May 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Verto calls
by Agusti Ubalde 16 May '17

16 May '17

Hi all, I am trying to call from Verto extension to another Verto extension. Both are successfully registered (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): EXECUTE verto.rtc/1001 bridge() Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)presenceco.com<mailto:aubalde@presenceco.com> Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH

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REFER message with replaces from Alcate OmniPCX Server
by hancymoon 16 May '17

16 May '17

Hello everyone: I have a internal sip phone(A) registered to Freeswitch. If A call a number via siptrunk to Alcate's phone(B). B answered, then transfer this call to Alcate's phone(C), if C is still ringing, but B hangup, then C stop ringing and hangup, and there is no connection bwtween A and C. Please help me to check why A cannot connect to C. Here is the log: recv 688 bytes from udp/[160.7.237.4]:5060 at 12:18:20.272035: ------------------------------------------------------------------------ REFER sip:gw+siptrunk1@160.7.237.89:5080;transport=udp;gw=siptrunk1 SIP/2.0 Contact: sip:160.7.237.4 Supported: timer,path,100rel User-Agent: OmniPCX Enterprise R11.2.2 l2.300.33.a Refer-To: <sip:7950@160.7.237.4;user=phone?REPLACES=5334d53d-b3c8-1235-dfbf-109836ab50bb%3bto-tag%3dvaZK46cXrgeyQ%3bfrom-tag%3d85e354db02f8d15fe12b28a385cb8861> Referred-By: sip:1260@160.7.237.4 To: sip:66577160@160.7.237.89;tag=vaZK46cXrgeyQ From: sip:1260@160.7.237.4:5060;tag=85e354db02f8d15fe12b28a385cb8861 Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 1904933290 REFER Via: SIP/2.0/UDP 160.7.237.4;branch=z9hG4bK7d39c893d0c0fd0deab78b450e6efd31 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] sofia.c:8487 Process REFER to [7950(a)160.7.237.4] 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] sofia.c:8513 Replaces: [5334d53d-b3c8-1235-dfbf-109836ab50bb] send 613 bytes to udp/[160.7.237.4]:5060 at 12:18:20.290107: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 160.7.237.4;branch=z9hG4bK7d39c893d0c0fd0deab78b450e6efd31 From: sip:1260@160.7.237.4:5060;tag=85e354db02f8d15fe12b28a385cb8861 To: sip:66577160@160.7.237.89;tag=vaZK46cXrgeyQ Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 1904933290 REFER Contact: <sip:gw+siptrunk1@160.7.237.89:5080;transport=udp;gw=siptrunk1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.6.15~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [NOTICE] sofia.c:8710 Attended Transfer [791cb18a-3925-11e7-9cf0-29f19fd18425][791cb18a-3925-11e7-9cf0-29f19fd18425] 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] switch_ivr_bridge.c:1982 (FreeTDM/1:21/1260) State Change CS_EXECUTE -> CS_HIBERNATE 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] switch_ivr_bridge.c:1984 (FreeTDM/1:21/1260) State Change CS_HIBERNATE -> CS_CONSUME_MEDIA 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] switch_channel.c:2029 (sofia/external/1260) Callstate Change HELD -> UNHELD 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:20.270464 [DEBUG] sofia.c:8785 (sofia/external/1260) State Change CS_EXCHANGE_MEDIA -> CS_PARK send 774 bytes to udp/[160.7.237.4]:5060 at 12:18:20.290288: ------------------------------------------------------------------------ NOTIFY sip:160.7.237.4 SIP/2.0 Via: SIP/2.0/UDP 160.7.237.89:5080;rport;branch=z9hG4bKH8BaH8ejBemBB Max-Forwards: 70 From: "66577160" <sip:66577160@160.7.237.89>;tag=vaZK46cXrgeyQ To: <sip:1260@160.7.237.4:5060>;tag=85e354db02f8d15fe12b28a385cb8861 Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 107074877 NOTIFY Contact: <sip:gw+siptrunk1@160.7.237.89:5080;transport=udp;gw=siptrunk1> User-Agent: FreeSWITCH-mod_sofia/1.6.15~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Event: refer;id=1904933290 Allow-Events: talk, hold, conference, refer Subscription-State: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 16 SIP/2.0 200 OK ------------------------------------------------------------------------ 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.290465 [DEBUG] switch_ivr_play_say.c:1942 done playing file local_stream://moh recv 545 bytes from udp/[160.7.237.4]:5060 at 12:18:20.304851: ------------------------------------------------------------------------ SIP/2.0 200 OK Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO Contact: sip:160.7.237.4 Supported: replaces,timer,path,100rel User-Agent: OmniPCX Enterprise R11.2.2 l2.300.33.a To: <sip:1260@160.7.237.4:5060>;tag=85e354db02f8d15fe12b28a385cb8861 From: "66577160" <sip:66577160@160.7.237.89>;tag=vaZK46cXrgeyQ Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 107074877 NOTIFY Via: SIP/2.0/UDP 160.7.237.89:5080;received=160.7.237.89;rport=5080;branch=z9hG4bKH8BaH8ejBemBB Content-Length: 0 ------------------------------------------------------------------------ recv 470 bytes from udp/[160.7.237.4]:5060 at 12:18:20.310591: ------------------------------------------------------------------------ BYE sip:gw+siptrunk1@160.7.237.89:5080;transport=udp;gw=siptrunk1 SIP/2.0 Supported: replaces,timer,path,100rel User-Agent: OmniPCX Enterprise R11.2.2 l2.300.33.a To: sip:66577160@160.7.237.89;tag=vaZK46cXrgeyQ From: sip:1260@160.7.237.4:5060;tag=85e354db02f8d15fe12b28a385cb8861 Call-ID: 5334d53d-b3c8-1235-dfbf-109836ab50bb CSeq: 1904933291 BYE Via: SIP/2.0/UDP 160.7.237.4;branch=z9hG4bKf69a857f743ba04db1f37bb7af0337f8 Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:20.310465 [DEBUG] switch_ivr_bridge.c:752 BRIDGE THREAD DONE [FreeTDM/1:21/1260] 791cb18a-3925-11e7-9cf0-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] switch_ivr_bridge.c:1729 (FreeTDM/1:21/1260) State Change CS_CONSUME_MEDIA -> CS_RESET 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [NOTICE] fssession.cpp:139 Hangup sofia/external/1260 [CS_PARK] [NORMAL_CLEARING] 2017-05-15 12:18:25.530465 [INFO] mod_v8.cpp:627 Javascript result: [true] 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] switch_ivr_bridge.c:752 BRIDGE THREAD DONE [sofia/external/1260] 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] switch_core_state_machine.c:653 (sofia/external/1260) State EXCHANGE_MEDIA going to sleep 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] switch_core_state_machine.c:584 (sofia/external/1260) Running State Change CS_HANGUP (Cur 5 Tot 17168) 7bd68496-3925-11e7-9cfa-29f19fd18425 2017-05-15 12:18:25.530465 [DEBUG] sofia.c:1453 Channel is already hungup. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/REFER-message-with-replaces-f… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Verto calls
by Agusti Ubalde 16 May '17

