treitinger > > wrote: > > Hi, I'm trying to make sip calls with encryption. > Everything works fine with AES 128, but with AES 256 there is no audio > but also no error reported. > > How do I use AES 256? Is it implemented in FreeSWITCH already? > > Thanks > Melanie > > _________________________________________________________________________ > 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 > > > > E-Mail ist virenfrei. > Von AVG ?berpr?ft - www.avg.de > Version: 2013.0.3495 / Virendatenbank: 4257/9014 - Ausgabedatum: 28.01.2015 > From giggsey at gmail.com Thu Jan 29 12:20:48 2015 From: giggsey at gmail.com (Joshua Gigg) Date: Thu, 29 Jan 2015 09:20:48 +0000 Subject: [Freeswitch-users] Update CLI on Freeswitch originated call Message-ID: Hi, Is it possible to update the CLI at will once a Freeswitch originated call has been answered? I know it can update during a transfer, but I want to be able to control it directly myself. -- Joshua Gigg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/9b1540a5/attachment.html From bpriddy at bryantschools.org Thu Jan 29 17:28:14 2015 From: bpriddy at bryantschools.org (Blake Priddy) Date: Thu, 29 Jan 2015 08:28:14 -0600 Subject: [Freeswitch-users] Porting Message-ID: Hi all! I was wondering what you have experienced with flowroute and their porting? I love their service but it seems when I try to port a number over to them from someone who wants to get away from their current provider. Flowroute doesn't service the area.. Is there a provider I can port a number to then have flowroute get the number from them? Like a proxy somewhat. I have schools that are wanting to switch to freeswitch and I always have to do remote call forwarding on their line. So they pay flowroute and still their current provider for the RCF feature. Just trying to get advice on what you all have had experiences with and gather any ideas if I can. Thanks! -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/975a8257/attachment.html From krice at freeswitch.org Thu Jan 29 17:35:08 2015 From: krice at freeswitch.org (Ken Rice) Date: Thu, 29 Jan 2015 08:35:08 -0600 Subject: [Freeswitch-users] Porting In-Reply-To: Message-ID: There in lies the problem with VoIP in Rural Areas... If they have a small rural ILEC there good luck on porting them... What you?ll end up having to do is either get that LEC to deliver it as SIP or use a TDM -> SIP gateway of some sort On 1/29/15, 8:28 AM, "Blake Priddy" wrote: > Hi all! I was wondering what you have experienced with flowroute and their > porting? I love their service but it seems when I try to port a number over to > them from someone who wants to get away from their current provider. Flowroute > doesn't service the area.. Is there a provider I can port a number to then > have flowroute get the number from them? Like a proxy somewhat. I have schools > that are wanting to switch to freeswitch and I always have to do remote call > forwarding on their line. So they pay flowroute and still their current > provider for the RCF feature. Just trying to get advice on what you all have > had experiences with and gather any ideas if I can.? > > Thanks! > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/a6ec9f64/attachment.html From brian at freeswitch.org Thu Jan 29 17:49:31 2015 From: brian at freeswitch.org (Brian West) Date: Thu, 29 Jan 2015 08:49:31 -0600 Subject: [Freeswitch-users] SRTP with AES 256 In-Reply-To: <54C9E712.2060107@as-infodienste.de> References: <54C90672.2050506@as-infodienste.de> <54C9E712.2060107@as-infodienste.de> Message-ID: Those phones probably do not support it, I'm unaware of any endpoints currently other than freeswitch that can do big AES. On Thu, Jan 29, 2015 at 1:53 AM, Melanie Treitinger < treitinger at as-infodienste.de> wrote: > I'm using Version 1.5.15b git d199060 2015-01-09 00:01:19Z 32bit > > Endpoints are Innovaphone IP810 telephone system, registered as gateway, > and Innovaphone IP241 phone wich is calling e.g. Tetris via this gateway. > > In my dialplan I have set > data="rtp_secure_media=mandatory:AES_CM_128_HMAC_SHA1_80,AES_CM_128_HMAC_SHA1_32"/> > With AES128, I can hear Tetris. > > When I change this to "AES_CM_256_HMAC_SHA1_80", which I would prefer, I > hear nothing. > > In the file mime.types I see only "audio/rtp-enc-aescm128". > > Do you need more information? > > > Am 28.01.2015 um 17:01 schrieb Brian West: > > Yes, First off what revision of FreeSWITCH are you using? And what > > endpoints are you working with? > > > > On Wed, Jan 28, 2015 at 9:55 AM, Melanie Treitinger > > > > wrote: > > > > Hi, I'm trying to make sip calls with encryption. > > Everything works fine with AES 128, but with AES 256 there is no > audio > > but also no error reported. > > > > How do I use AES 256? Is it implemented in FreeSWITCH already? > > > > Thanks > > Melanie > > > > > _________________________________________________________________________ > > 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 > > > > > > > > E-Mail ist virenfrei. > > Von AVG ?berpr?ft - www.avg.de > > Version: 2013.0.3495 / Virendatenbank: 4257/9014 - Ausgabedatum: > 28.01.2015 > > > > _________________________________________________________________________ > 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/20150129/5c01033b/attachment-0001.html From kworm at sofnet.com Thu Jan 29 17:51:38 2015 From: kworm at sofnet.com (Kevin Wormington) Date: Thu, 29 Jan 2015 08:51:38 -0600 Subject: [Freeswitch-users] Porting In-Reply-To: References: Message-ID: <54CA48FA.2010104@sofnet.com> The problem isn't in porting the actual number itself...it is the underlying carrier(s) of Flowroute (or any other provider) not having some type of trunking connecting to the rate center or tandem you are trying to port the number from along with an LRN for that rate center. Along with other issues like 911/E911 support, etc. especially if it's rural. The costs to a carrier could be several hundred dollars per month to get into a rate center so it's difficult when it's just a few numbers. On 01/29/2015 08:35 AM, Ken Rice wrote: > There in lies the problem with VoIP in Rural Areas... If they have a > small rural ILEC there good luck on porting them... What you?ll end up > having to do is either get that LEC to deliver it as SIP or use a TDM -> > SIP gateway of some sort > > > On 1/29/15, 8:28 AM, "Blake Priddy" wrote: > > Hi all! I was wondering what you have experienced with flowroute and > their porting? I love their service but it seems when I try to port > a number over to them from