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uuid_getvar / uuid_setvar Safety
by Colin Morelli 22 Nov '16

22 Nov '16

Hey all, This should hopefully be a very simple question. If I have a variable in one channel that I want to copy to another (these are unrelated, un-bridged channels) is it safe to just do, for example uuid_setvar some_var ${uuid_getvar other-call-id some_var} (from the dialplan)? Is there a better alternative? Without being an expert in how the variable expansion works, I want to make sure this would work for all possible values of the var, and I won't be caught off-guard with buffer size issues, separator issues, etc. Thanks in advance, Colin

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mod_fifo and early media
by Dominique Jeannerod 22 Nov '16

22 Nov '16

Hello, i'm working on a project where I need to manage a call waiting queue, in early media mode, to have a free waiting time, and start billing only after the call is really answered. The general call flow is : 1- incoming call to a service number 2- early media answer (pre-answer) to manage voice messages 3- the call is put in a waiting queue, and the caller ears a message (MOH), still in early media 4- the queue is associated to a unique destination number, and then one an only call at a time is picked-up from the queue and bridged to the destination. 5- When the call is established (200 OK), the incoming call is also answered (200 OK), which starts the billing. Question : this looks like a basic call queue handling, and mod_fifo could be a perfect fit for that ... except that mod_fifo answers the call with a 200 OK Is it possible to manage early media with mod_fifo ? Does someone have a best practice, experience, or advice to share on this matter ? Thanks with anticipation D. Jeannerod

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one way audio coming off hold
by Jarad Morgan 22 Nov '16

22 Nov '16

Hello, we are running into some issues with one way audio coming off hold with messaging between Avaya's. Running FS 1.6.10. Is there a FS config setting to leave out the media type attribute (sendonly,sendrecv) in the 200 OK sent by FS when doing late media negotiation between a Avaya device? Thanks!!

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aravindan@teledna.com
by vijaya alavalpati 22 Nov '16

22 Nov '16

Hi , I have configured freeswitch server for making one-one call.I am making test call from IMSDroid(https://github.com/DoubangoTelecom/imsdroid) android client to SIPML5 client( i.e https://www.doubango.org/sipml5/call.htm#) When both are in the same wifi call is working. But when my android phone ison 3g and SIPML5 client is on wifi, call is established but voice is notaudible. Can you please help us what is the issue and how to fix. My Freeswitchserver is natted to a public IP. Following is the invite going from my android client.1004 user is SIPLM5 client and 1003 user is Android client. INVITE sip:1004@freeswitch.nowconfer.comSIP/2.0 Via: SIP/2.0/TCP10.44.15.65:50583;branch=z9hG4bK-600328989;rport From: <sip:1003@freeswitch.nowconfer.com>;tag=1909344009 To: <sip:1004@freeswitch.nowconfer.com> Contact: <sip:1003@10.44.15.65:50583;transport=tcp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Call-ID: bfaff8ce-8a80-cbdc-c106-7713da878fcb CSeq: 571345640 INVITE Content-Type: application/sdp Content-Length: 905 Max-Forwards: 70 Accept-Contact:*;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Allow: INVITE, ACK, CANCEL, BYE, MESSAGE,OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none User-Agent: IM-client/OMA1.0android-ngn-stack/v2.586.1328 (doubango r1328 - MotoG3) P-Preferred-Identity: <sip:1003@freeswitch.nowconfer.com> Supported: 100rel v=0 o=doubango 1983 678902 IN IP4 10.44.15.65 s=- c=IN IP4 10.44.15.65 t=0 0 a=tcap:1 RTP/AVPF m=audio 45812 RTP/AVP 98 97 8 0 113 112 101 1119 a=ptime:20 a=minptime:1 a=maxptime:255 a=silenceSupp:off - - - - a=rtpmap:98 SPEEX/16000/1 a=rtpmap:97 SPEEX/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:113 AMR/8000/1 a=imageattr:113 octet-align=1 a=fmtp:113 octet-align=1 a=rtpmap:112 AMR/8000/1 a=imageattr:112 octet-align=0 a=fmtp:112 octet-align=0 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-16 a=rtpmap:111 opus/48000/2 a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000; stereo=0; sprop-stereo=0; useinbandfec=0; usedtx=0 a=rtpmap:9 G722/8000/1 a=pcfg:1 t=1 a=sendrecv a=rtcp-mux a=ssrc:4194994927cname:a797995ff45befd0f562f852791dc543 a=ssrc:4194994927mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2 a=ssrc:4194994927 label:doubango@audio RegardsVijaya

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Help calling phrase macros from Lua
by Ned Ludd 21 Nov '16

21 Nov '16

So I'm trying to develop a simple voicemail IVR. In "sounds.xml" there's a nice macro for "voicemail_message_count", but I can't seem to figure out how to get the count to feed it. Can you help? I know I can use this to get the count in JSON format: count = api:executeString("vm_fsdb_msg_count default default " .. domain .. " " .. extension .. " inbox"); But how do I get a count that I can send to that phrase macro? Confused. (And excuse my ignorance). ----- Ned Ludd

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FS not receiving incoming calls
by Ignacio Durli 21 Nov '16