16 May '17

Hi all, I am trying to call from Verto extension to another Verto extension. Both are successfully registered (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): EXECUTE verto.rtc/1001 bridge() Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)presenceco.com<mailto:aubalde@presenceco.com> Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH

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Re: [Freeswitch-users] Question about enabling SCTP on freeswitch on ubuntu 16.04 from apt repository.
by Ken Rice 16 May '17

16 May '17

Packages are not build with sctp support enabled. Sent from my iPhone > On May 15, 2017, at 17:56, Ravi sanyal <virtualguard2015(a)gmail.com> wrote: > > Hi, > > I know this will be specific to my installation, so I'm running ubuntu 16.04 and downloaded freeswitch from the apt repository. Oh yeah, i also just did a clean install of freeswitch. I've only been using it for a few weeks. > > If i run the sofia dig command on the domain it only shows udp and tcp. > doing 'sofia status profile internal/external' does not have the line for sctp either. > > First thing, will i have to uninstall the repository and build it? i know that guides say to run the build with --enable-sctp i tried looking for sctp module (i know its a bad name, it's not a module) using apt however not familiar with what the name would be called. freeswitch-* is a mammoth list > > doing apt-cache search freeswitch | grep sctp doesn't find anything. > > Once i've set up freeswitch to handle it, i'll still need to do naptr and srv records on the domain. im running with rackspace. Has anyone used rackspace before? they say that naptr and srv can be added using their system (however there is no naptr record adding) > > Nathan > > -- > Virtual Guard Ltd > info(a)virtualguard.co.nz

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Question about enabling SCTP on freeswitch on ubuntu 16.04 from apt repository.
by Ravi sanyal 15 May '17

15 May '17

Hi, I know this will be specific to my installation, so I'm running ubuntu 16.04 and downloaded freeswitch from the apt repository. Oh yeah, i also just did a clean install of freeswitch. I've only been using it for a few weeks. If i run the sofia dig command on the domain it only shows udp and tcp. doing 'sofia status profile internal/external' does not have the line for sctp either. First thing, will i have to uninstall the repository and build it? i know that guides say to run the build with --enable-sctp i tried looking for sctp module (i know its a bad name, it's not a module) using apt however not familiar with what the name would be called. freeswitch-* is a mammoth list doing apt-cache search freeswitch | grep sctp doesn't find anything. Once i've set up freeswitch to handle it, i'll still need to do naptr and srv records on the domain. im running with rackspace. Has anyone used rackspace before? they say that naptr and srv can be added using their system (however there is no naptr record adding) Nathan -- *Virtual Guard Ltd* *info(a)virtualguard.co.nz <info(a)virtualguard.co.nz>*

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Freeswitch Time of Day Routing XML_curl
by Piccinin Francesco 15 May '17