someone who wants to get away from their > current provider. Flowroute doesn't service the area.. Is there a > provider I can port a number to then have flowroute get the number > from them? Like a proxy somewhat. I have schools that are wanting to > switch to freeswitch and I always have to do remote call forwarding > on their line. So they pay flowroute and still their current > provider for the RCF feature. Just trying to get advice on what you > all have had experiences with and gather any ideas if I can. > > Thanks! > > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > > > _________________________________________________________________________ > 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 > From mvar78 at gmail.com Thu Jan 29 17:54:01 2015 From: mvar78 at gmail.com (Massimo Varriale) Date: Thu, 29 Jan 2015 15:54:01 +0100 Subject: [Freeswitch-users] Porting In-Reply-To: References: Message-ID: <5171B3B7-87AD-411D-A9C2-52D8C555F1E3@gmail.com> Hi Blake, I don't have direct experience with flowroute but as far as I can see the situation you are facing is the same with some italian providers... In situation where I'm not able to port the number to another provider I'm using an FXO converter (Could be a Linksys/Cisco SPA 3000) to get analog lines and to redirect them to my FS box. In my FS I have an incoming configuration (that is coming from the ATA) and several different outbound Gateways so I do not need to pay for any forward to get incoming calls and also I'm not paying too expenses bill from the provider because I only keep it for incoming calls and I'm using cheaper Gateways for outgoing calls.. I hope this could help Cheers Max Il giorno 29/gen/2015, alle ore 15:35, Ken Rice ha scritto: > There in lies the problem with VoIP in Rural Areas... If they have a small rural ILEC there good luck on porting them... What you?ll end up having to do is either get that LEC to deliver it as SIP or use a TDM -> SIP gateway of some sort > > > On 1/29/15, 8:28 AM, "Blake Priddy" wrote: > >> Hi all! I was wondering what you have experienced with flowroute and their porting? I love their service but it seems when I try to port a number over to them from someone who wants to get away from their current provider. Flowroute doesn't service the area.. Is there a provider I can port a number to then have flowroute get the number from them? Like a proxy somewhat. I have schools that are wanting to switch to freeswitch and I always have to do remote call forwarding on their line. So they pay flowroute and still their current provider for the RCF feature. Just trying to get advice on what you all have had experiences with and gather any ideas if I can. >> >> Thanks! >> >> > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/c54df35b/attachment.html From treitinger at as-infodienste.de Thu Jan 29 18:04:18 2015 From: treitinger at as-infodienste.de (Melanie Treitinger) Date: Thu, 29 Jan 2015 16:04:18 +0100 Subject: [Freeswitch-users] SRTP with AES 256 In-Reply-To: References: <54C90672.2050506@as-infodienste.de> <54C9E712.2060107@as-infodienste.de> Message-ID: <54CA4BF2.5090105@as-infodienste.de> Yes, they can use AES 256. We have been using this encryption already without freeswitch. See http://wiki.innovaphone.com/index.php?title=Howto:SIPS_will_work_with_V7 Am 29.01.2015 um 15:49 schrieb Brian West: > Those phones probably do not support it, I'm unaware of any endpoints > currently other than freeswitch that can do big AES. > > On Thu, Jan 29, 2015 at 1:53 AM, Melanie Treitinger > > wrote: > > I'm using Version 1.5.15b git d199060 2015-01-09 00:01:19Z 32bit > > Endpoints are Innovaphone IP810 telephone system, registered as gateway, > and Innovaphone IP241 phone wich is calling e.g. Tetris via this > gateway. > > In my dialplan I have set data="rtp_secure_media=mandatory:AES_CM_128_HMAC_SHA1_80,AES_CM_128_HMAC_SHA1_32"/> > With AES128, I can hear Tetris. > > When I change this to "AES_CM_256_HMAC_SHA1_80", which I would prefer, I > hear nothing. > > In the file mime.types I see only "audio/rtp-enc-aescm128". > > Do you need more information? > > > Am 28.01.2015 um 17:01 schrieb Brian West: > > Yes, First off what revision of FreeSWITCH are you using? And what > > endpoints are you working with? > > > > On Wed, Jan 28, 2015 at 9:55 AM, Melanie Treitinger > > > >> wrote: > > > > Hi, I'm trying to make sip calls with encryption. > > Everything works fine with AES 128, but with AES 256 there is no audio > > but also no error reported. > > > > How do I use AES 256? Is it implemented in FreeSWITCH already? > > > > Thanks > > Melanie > > > > _________________________________________________________________________ > > 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 > > > > > > > > E-Mail ist virenfrei. > > Von AVG ?berpr?ft - www.avg.de > > > Version: 2013.0.3495 / Virendatenbank: 4257/9014 - Ausgabedatum: > 28.01.2015 > > > > _________________________________________________________________________ > 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 > > > > E-Mail ist virenfrei. > Von AVG ?berpr?ft - www.avg.de > Version: 2013.0.3495 / Virendatenbank: 4257/9020 - Ausgabedatum: 29.01.2015 > From msc at freeswitch.org Thu Jan 29 20:58:54 2015 From: msc at freeswitch.org (Michael Collins) Date: Thu, 29 Jan 2015 09:58:54 -0800 Subject: [Freeswitch-users] Update CLI on Freeswitch originated call In-Reply-To: References: Message-ID: Could you expound upon this question a bit? What does "update the CLI" mean? Thanks, MC On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg wrote: > Hi, > > Is it possible to update the CLI at will once a Freeswitch originated call > has been answered? > > I know it can update during a transfer, but I want to be able to control > it directly myself. > > -- > Joshua Gigg > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/c2a2cb3b/attachment.html From aqsyounas at gmail.com Thu Jan 29 21:43:22 2015 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 29 Jan 2015 23:43:22 +0500 Subject: [Freeswitch-users] Can't set/get values using mod_redis In-Reply-To: <54C9251F.6090903@quentustech.com> References: <54C9251F.6090903@quentustech.com> Message-ID: Thanks for your reply. I used mod_memcache for above purpose. On 28 January 2015 at 23:06, William King wrote: > The current mod_redis implementation does not support arbitrary redis > commands. It was build as a channel limit(with only increment and > decrement functions). > > The functionality it appears you are trying to use would require > additional feature development. For instance using the hiredis C library > and supporting arbitrary redis commands. