February 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Inbound calls mis-routing routing to internal extension with external IP?
by Brian Chow 25 Feb '16

25 Feb '16

Hello all! I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT. My Setup: Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance. SIP Provider: Flowroute NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively. I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine). When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip? I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@*<externalip>* :46072 Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension> ? Sofia Status says: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@<external_ip>:5080 RUNNING (0) external::<uuid> gateway sip:<flowrouteid>@sip.flowroute.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@<internal_ip>:5060 RUNNING (0) ================================================================================================= If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be. Thanks, -Brian

3 3

Difference between mod_avmd and mod_com_amd
by Grant Bagdasarian 25 Feb '16

25 Feb '16

Hello, Could someone explain to me the difference between the modules mod_avmd and mod_com_amd, except for the price different? Regards, Grant

2 3

mod_rtmp rtmps support
by Alessandro Illiano 25 Feb '16

25 Feb '16

Hi all, does mod_rtmp support rtmps? Thanks Alessandro

1 0

serving configuration with mod_lua, and sofia.conf
by Artur Mega 25 Feb '16

25 Feb '16

Hello all, I try to serve configuration via lua like it's described at https://freeswitch.org/confluence/display/FREESWITCH/Serving+Configuration+… <configuration name="lua.conf" description="LUA Configuration"> <settings> <param name="xml-handler-script" value="configuration.lua"/> <param name="xml-handler-bindings" value="configuration"/> </settings> </configuration> In configuration.lua script I set XML_STRING to valid xml, taken from autoload_configs. Similar to this method, i did the same with mod_xml_curl, and all works fine. But with mod_lua sofia profiles not loading. I think, the problem is in variables. Let me explain. When I turn off mod_lua and then make `reload mod_sofia` in fs_cli i see something like ... 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 context [internal] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rfc2833-pt [101] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 sip-port [5060] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dialplan [XML] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dtmf-duration [2000] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 inbound-codec-prefs [OPUS,G722,PCMU,PCMA,VP8] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 outbound-codec-prefs [OPUS,G722,PCMU,PCMA,VP8] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-timer-name [soft] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-ip [xxx.xx.x.x] .... But when i turn on mod_lua and relod mod_sofia, in logs i see, that variables not calculated! 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 context [internal] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rfc2833-pt [101] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 sip-port [$${internal_sip_port}] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dialplan [XML] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 dtmf-duration [2000] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 inbound-codec-prefs [$${global_codec_prefs}] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 outbound-codec-prefs [$${global_codec_prefs}] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-timer-name [soft] 2016-02-25 13:04:03.875154 [DEBUG] sofia.c:4237 rtp-ip [$${local_ip_v4}] If i run in fs_cli `eval $${local_ip_v4}`, it prints calculated IP. Maybe I missed something? Thanks -- Arthur

1 0

Web RTC -> No Candidate found
by Alexander Haugg 25 Feb '16

25 Feb '16

Hi, I have the problem, that the freeswitch don't accept the network internal candidate. The local IP address of the fs is 192.168.241.5 The local IP address of the webrtc Client is 192.168.241.2 In the acl.conf.xml is set: <list name="localnet.auto" default="allow"> <node type="allow" cidr="192.168.241.0/24"/> </list> In the sofia profile is set: <param name="local-network-acl" value="localnet.auto"/> <param name="apply-inbound-acl" value="localnet.auto"/> The following SDP is send by the client: v=0 o=- 222075295642546336 815544600 IN IP4 127.0.0.1 s=IceLink t=0 0 m=audio 30450 RTP/SAVPF 0 8 c=IN IP4 192.168.62.1 a=rtcp:30450 IN IP4 192.168.62.1 a=ice-ufrag:532479e3 a=ice-pwd:4b066f2e64633d218085fc39c49a443c a=sendrecv a=rtcp-mux a=fingerprint:sha-256 FD:DE:DB:44:C8:66:10:B0:DF:3A:D5:17:4D:89:0E:1A:39:0C:99:D3:AE:BE:1F:F2:D7:78:2E:8E:A6:6B:F1:36 a=setup:actpass a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ssrc:967296980 cname:c2088983 a=candidate:954dc3ed263236386e8615555070c83dg 1 udp 2130706331 192.168.241.2 44376 typ host a=candidate:954dc3ed263236386e8615555070c83dg 2 udp 2130706331 192.168.241.2 44376 typ host The output on the fs_cli: 2016-02-18 07:36:33.568868 [DEBUG] switch_channel.c:3759 (sofia/H3KSip/170) Callstate Change RINGING -> ACTIVE 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:2898 Set Codec sofia/H3KSip_B2Bua/170@192.168.241.2:5058 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2016-02-18 07:36:33.588870 [DEBUG] switch_core_codec.c:111 sofia/H3KSip_B2Bua/170@192.168.241.2:5058 Original read codec set to PCMU:0 2016-02-18 07:36:33.588870 [WARNING] switch_core_media.c:3266 NO candidate ACL defined, Defaulting to wan.auto 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.241.2:44376 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3296 Save audio Candidate cid: 2 proto: udp type: host addr: 192.168.241.2:443760 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtp candidate. 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3336 Searching for rtcp candidate. 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:3380 sofia/H3KSip_B2Bua/170@192.168.241.2:5058 no suitable candidates found. 2016-02-18 07:36:33.588870 [DEBUG] switch_core_media.c:4480 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2016-02-18 07:36:33.588870 [NOTICE] switch_channel.c:3798 Hangup sofia/H3KSip_B2Bua/170@192.168.241.2:5058 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION Thanks a lot!