21 Nov '16

I upgraded to FS 1.6.12 which I installed in a new Debian 8 instance ran in VirtualBox. After compiling and running all the config scripts I'm not able to establish a SIP call. After some troubleshooting I found that calls are being answered on my eth1 interface (10.0.x.x) but not on my eth0 interface (192.168.99.x). The IP is configured in vars.xml properly and sofia status is showing: freeswitch@freeswitch> sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6 profile sip:mod_sofia@[::1]:5081 RUNNING (0) (TLS) external profile sip:mod_sofia@192.168.99.103:5080 RUNNING (0) external profile sip:mod_sofia@192.168.99.103:5081 RUNNING (0) (TLS) 192.168.99.103 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5061 RUNNING (0) (TLS) internal profile sip:mod_sofia@192.168.99.103:5060 RUNNING (0) internal profile sip:mod_sofia@192.168.99.103:5061 RUNNING (0) (TLS) ================================================================================================= 4 profiles 1 alias Which looks fine. Also the ifconfig of the new Debian instance seems OK, since it's the same that I have on another VM where an older FS version is running. root@osboxes:/usr/local/freeswitch/bin# ifconfig eth0 Link encap:Ethernet HWaddr 08:00:27:3a:f0:c6 inet addr:192.168.99.103 Bcast:192.168.99.255 Mask:255.255.255.0 inet6 addr: fe80::a00:27ff:fe3a:f0c6/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:2482 errors:0 dropped:0 overruns:0 frame:0 TX packets:1034 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:353837 (345.5 KiB) TX bytes:227253 (221.9 KiB) eth1 Link encap:Ethernet HWaddr 08:00:27:f7:f1:24 inet addr:10.0.3.15 Bcast:10.0.3.255 Mask:255.255.255.0 inet6 addr: fe80::a00:27ff:fef7:f124/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:4298 errors:0 dropped:0 overruns:0 frame:0 TX packets:7037 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:437241 (426.9 KiB) TX bytes:829683 (810.2 KiB) lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:65536 Metric:1 RX packets:49100 errors:0 dropped:0 overruns:0 frame:0 TX packets:49100 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:8431576 (8.0 MiB) TX bytes:8431576 (8.0 MiB) I honestly don't know what's going on. Acls are fine, and I don't see any error in the FS log. I traced the network traffic and when trying to call to 192.168.99.103:5060 I get an ICM (Destination unreachable) which doesn't make sense according to what sofia status is showing. Any hint on this?

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Members in fifo at startup
by Jose Fco. Irles Durá 21 Nov '16

21 Nov '16

Hi, I'm testing the fifo module with the default configuration (configuration packaged in debs) and I am not able to make it work. The default configuration (/etc/freeswitch/autoload_configs/fifo.conf.xml): <configuration name="fifo.conf" description="FIFO Configuration"> <settings> <param name="delete-all-outbound-member-on-startup" value="false"/> </settings> <fifos> <fifo name="cool_fifo@$${domain}" importance="0" outbound_per_cycle="3" > <member timeout="10" simo="1" lag="5">{fifo_member_wait=nowait}user/1001@$${domain}</member> <member timeout="10" simo="1" lag="5">{fifo_member_wait=nowait}user/1002@$${domain}</member> </fifo> </fifos> </configuration> I can see two members in the "cool_fifo" (1001 and 1002) configured, but when I start freeswitch, If I execute the command "fifo list" I don't see thats members (the fifo has no members) Only when I execute the command "fifo reparse" after freeswitch is started I can see the "cool_fifo" with the members. Where is the problem? Should I execute "fifo reparse" once freeswitch is started? Thanks in advance -- Jose Fco. Irles Durá

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bypass_media after receiving invite with replaces header
by Sjoerd Boomstra 21 Nov '16

21 Nov '16

Hello, I've got a problem with FreeSwitch version 1.4.26. Our setup is like this: We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). We aim to bypass the media at the FreeSwitch. At first this works as expected, the media does not go through the FreeSwitch. However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. The original channel has the variables bypass_media and bypass_media_after_bridge set to true. It seems that these variables are not checked when the invite with replaces header is handled. Anyone an idea how to fix this? Best regards, Sjoerd. ----- Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E‑mail: s.boomstra(a)telecats.nl |

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Editing confluence
by David Villasmil 20 Nov '16

20 Nov '16

Hello guys, I'm trying to edit https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia to document the new "sofia filter", but i don't seem to have permissions? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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How to play multiple video files (not video calls) in a conference
by Saumar Hajjar 19 Nov '16

19 Nov '16

Hi, I'm working in a PoC and I'm considering FS. I'd like to create a conference and have several video files playing simultaneously in the canvas. I already tried: conference name play av:///var/www/vid/video.mp4 and it works great. But I need multiple files playing and apparently the above command queues a video file (or freezes if it's a rtsp stream... Later I'll confirm this and file a Jira) I also tried creating a loopback leg, joining the conference, and playing a video just to this particular member. But it fails because the member doesn't support video. Basically I'd like to do something like the examples found at the bottom of https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video. Probably I'm missing some trivial stuff... I really appreciate any advice on this. Thanks in advance, Saumar

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