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 01/28/2015 11:53 AM, Aqs Younas wrote: > > Hi, list. > > > > I have successfully installed redis-server on my system and mod_redis on > > freeswitch. > > But when i try to set a value with redis in my default.xml, i see > > nothing. This is my default.xml file. > > > > > > > > > > > > > > > > > > data="dst=${destination_number}"/> > > > > data="src=${caller_id_number}"/> > > * > > > > * > > data="vlc:///opt/song.mp3"/> > > > > > > > > > > > > And what I see in my logs is this. > > > > 2015-01-28 22:48:19.542161 [NOTICE] mod_dptools.c:1258 Channel > > [sofia/internal/14048002020 @1003] has been answered > > 2015-01-28 22:48:19.542161 [DEBUG] switch_core_session.c:1053 Send > > signal sofia/internal/14048002020 @1003 [BREAK] > > 2015-01-28 22:48:19.542161 [DEBUG] switch_core_session.c:1053 Send > > signal sofia/internal/14048002020 @1003 [BREAK] > > 2015-01-28 22:48:19.542161 [DEBUG] switch_core_session.c:1053 Send > > signal sofia/internal/14048002020 @1003 [BREAK] > > 2015-01-28 22:48:19.562178 [DEBUG] switch_channel.c:3689 > > (sofia/internal/14048002020 @1003) Callstate Change > > EARLY -> ACTIVE > > 2015-01-28 22:48:19.562178 [DEBUG] sofia.c:6614 Channel > > sofia/internal/14048002020 @1003 entering state > > [ready][200] > > EXECUTE sofia/internal/14048002020 @1003 > > set(max_forwards=100) > > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > > sofia/internal/14048002020 @1003 SET > [max_forwards]=[100] > > EXECUTE sofia/internal/14048002020 @1003 > > set(dst=19292461002 ) > > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > > sofia/internal/14048002020 @1003 SET [dst]=[19292461002 > > ] > > EXECUTE sofia/internal/14048002020 at 1003 > > log(c63b137d-4553-41ed-a4e8-fff431889c40) > > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1628 > > c63b137d-4553-41ed-a4e8-fff431889c40 > > EXECUTE sofia/internal/14048002020 at 1003 set(src=14048002020) > > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > > sofia/internal/14048002020 at 1003 SET [src]=[14048002020] > > *EXECUTE sofia/internal/14048002020 at 1003 set(ignore=)* > > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > > sofia/internal/14048002020 at 1003 SET [ignore]=[UNDEF] > > *EXECUTE sofia/internal/14048002020 at 1003 set(ignore=)* > > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1435 > > sofia/internal/14048002020 at 1003 SET [ignore]=[UNDEF]* > > EXECUTE sofia/internal/14048002020 at 1003 log()* > > 2015-01-28 22:48:19.562178 [DEBUG] mod_dptools.c:1628 > > > > I see nothing. Neither a value being set or an error. > > > > Your help would be much appreciated. > > Thanks. > > > > > > > > > > _________________________________________________________________________ > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/2937fda9/attachment-0001.html From giggsey at gmail.com Thu Jan 29 23:06:22 2015 From: giggsey at gmail.com (Joshua Gigg) Date: Thu, 29 Jan 2015 20:06:22 +0000 Subject: [Freeswitch-users] Update CLI on Freeswitch originated call In-Reply-To: References: Message-ID: When a transfer completes, FreeSWITCH will send an UPDATE message to the SIP server updating the caller id. Is there a way of making FreeSWITCH send this message via a dialplan/command? On 29 January 2015 at 17:58, Michael Collins wrote: > Could you expound upon this question a bit? What does "update the CLI" > mean? > Thanks, > MC > > On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg wrote: > >> Hi, >> >> Is it possible to update the CLI at will once a Freeswitch originated >> call has been answered? >> >> I know it can update during a transfer, but I want to be able to control >> it directly myself. >> >> -- >> Joshua Gigg >> >> _________________________________________________________________________ >> 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 > -- Joshua Gigg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/2fe6356c/attachment.html From you.kissmyarse at gmail.com Fri Jan 30 06:12:42 2015 From: you.kissmyarse at gmail.com (Benjamin Rowe) Date: Thu, 29 Jan 2015 19:12:42 -0800 Subject: [Freeswitch-users] Issues with originate, webrtc and xml_curl Message-ID: Hello All, I have a basic system up and running i am able to dial out via my provider on both sip endpoints. I have setup 2 users 1000 and 1004. 1000 is created from the flat file xml and 1004 is from the xml_curl request. when i attempt to originate a call from user 1000 to an external number it passes through fine however when i originate a call from user 1004 i get subscriber absent both are running the same setup to my knowledge. Any Ideas? Errors: 2015-01-29 18:12:17.745051 [WARNING] mod_dptools.c:3979 Can't find user [1004@] 2015-01-29 18:12:17.745051 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2015-01-29 18:12:17.745051 [DEBUG] switch_ivr_originate.c:3723 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] 2015-01-29 18:12:17.745051 [INFO] mod_dptools.c:3234 Originate Failed. Cause: SUBSCRIBER_ABSENT 2015-01-29 18:12:17.745051 [NOTICE] switch_channel.c:4724 Hangup loopback/1004-b [CS_EXECUTE] [SUBSCRIBER_ABSENT] Sofia Registrations: ================================================================================================= Call-ID: ocq57n8fn23uo0s7rh2025 User: 1004@ Contact: "" %3A59613%3Btransport%3Dws> Agent: SIP.js/0.6.3-devel BB Status: Registered(WS-NAT)(unknown) EXP(2015-01-29 18:24:02) EXPSECS(571) Ping-Status: Reachable Host: fs01-a IP: Port: 59613 Auth-User: 1004 Auth-Realm: MWI-Account: 1004@ Originate command: originate {ignore_early_media=true,origination_caller_id_number=1004}loopback/1004 &bridge(sofia/gateway/test/555123456) Many Thanks Benjamin Rowe -- Benjamin Rowe Lazypeople Mobile: +44 (0)7904 026869 Phone: +44 (0)845 86 99 892 Email: ben at lazypeople.co.uk Web: http://www.lazypeople.