15 May '17

Hi all, we are trying to setup Freeswitch ToD routing using XML curl for XML dialplan dynamic generation. Please note that our dialplan is mainly stored on static file (plus lua support) with few exceptions like IVRs. In order to avoid freeswitch sending http post on every call (as would happen adding binding=directory on xml_curl configuration module), we would like to find a way to generate dynamic xml dialplan only on particular case like ToD. Can you please give us some clues about this? Thanks Regards Francesco

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MOD_SOFIA
by Caio Assis 15 May '17

15 May '17

Good Afternoon. Is there a command I type that loads a new SIP account? When I type reload mod_sofia and it's already in use, it gives me an error message "+OK Reloading XML -ERR unloading module [Module in use.] -ERR loading module [Module already loaded]" I'm looking for an asterisk 'sip reload' freeswitch equivalent. If I have calls in progress and create a SIP account to use, I either have to wait for all calls to finish or restart Freeswitch, which makes it almost inviable. Can anyone help me with this? Thanks.

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SIPS scheme
by David Villasmil 15 May '17

15 May '17

Hello guys. I've got this setup, where the client is sending as contact "sip:" coming from kamailio proxy in tls ip until kamailio. Kamailio forwards to FS in tcp, but freeswitch is answering with "sip:" and this breaks the client who answers "SIPS Required"... Is there a way of specifying the contact should be sips ? Has anyone encountered this and solved it? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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300 Multiple Choices - how to force FS to replace From field
by Lyubo Popov 15 May '17

15 May '17

Hello everyone, I would like to express my thanks in advance to anyone who may be able to help me with some insides. I am using a routing software with SIP Redirect to send routes to FS with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In the Sip redirect server I am manipulating as well the FROM number and sending back to FS, but FS will not respect this and continue using the SIP account that sent the call in the first place in the FROM field. Here are some SIP packets from both sides to clarify the whole process. 1. Sending call to FS with Zoiper, destination 556230951662 INVITE sip:556230951662@216.x.x.x:5080;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK-d8754z-038f1c7d251308c2-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:551000@177.41.146.98:1048;transport=UDP> To: <sip:556230951662@216.x.x.x:5080;transport=UDP> From: "551000"<sip:551000@216.x.x.x:5080;transport=UDP>;tag=25599d20 Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper for Windows 2.43 r24984 Allow-Events: presence, kpml Content-Length: 232 v=0 o=Zoiper_user 0 0 IN IP4 177.x.x.x s=Zoiper_session c=IN IP4 177.x.x.x t=0 0 m=audio 8000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 2. FS sending INVITE to SIP Redirect server INVITE sip:556230951662@69.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa Max-Forwards: 69 From: "551000" <sip:551000@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: 69.x.x.x:5060 == ^69\.x\.x\.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 22476 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 22476 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 3. SIP Redirect returns 300 Multiple choices with the termination IP in Contact and with FROM field as instructed ( update 551000 with 1140031556) SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa From: "1140031556" <sip:1140031556@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Contact: <sip:556230951662@162.x.x.x:5060>;q=1.00 Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Max-Forwards: 69 Content-Length: 0 Server: SIP Redirect Server 4. FS will send the call to the Termination IP WITHOUT changing the FROM field INVITE sip:556230951662@162.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp Max-Forwards: 68 From: "551000" <sip:551000@216.x.x.x>;tag=gQepU2j7X9BKr To: <sip:556230951662@162.x.x.x:5060> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true match: 162.x.x.x:5060 == ^69\.x\.x\.x User-Agent: FreeSWITCH 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: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 20696 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 20696 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 This is the dialplan I use to send calls to SIP redirect server <include> <extension name="rejections"> <condition field="${radius_auth_result}" expression="2"> <action application="hangup" data="CALL_REJECTED"/> </condition> </extension> <extension name="timedouts"> <condition field="${radius_auth_result}" expression="1"> <action application="hangup" data="SWITCH_CONGESTION"/> </condition> </extension> <extension name="SIP Redirect Server"> <condition field="${radius_auth_result}" expression="0"/> <condition field="destination_number" expression="^(.+)$">  <action application="export" data="nolocal:h323-call-origin=originate"/> <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="inherit_codec=true" /> <action application="set" data="call_timeout=20"/> <action application="set" data="fail_on_single_reject=USER_BUSY" /> <action application="set" data="origination_caller_id_name=${sip_req_user}"/> <action application="set" data="origination_caller_id_number=${sip_from_user}"/> <action application="set" data="execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout" /> <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@ ${sip_network_ip}}sofia/internal/${destination_number}@69.x.x.x:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> </extension> </include Is there any variable that will force FS to change the FROM field as returned by the SIP Redirect server and send it to the termination provider? Any help on this is really greatly appreciated! Best regards, L. Popov

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Presence Server
by kk-mailinglist＠ednt.de 13 May '17

13 May '17

i try to use the LED´s on the Phone to sgive user a indication of the state of my Phone is redirected or ithis Phone is subscribed in a Queue. For this i use a free extension in sample 1000 and put on the Phone the LED to BLF 1000. This Works until the phone subscribe again after 3600 sec. The my manipulation will be lost in the presence server. So i need a way to take care about this state. Is there a common solution for this issue ? regards

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