- FreeSWITCH-users - FreeSWITCH Mailing Lists

Fwd: Change the signaling port after the ACK
by Joaquin Peral 16 May '17

16 May '17

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Re: [Freeswitch-users] voicemail_ivr examples when checking voicemail
by Jonathan Hunter 16 May '17

16 May '17

Hi Michael, Thanks for the response, so I can just exclude phrases in; etc/freeswitch/lang/en/vm/sounds.xml to stop it reading out particular options, as essentially I dont want the user ringing in to check their voicemail to not have options to record multiple greetings, just want them to be able to record one, thats only change I need 😊 Jon ---------- Forwarded message ---------- From: Michael Jerris <mike(a)jerris.com<mailto:mike@jerris.com>> Date: Wed, May 10, 2017 at 7:49 PM Subject: Re: [Freeswitch-users] voicemail_ivr examples when checking voicemail To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> On May 10, 2017, at 2:34 PM, Jonathan Hunter <hunterj91(a)hotmail.com<mailto:hunterj91@hotmail.com>> wrote: Hey guys, Sorry for the noise! I am looking to use voicemail_ivr so I can ensure my own phrases are played and I can avoid giving the user options to record just one voicemail message (as they have the option to upload a file from our portal too). You can do all this with regular voicemail app by modifying the phrases involved and options I think. No need to do your own completely. Take a look at the phrase configs used for voicemail and voicemail config file and let us know where it might be an issue. Can this be achieved using voicemail_ivr and if so are there any examples as confluence is good but doesnt have many configuration examples if you want to miss out phrases etc. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Jonathan Hunter Technical Director /Telephony Developer M:(+44) 7917 190 438 Email:jhunter@voxboxcoms.co.uk<mailto:Email%3Ajhunter@voxboxcoms.co.uk>

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Limit module using hiredis
by John Nash 16 May '17

16 May '17

I am using limit module as per https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Limit and using it to limit number of concurrent calls for a given resource (Unique ID for a group of Source IPs). I am using hiredis module and my redis server is on another box. It all seemed to run fine when I implemented but after few days of running (Freeswitch was restarted but not redis server) I saw that redis entries have minus values against keys. In what condition they will cause entries to be negative?.

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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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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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