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/ea5c1872/attachment.html From nreis at wavecom.pt Fri Jan 30 06:47:40 2015 From: nreis at wavecom.pt (Nuno Reis) Date: Fri, 30 Jan 2015 03:47:40 +0000 Subject: [Freeswitch-users] weird beahivior when using outbound_proxy/fs_path Message-ID: Hello everyone. I've been using outbound_proxy for a sometime now to make SIP go trough a kamailio proxy server. When I put freeswitch under some stress and measure response times, I start seeing a weird behavior. FS starts like "hanging" after a fast sequence of INVITES usually after the first 5 INVITE sipke(not that much right) and hangs for +/- 5 seconds before routing the call out to kamailio. If i tell FS to route the call using fs_path there is no delay at all. Another weird thing about this one is that I have some installs where outbound_proxy works fine with max performance everytime and I'm always using the same install binaries, the same OS the same versions of everything (checked and double checked). You could tell me to use fs_path if it solves the problem and I would say you are right but the thing is fs_path doesn't behave exactly the same as outbound_proxy does in every situation. It's usually common to see the variable destination_number set with the IP address of the proxy and some other annoying issues and above all outbound_proxy works at the SIP profile level while fs_path doesn't which force me to explicitly use it literally everywhere. Any ideas on what could be causing this behavior. It looks like a timer of some kind but I don't see nothing about such a timer anywhere. Looking forward to hear from you. Cheers, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis at wavecom.pt WAVECOM-Solu??es R?dio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS | www.wavecom.pt ** * [image: Description: Description: WavecomSignature] [image: Publicity] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/dcac97e5/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 16423 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/dcac97e5/attachment-0001.png From rogelio.perez at gmail.com Fri Jan 30 06:53:55 2015 From: rogelio.perez at gmail.com (rogelio.perez at gmail.com) Date: Thu, 29 Jan 2015 21:53:55 -0600 Subject: [Freeswitch-users] mod_portaudio compile error Message-ID: Hi, I've successfully built FS v1.4 from the git source in Raspbian, but I'm getting this error when compiling mod_portaudio: making all mod_portaudio make[3]: Entering directory '/home/pi/freeswitch/src/mod/endpoints/mod_portaudio' Makefile:785: *** You must install portaudio19-dev to build mod_portaudio. Stop. make[3]: Leaving directory '/home/pi/freeswitch/src/mod/endpoints/mod_portaudio' Makefile:542: recipe for target 'mod_portaudio-all' failed make[2]: *** [mod_portaudio-all] Error 1 make[2]: Leaving directory '/home/pi/freeswitch/src/mod' Makefile:577: recipe for target 'mod_portaudio' failed make[1]: *** [mod_portaudio] Error 2 make[1]: Leaving directory '/home/pi/freeswitch/src/mod' Makefile:3047: recipe for target 'mod_portaudio' failed make: *** [mod_portaudio] Error 2 I have installed the portaudio19-dev package and also compiled portaudio from source, but none worked. What else should I try to make this work? Thanks, Rogelio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150129/7edb42a1/attachment.html From achinthau at gmail.com Fri Jan 30 08:37:44 2015 From: achinthau at gmail.com (Achintha) Date: Fri, 30 Jan 2015 11:07:44 +0530 Subject: [Freeswitch-users] g729 for freeswitch Message-ID: hi I used freeswitch server with Ubuntu 12.04. It is run as virtual machine (VM). I need to get g729 license for this. Please advise me how to do this. -- Best Regards.. Achintha Udukumbura -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/9560ad8f/attachment.html From asilva at wirelessmundi.com Fri Jan 30 11:50:47 2015 From: asilva at wirelessmundi.com (Antonio Silva) Date: Fri, 30 Jan 2015 09:50:47 +0100 Subject: [Freeswitch-users] g729 for freeswitch In-Reply-To: References: Message-ID: <54CB45E7.4030200@wirelessmundi.com> Hi, You can find the licences here: https://freeswitch.com/cart.php?gid=2 Also more info on use g729 in fs: https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 On 01/30/2015 06:37 AM, Achintha wrote: > hi > > I used freeswitch server with Ubuntu 12.04. It is run as virtual > machine (VM). I need to get g729 license for this. Please advise me > how to do this. > > > -- > Best Regards.. > Achintha Udukumbura > > > _________________________________________________________________________ > 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 -- --- Ant?nio Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/6014f173/attachment.html From idokan at gmail.com Fri Jan 30 13:02:31 2015 From: idokan at gmail.com (ik) Date: Fri, 30 Jan 2015 12:02:31 +0200 Subject: [Freeswitch-users] streaming video in a call Message-ID: Hello, I have a normal call between two legs. Is there a way to stream a video (mp4 file for example) to a specific leg ? I found the mod_vlc but it does not seems to support video streaming, only audio, and I found mod_mp4v but no documentation about how to stream a file. Thanks, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/d980967f/attachment.html From is.yaltunay at gmail.com Fri Jan 30 11:25:58 2015 From: is.yaltunay at gmail.com (=?UTF-8?Q?Y=C3=BCcel_ALTUNAY?=) Date: Fri, 30 Jan 2015 10:25:58 +0200 Subject: [Freeswitch-users] Using Freeswitch with TLS and without TLS together Message-ID: Hi, I want to use freeswitch with TLS on my mobile phones and without TLS on my GSM gateways together. I see some examples but i coudn't do it. Is someone has any example to do this? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/0a142d7a/attachment.html From krice at freeswitch.org Fri Jan 30 18:02:51 2015 From: krice at freeswitch.org (Ken Rice) Date: Fri, 30 Jan 2015 15:02:51 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <54cb9d1b3d373_496810353309629e@ip-10-179-128-163.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/c8df53b7/attachment.html From brian at freeswitch.org Fri Jan 30 19:10:04 2015 From: brian at freeswitch.org (Brian West) Date: Fri, 30 Jan 2015 10:10:04 -0600 Subject: [Freeswitch-users] streaming video in a call In-Reply-To: References: Message-ID: Not possible currently. On Fri, Jan 30, 2015 at 4:02 AM, ik wrote: > Hello, > > I have a normal call between two legs. > Is there a way to stream a video (mp4 file for example) to a specific leg ? > > I found the mod_vlc but it does not seems to support video streaming, only > audio, and I found mod_mp4v but no documentation about how to stream a file. > > Thanks, > Ido > > _________________________________________________________________________ > 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/20150130/6c70e786/attachment.html From lists at kavun.ch Fri Jan 30 20:10:36 2015 From: lists at kavun.ch (Emrah) Date: Fri, 30 Jan 2015 18:10:36 +0100 Subject: [Freeswitch-users] Random calls failing with WRONG_CALL_STATe when using TLS Message-ID: <45FAC76E-D2B7-483A-88AB-9FB98600C42B@kavun.ch> Hi all, I am facing a very frustrating issue. I often have