2 2

NUBE to FreeSWITCH - Windows?
by Shawn Wheeler 25 Feb '16

25 Feb '16

I know there is a windows installer for FreeSWITCH. Does anyone know if this will work with Win 7 32bit? It is what I have a physical Test box Thank you in advance Shawn

3 2

libyuv problem
by Veerabhadrarao Kankatala 24 Feb '16

24 Feb '16

hello, I am getting following error while configrung freeswitch 1.6.5 ./configure (success) but when i run make command i am facing following error making all mod_fsv make[4]: Entering directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' Makefile:797: *** You must install libyuv-dev to build mod_fsv. Stop. make[4]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' make[3]: *** [mod_fsv-all] Error 1 make[3]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5' make: *** [all] Error 2 but i already installed libyuv packages yum install libyuv-devel Loaded plugins: langpacks, presto, refresh-packagekit Package libyuv-devel-0-0.17.20121221svn522.fc18.x86_64 already installed and latest version Nothing to do i would like to know which version of libyuv, libyuv-devel is require for freeswitch 1.6.5V I am using Fedora 18 64Bit, Please help me to come out from this problem Thanks in advance -- Thanks & Regards Veerabhadrarao Kankatala Software Developer (C & Unix ) PANAMAX INFOTECH LIMITED Mobile: +91-8401231249 Messenger Id: Skype: veerabhadrarao.kankatala E-mail: veerabhadrarao.kankatala(a)panamaxil.com

2 1

Do we have option invalid-sound-long in freeswitch IVR menu ?
by Naveen Tamanam 24 Feb '16

24 Feb '16

I would like to play different invalid sound for the first failure which would be long sound and some other invalid sound for next retries. Do we have any kind of option like greet-long and greet-short to play different invalid sounds, like invalid-long invalid-short ? -- Thanks & Regards, Naveen Tamanam

2 1

Lib yuv problem
by Kankatala Bhadra Rao 24 Feb '16

24 Feb '16

hello, I am getting following error while configrung freeswitch 1.6.5 ./configure (success) but when i run make command i am facing following error making all mod_fsv make[4]: Entering directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' Makefile:797: *** You must install libyuv-dev to build mod_fsv. Stop. make[4]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod/applications/mod_fsv' make[3]: *** [mod_fsv-all] Error 1 make[3]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/src/bhadra/freeswitch-1.6.5' make: *** [all] Error 2 but i already installed libyuv packages yum install libyuv-devel Loaded plugins: langpacks, presto, refresh-packagekit Package libyuv-devel-0-0.17.20121221svn522.fc18.x86_64 already installed and latest version Nothing to do i would like to know which version oflibyuv, libyuv-devel is require for freeswitch 1.6.5V I am using Fedora 18 64Bit, Please help me to come out from this problem Thanks in advance

1 0

Multiple Choice Caller-ID
by Steven Schoch 24 Feb '16

24 Feb '16

We use the Polycom SoundPoint IP phones in the office, which work great. One user has 3 different numbers to reach that phone (one is the main office number with an extension, the other 2 ring direct). These phones have 2 line buttons. I would like this user to be able to choose from 3 different Caller-ID numbers when placing a call. I already do this with 2 numbers by using the 2 different line buttons, but I'm trying to figure out a way to choose a 3rd number when placing a call. Any ideas will be appreciated. -- Steve

4 3