to dial twice when using my Yealink phone with TLS because the first attempt times out. The logs on the Yealink indicate that the first invite is successfully received, to which my FS sends a 100 trying and 407 proxy auth required. It is subsequently when my phone sends back the invite that the connection crashes with the following error: SSL ERROR SYSCALL Is this something common? Why does the SSL connection crashes when the phone attempts to send the second invite? My phone is behind NAT. It is going to be a crazy expedition to collect the logs and Pastebin them, so I am tempting my luck on the list first to see if you have any pointers. As a last piece, my Bria on my iPHone, among other clients, never had this issue. I did experience it from time to time with Blink on Mac OS X. Any help appreciated. Emrah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/296e9f89/attachment-0001.html From denis at ringme.ru Fri Jan 30 20:19:35 2015 From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=) Date: Fri, 30 Jan 2015 20:19:35 +0300 Subject: [Freeswitch-users] freeswitch+ovz - poor sound, terrible timings Message-ID: <54CBBD27.6020705@ringme.ru> timer_test 20 200 ... 2015-01-29 15:26:32.717637 [CONSOLE] mod_commands.c:846 Timer Test: 96 sleep 20 19975 2015-01-29 15:26:32.738371 [CONSOLE] mod_commands.c:846 Timer Test: 97 sleep 20 35287 2015-01-29 15:26:32.738371 [CONSOLE] mod_commands.c:846 Timer Test: 98 sleep 20 4676 2015-01-29 15:26:32.777674 [CONSOLE] mod_commands.c:846 Timer Test: 99 sleep 20 20051 timer_test 10 200 ... 2015-01-30 20:17:50.717723 [CONSOLE] mod_commands.c:846 Timer Test: 184 sleep 10 10084 2015-01-30 20:17:50.754812 [CONSOLE] mod_commands.c:846 Timer Test: 185 sleep 10 37157 2015-01-30 20:17:50.760749 [CONSOLE] mod_commands.c:846 Timer Test: 186 sleep 10 6206 2015-01-30 20:17:50.760749 [CONSOLE] mod_commands.c:846 Timer Test: 187 sleep 10 9680 2015-01-30 20:17:50.777621 [CONSOLE] mod_commands.c:846 Timer Test: 188 sleep 10 6890 2015-01-30 20:17:50.777621 [CONSOLE] mod_commands.c:846 Timer Test: 189 sleep 10 10005 On host node - all ok. Timers - with posix, soft, timerfd... What can i do? From vipkilla at gmail.com Fri Jan 30 21:08:46 2015 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 30 Jan 2015 13:08:46 -0500 Subject: [Freeswitch-users] weird beahivior when using outbound_proxy/fs_path In-Reply-To: References: Message-ID: Are you running latest version of FS? On Thu, Jan 29, 2015 at 10:47 PM, Nuno Reis wrote: > Hello everyone. > > I've been using outbound_proxy for a sometime now to make SIP go trough a > kamailio proxy server. When I put freeswitch under some stress and measure > response times, I start seeing a weird behavior. FS starts like "hanging" > after a fast sequence of INVITES usually after the first 5 INVITE sipke(not > that much right) and hangs for +/- 5 seconds before routing the call out to > kamailio. If i tell FS to route the call using fs_path there is no delay at > all. > > Another weird thing about this one is that I have some installs where > outbound_proxy works fine with max performance everytime and I'm always > using the same install binaries, the same OS the same versions of > everything (checked and double checked). > You could tell me to use fs_path if it solves the problem and I would say > you are right but the thing is fs_path doesn't behave exactly the same as > outbound_proxy does in every situation. > > It's usually common to see the variable destination_number set with the IP > address of the proxy and some other annoying issues and above all > outbound_proxy works at the SIP profile level while fs_path doesn't which > force me to explicitly use it literally everywhere. > > Any ideas on what could be causing this behavior. It looks like a timer of > some kind but I don't see nothing about such a timer anywhere. > Looking forward to hear from you. > > Cheers, > > -- > > *Nuno Miguel Reis* | *Unified Communication** Systems* > M. +351 913907481 | nreis at wavecom.pt > WAVECOM-Solu??es R?dio, S.A. > Cacia Park | Rua do Progresso, Lote 15 > 3800-639 AVEIRO | Portugal > T. +351 309 700 225 | F. +351 234 919 191 > *GPS > > | www.wavecom.pt ** * > > [image: Description: Description: WavecomSignature] > > > [image: Publicity] > > > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/d07eeeb6/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 16423 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/d07eeeb6/attachment-0001.png From nreis at wavecom.pt Fri Jan 30 21:31:41 2015 From: nreis at wavecom.pt (Nuno Reis) Date: Fri, 30 Jan 2015 18:31:41 +0000 Subject: [Freeswitch-users] weird beahivior when using outbound_proxy/fs_path In-Reply-To: References: Message-ID: Hi! Yes. I'm using latest v1.4. -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis at wavecom.pt WAVECOM-Solu??es R?dio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS | www.wavecom.pt ** * [image: Description: Description: WavecomSignature] [image: Publicity] On Fri, Jan 30, 2015 at 6:08 PM, Vik Killa wrote: > Are you running latest version of FS? > > > On Thu, Jan 29, 2015 at 10:47 PM, Nuno Reis wrote: > >> Hello everyone. >> >> I've been using outbound_proxy for a sometime now to make SIP go trough a >> kamailio proxy server. When I put freeswitch under some stress and measure >> response times, I start seeing a weird behavior. FS starts like "hanging" >> after a fast sequence of INVITES usually after the first 5 INVITE sipke(not >> that much right) and hangs for +/- 5 seconds before routing the call out to >> kamailio. If i tell FS to route the call using fs_path there is no delay at >> all. >> >> Another weird thing about this one is that I have some installs where >> outbound_proxy works fine with max performance everytime and I'm always >> using the same install binaries, the same OS the same versions of >> everything (checked and double checked). >> You could tell me to use fs_path if it solves the problem and I would say >> you are right but the thing is fs_path doesn't behave exactly the same as >> outbound_proxy does in every situation. >> >> It's usually common to see the variable destination_number set with the >> IP address of the proxy and some other annoying issues and above all >> outbound_proxy works at the SIP profile level while fs_path doesn't which >> force me to explicitly use it literally everywhere. >> >> Any ideas on what could be causing this behavior. It looks like a timer >> of some kind but I don't see nothing about such a timer anywhere. >> Looking forward to hear from you. >> >> Cheers, >> >> -- >> >> *Nuno Miguel Reis* | *Unified Communication** Systems* >> M. +351 913907481 | nreis at wavecom.pt >> WAVECOM-Solu??es R?dio, S.A. >> Cacia Park | Rua do Progresso, Lote 15 >> 3800-639 AVEIRO | Portugal >> T. +351 309 700 225 | F. +351 234 919 191 >> *GPS >> >> | www.wavecom.pt ** * >> >> [image: Description: Description: WavecomSignature] >> >> >> [image: Publicity] >> >> >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/158641bc/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 16423 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/158641bc/attachment-0001.png From olegstolyar at gmail.com Fri Jan 30 22:30:28 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Fri, 30 Jan 2015 11:30:28 -0800 Subject: [Freeswitch-users] glibc GHOST vulnerability Message-ID: Sorry if the question is naive - trying to be paranoid here. On my CentOS machines I updated my glibc version to one that fixed the GHOST vulnerability. Do I need to rebuild FS or is the library linked dynamically, so there is no need to rebuild? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/ded79101/attachment.html From sos at sokhapkin.dyndns.org Fri Jan 30 22:37:10 2015 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 30 Jan 2015 14:37:10 -0500 Subject: [Freeswitch-users] glibc GHOST vulnerability In-Reply-To: References: Message-ID: <142268237.LKhDd2F0Fm@sos> There is no need to rebuild an application linked against a dynamic library. On Friday 30 January 2015 11:30:28 Oleg Stolyar wrote: > Sorry if the question is naive - trying to be paranoid here. > > On my CentOS machines I updated my glibc version to one that fixed the > GHOST vulnerability. > > Do I need to rebuild FS or is the library linked dynamically, so there is > no need to rebuild? From olegstolyar at gmail.com Fri Jan 30 23:01:03 2015 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Fri, 30 Jan 2015 12:01:03 -0800 Subject: [Freeswitch-users] glibc GHOST vulnerability In-Reply-To: <142268237.LKhDd2F0Fm@sos> References: <142268237.LKhDd2F0Fm@sos> Message-ID: Yep, just being paranoid and want to absolutely confirm that the standard FreeSWITCH build links libraries (including glibc) dynamically. On Jan 30, 2015 11:39 AM, "Sergey Okhapkin" wrote: > There is no need to rebuild an application linked against a dynamic > library. > > On Friday 30 January 2015 11:30:28 Oleg Stolyar wrote: > > Sorry if the question is naive - trying to be paranoid here. > > > > On my CentOS machines I updated my glibc version to one that fixed the > > GHOST vulnerability. > > > > Do I need to rebuild FS or is the library linked dynamically, so there is > > no need to rebuild? > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/b3fe482d/attachment.html From you.kissmyarse at gmail.com Fri Jan 30 23:29:55 2015 From: you.kissmyarse at gmail.com (Benjamin Rowe) Date: Fri, 30 Jan 2015 12:29:55 -0800 Subject: [Freeswitch-users] Issues with originate, webrtc and xml_curl In-Reply-To: References: Message-ID: I've been trying to dig into this more but can't find the issue i think it must have something to do with the dial-string. Here is curl response from my PHP script for 1004's authentication.
On Thu, Jan 29, 2015 at 6:20 PM, Benjamin Rowe wrote: > Evening All, > > I have a basic system up and running i am able to dial out via my provider > on both sip endpoints. I have setup 2 users 1000 and 1004. 1000 is created > from the flat file xml and 1004 is from the xml_curl request. when i > attempt to originate a call from user 1000 to an external number it passes > through fine however when i originate a call from user 1004 i get > subscriber absent both are running the same setup to my knowledge. Any > Ideas? > > Errors: > 2015-01-29 18:12:17.745051 [WARNING] mod_dptools.c:3979 Can't find user > [1004@] > 2015-01-29 18:12:17.745051 [NOTICE] switch_ivr_originate.c:2735 Cannot > create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] > 2015-01-29 18:12:17.745051 [DEBUG] switch_ivr_originate.c:3723 Originate > Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] > 2015-01-29 18:12:17.745051 [INFO] mod_dptools.c:3234 Originate Failed. > Cause: SUBSCRIBER_ABSENT > 2015-01-29 18:12:17.745051 [NOTICE] switch_channel.c:4724 Hangup > loopback/1004-b [CS_EXECUTE] [SUBSCRIBER_ABSENT] > > Sofia Registrations: > > ================================================================================================= > Call-ID: ocq57n8fn23uo0s7rh2025 > User: 1004@ > Contact: "" ;transport=ws;fs_nat=yes;fs_path=sip%3Amp2337gf%40%3A59613%3Btransport%3Dws> > Agent: SIP.js/0.6.3-devel BB > Status: Registered(WS-NAT)(unknown) EXP(2015-01-29 18:24:02) > EXPSECS(571) > Ping-Status: Reachable > Host: fs01-a > IP: > Port: 59613 > Auth-User: 1004 > Auth-Realm: > MWI-Account: 1004@ > > > Originate command: > originate > {ignore_early_media=true,origination_caller_id_number=1004}loopback/1004 > &bridge(sofia/gateway/test/555123456) > > Many Thanks > Benjamin Rowe > -- Benjamin Rowe Lazypeople Mobile: +44 (0)7904 026869 Phone: +44 (0)845 86 99 892 Email: ben at lazypeople.co.uk Web: http://www.lazypeople.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/bafb1f32/attachment.html From lists at kavun.ch Fri Jan 30 23:49:14 2015 From: lists at kavun.ch (Emrah) Date: Fri, 30 Jan 2015 21:49:14 +0100 Subject: [Freeswitch-users] Using Freeswitch with TLS and without TLS together In-Reply-To: References: Message-ID: Ucel, The example configurations let you do this out of the box. You have a port for TLS and a port for SIP over UDP and TCP on your internal profile. How are your GSM gateways connected? Are their registering as users or do you use them as gateway endpoints? Your external profile shouldn't be configured with support for TLS by default. Give us more detail about your setup. Best, Emrah > On Jan 30, 2015, at 9:25 AM, Y?cel ALTUNAY wrote: > > Hi, > I want to use freeswitch with TLS on my mobile phones and without TLS on my GSM gateways together. I see some examples but i coudn't do it. > Is someone has any example to do this? > Thank you. > _________________________________________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/66d8a16f/attachment-0001.html From ahabiba at gmail.com Sat Jan 31 00:06:52 2015 From: ahabiba at gmail.com (Ahmed Habiba) Date: Sat, 31 Jan 2015 00:06:52 +0300 Subject: [Freeswitch-users] mod_snmp not loading In-Reply-To: References: Message-ID: <5FF889B0-EF0E-4396-BE2F-359BE90A6EDC@gmail.com> Dears, Any help in this regard. you kind help will be appreciated. > On Jan 26, 2015, at 6:30 PM, Ahmed Habiba wrote: > > Thank you really Vik > > here is below my linux version: > > > Distributor ID: Ubuntu > Description: Ubuntu 12.04.5 LTS > Release: 12.04 > Codename: precise > > > From: Vik Killa > > To: FreeSWITCH Users Help > > Date: January 26, 2015 at 4:14:35 PM GMT+3 > Reply-To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] mod_snmp not loading > > > What distro of linux are you using? > Last I checked, mod_snmp had issues on centos because of a dependencie > > On Sat, Jan 24, 2015 at 9:05 AM, Ahmed Habiba > wrote: > Dears, > > kindly I tried to use mod_snmp I compile it using make && make install, however when I tried to load it I got the below message:, your kind usual support will be appreciated: > > 2015-01-24 15:05:58.673046 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_snmp.so > **/usr/lib/libnetsnmpagent.so.15: undefined symbol: netsnmp_register_null_context** > > Thanks, > > Ahmed Habiba. > > _________________________________________________________________________ > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150131/538e4a17/attachment.html From cinthia721 at gmail.com Sat Jan 31 04:28:50 2015 From: cinthia721 at gmail.com (Cinthia Leung) Date: Fri, 30 Jan 2015 20:28:50 -0500 Subject: [Freeswitch-users] Need help with Video proxy-media mode Message-ID: Hi there, I'm trying to setup FS and proxy media. Here's my setup. Kamailio SBC <-> FS <-> MCU. >show codec sees H.264 as passthru late-negotiation = true Audio is ok but I get no video. Tshark capture shows that video streams enter FS from both Kamailio and MCU. FS was just not forwarding video packets to the other side. Codec is H.264. Tested with FS Version 1.14.15 and 1.5.15b+git~20150130T165344Z~4ed7b4811a~64bit Bypass media works fine. Being able to proxy media is a requirement for us. I hope it's just something obvious that I missed. I'll provide logs if needed. Thanks in advance! Cindy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/abdfd4aa/attachment.html From cinthia721 at gmail.com Sat Jan 31 04:28:50 2015 From: cinthia721 at gmail.com (Cinthia Leung) Date: Fri, 30 Jan 2015 20:28:50 -0500 Subject: [Freeswitch-users] Need help with Video proxy-media mode Message-ID: Hi there, I'm trying to setup FS and proxy media. Here's my setup. Kamailio SBC <-> FS <-> MCU. >show codec sees H.264 as passthru late-negotiation = true Audio is ok but I get no video. Tshark capture shows that video streams enter FS from both Kamailio and MCU. FS was just not forwarding video packets to the other side. Codec is H.264. Tested with FS Version 1.14.15 and 1.5.15b+git~20150130T165344Z~4ed7b4811a~64bit Bypass media works fine. Being able to proxy media is a requirement for us. I hope it's just something obvious that I missed. I'll provide logs if needed. Thanks in advance! Cindy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150130/abdfd4aa/attachment-0001.html From jtbock at synacktics.com Sat Jan 31 08:07:37 2015 From: jtbock at synacktics.com (Tim Bock) Date: Fri, 30 Jan 2015 22:07:37 -0700 Subject: [Freeswitch-users] Call intercept problem Message-ID: <54CC6319.2000005@synacktics.com> Hello list, Running freeswitch 1.4.6, though have also seen this issue with older versions. After the initial freeswitch install, I tested the call intercept function, and it worked fine. Fast forward several months ahead, and now call intercept doesn't work. Looking at the logs, the message is that the channel is no longer available. Indeed, when I examine the uuid the call intercept function is trying to connect, the uuid is for the call immediately preceding the current call, sort of like an "off by one" indexing error. This is reliably repeatable. Any thoughts on why this is occurring? And more importantly, any way to fix it? Thank you, Tim -- Tim Bock Synacktics, LLC www.synacktics.com (505)795-1511 From regis.freeswitch.org at tornad.net Sat Jan 31 12:27:18 2015 From: regis.freeswitch.org at tornad.net (Regis M) Date: Sat, 31 Jan 2015 10:27:18 +0100 Subject: [Freeswitch-users] glibc GHOST vulnerability In-Reply-To: References: <142268237.LKhDd2F0Fm@sos> Message-ID: It's not paranoid, it's a logic and normal question. For me, but as to be confirmed by freeswitch dev and c expert, FS compile with linked library by default. And doing : $ ldd /bin/freeswitch show you the linked librairy with your binary on your system. If someone else can confirm my post too, I'm not 100% sure. Thanks 2015-01-30 21:01 GMT+01:00 Oleg Stolyar : > Yep, just being paranoid and want to absolutely confirm that the standard > FreeSWITCH build links libraries (including glibc) dynamically. > > > On Jan 30, 2015 11:39 AM, "Sergey Okhapkin" > wrote: > >> There is no need to rebuild an application linked against a dynamic >> library. >> >> On Friday 30 January 2015 11:30:28 Oleg Stolyar wrote: >> > Sorry if the question is naive - trying to be paranoid here. >> > >> > On my CentOS machines I updated my glibc version to one that fixed the >> > GHOST vulnerability. >> > >> > Do I need to rebuild FS or is the library linked dynamically, so there >> is >> > no need to rebuild? >> >> _________________________________________________________________________ >> 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150131/c89f1808/attachment-0001.html From kamil.nigmatullin at gmail.com Sat Jan 31 14:39:00 2015 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Sat, 31 Jan 2015 17:39:00 +0600 Subject: [Freeswitch-users] Unable to access channel vars after session terminates evens with set_zombie_exec() In-Reply-To: References: Message-ID: When exactly do you need this variable? Do you want it to appear in CDR table? Can you please explain what exactly you do and whet varable should appear in cdr? 2015-01-23 2:53 GMT+06:00 Muhammad Naseer Bhatti : > > I need to access some channel variables being set in the dialplan, but the > channel is already hanged up too quick, ORIGINATOR_CANCEL. Since the > session was still setting up the channel variables. I have tried app, > set_zombie_exec but still not able to see any channel vars set in the dial > plan. Am i doing something wrong or if there is a better way to see the > channel vars if the session terminates too soon? > > With: git a067a49 > > Dialplan: sofia/internal/9401404 at 10.211.55.26 Regex (PASS) [internal] > destination_number(1786866) =~ /$/ break=on-false > *Dialplan: sofia/internal/9401404 at 10.211.55.26 <9401404 at 10.211.55.26> > Action set_zombie_exec() * > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action > set(continue_on_fail=true) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action > set(fail_on_single_reject=USER_BUSY,NO_ANSWER,NO_USER_RESPONSE,ORIGINATOR_CANCEL) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action > set(disable_hold=true) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action > set(failed_json_cdr_prefix=failed_cdr_index) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(debug_cdr=0) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action set(debug_cdr_sql=1) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action > set(cust_default_lrn=intra) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action > set(cust_lrn_dip_cost=0.01817) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action > set(cust_jurisdiction=INTRASTATE) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit(hash > random_xgw 0.0.0.0 10/1 !FACILITY_REJECTED) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit_execute(hash > outbound random_xgw 20 bridge > [enable_heartbeat_events=5,nibble_rate=0.2192,nibble_increment=5,nibble_account=AB8KA191,carrier_switch=random_xgw,carrier_switch_id=1,carrier_ratecard_id=2,carrier_rate_rev=1,carrier_rate_type=lrn,carrier_id=1,carrier_connection_cost=0,carrier_rate=0.0119,carrier_interstate_cost=0.0119,carrier_intrastate_cost=0.0119,carrier_enable_billing=t,carrier_call_increment=1,carrier_min_duration=5,carrier_balance=17099.88924]sofia/gateway/random_xgw/1786866) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit(hash switch02 > 0.0.0.0 10/1 !FACILITY_REJECTED) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action limit_execute(hash > outbound switch02 20 bridge > [enable_heartbeat_events=5,nibble_rate=0.2192,nibble_increment=5,nibble_account=AB8KA191,carrier_switch=switch02,carrier_switch_id=3,carrier_ratecard_id=2,carrier_rate_rev=1,carrier_rate_type=lrn,carrier_id=1,carrier_connection_cost=0,carrier_rate=0.0119,carrier_interstate_cost=0.0119,carrier_intrastate_cost=0.0119,carrier_enable_billing=t,carrier_call_increment=1,carrier_min_duration=5,carrier_balance=17099.88924]sofia/gateway/switch02/1786866) > Dialplan: sofia/internal/9401404 at 10.211.55.26 Action hangup() > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:216 > (sofia/internal/9401404 at 10.211.55.26) State Change CS_ROUTING -> > CS_EXECUTE > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_session.c:1388 Send signal > sofia/internal/9401404 at 10.211.55.26 [BREAK] > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:528 > (sofia/internal/9401404 at 10.211.55.26) State ROUTING going to sleep > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:472 > (sofia/internal/9401404 at 10.211.55.26) Running State Change CS_EXECUTE > *2015-01-22 11:47:14.856013 [DEBUG] sofia.c:6614 Channel > sofia/internal/9401404 at 10.211.55.26 <9401404 at 10.211.55.26> entering state > [terminated][487]* > *2015-01-22 11:47:14.856013 [NOTICE] sofia.c:7530 Hangup > sofia/internal/9401404 at 10.211.55.26 <9401404 at 10.211.55.26> [CS_EXECUTE] > [ORIGINATOR_CANCEL]* > *2015-01-22 11:47:14.856013 [DEBUG] switch_channel.c:3222 Send signal > sofia/internal/9401404 at 10.211.55.26 <9401404 at 10.211.55.26> [KILL]* > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_session.c:1388 Send signal > sofia/internal/9401404 at 10.211.55.26 [BREAK] > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9401404 at 10.211.55.26) State EXECUTE > 2015-01-22 11:47:14.856013 [DEBUG] mod_sofia.c:178 sofia/internal/ > 9401404 at 10.211.55.26 SOFIA EXECUTE > 2015-01-22 11:47:14.856013 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9401404 at 10.211.55.26) State EXECUTE going to sleep > > > ? > Thanks, > Muhammad Naseer Bhatti > > _________________________________________________________________________ > 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 > -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150131/0e3906d4/attachment.html From john.nash778 at gmail.com Sat Jan 31 17:28:40 2015 From: john.nash778 at gmail.com (John Nash) Date: Sat, 31 Jan 2015 19:58:40 +0530 Subject: [Freeswitch-users] Correct progress time variable in CDR Message-ID: In my setup when Invite comes to freeswitch dialplan plays a sound file and then Bridge the call to destination. I am using csv CDR and trying to get timestamp when there is either 180 or 183 from end destination (After bridge). What variables should I use the correct progress timestamp. I printed variables in two extensions .. Extension 1 (where sound file is being played) ------------------------------------------------------------------ Caller-Profile-Created-Time: [1422696316256090] Caller-Channel-Created-Time: [1422696316256090] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Extension 2 (where call is being bridged transferred from Extension 1) ---------------------------------------------------------------------------------------------------- Caller-Profile-Created-Time: [1422696320776083] Caller-Channel-Created-Time: [1422696316256090] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [1422696316256090] >From this log I think progress timestamp is the one generated when sound file plays but I need the progress time when 180/183 received after bridge. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150131/1359a1cc/attachment.html From mayank19882001 at gmail.com Sat Jan 31 13:11:19 2015 From: mayank19882001 at gmail.com (Mayank Nakrani) Date: Sat, 31 Jan 2015 15:41:19 +0530 Subject: [Freeswitch-users] Help need for Lua Script development Message-ID: Hello Guys, Im starting with my Lua scripts dev. Have already had luck writing an Callback script. Now i would to go deeper integrating it with Mysql and other libraries and generating some dynamic dialplan. Till now i used to write my scripts in Text editor and would to know is it Possible in IDE like Eclipse. I installed Lua development plugin in Eclipse but was wondering Whats the correct way to import freeswitch libraries in the project so that i can see methods/functions in in freeswitch API for easier and faster coding. I tried loading "libfreeswitch" and "mod_lua" lib using "require" command but its failing with [com.naef.jnlua.LuaRuntimeException: error loading module 'libfreeswitch' from file './libfreeswitch.so': ./libfreeswitch.so: undefined symbol: apr_os_default_encoding ] Im gud in Java and can use mod_java but would like to stick with lua as i heard its more closer to freeswitch. Any help :) Regards Mayank Nakrani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150131/fb7fbe82/attachment.html