- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] One-way audio but not video on 10.4
by David P 17 Sep '20

17 Sep '20

Although we have STUN+TURN coturn (and FS) deployed in Oregon, a verto user in San Francisco today made a call in which only (private) host candidates were in the offer bundle SDP. They use IT-managed Ethernet, so any slowness in getting STUN is probably in their IT management of the connection...and not something we can do anything about unless our FS can support peer-reflexive candidates. They tell me their network has worked fine with Zoom, which is grating to hear. Is there any prospect of FS supporting peer-reflexive candidates? On Tue, Sep 1, 2020 at 12:22 AM < freeswitch-users-request(a)lists.freeswitch.org> wrote: > ---------- Forwarded message ---------- > From: David P <davidswalkabout(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 31 Aug 2020 08:36:05 +1200 > Subject: Re: [Freeswitch-users] One-way audio but not video on 10.4 > >> I strongly suspect the problem is due to FS choosing a different port > than is offered. I've seen this a few times by comparing the offer SDP > a=candidate entries with the address:port in FS log entries containing > 'Choose rtp'. This won't work if the user is behind a firewall that allows > incoming response only through ports opened for outgoing requests. > > Another thing I wonder about is when verto provides a BUNDLE offer, should > it use the same address and port for both audio and video. I think I've > seen that it doesn't, but maybe BUNDLE doesn't require it. > > --------- Forwarded message ---------- >> From: Nathan Stratton <nathan(a)robotics.net> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Bcc: >> Date: Sat, 29 Aug 2020 11:13:24 -0400 >> Subject: Re: [Freeswitch-users] One-way audio but not video on 10.4 >> Looks like your client is not hitting a stun server to get the public IP >> address, however, I have been wondering why it's not possible to make >> FreeSWITCH work without using STUN on the client as long as FreeSWITCH is >> on a public IP. >> >> ><> >> nathan stratton >> >> >> On Sat, Aug 29, 2020 at 3:00 AM David P <davidswalkabout(a)gmail.com> >> wrote: >> >>> The client is current verto running on current Chrome/Win10. >>> >>> The client's SDP shows srflx and relay candidates from Twilio at same >>> network cost as the "private" host candidates, and FSv10.4 chooses the host >>> candidates. >>> >>> ---------- Forwarded message ---------- >>>> From: Mike Jerris <mike(a)freeswitch.org> >>>> >>>> If client still isn’t putting in non rfc1918 candidates in audio you >>>> need to figure out why that is first. >>>> >>>> On Aug 26, 2020, at 7:17 PM, David P <davidswalkabout(a)gmail.com> wrote: >>>> >>>> I've introduced Twilio STUN and TURN to our iceServers since then. All >>>> have the same network cost in the SDP and FSv10.4 chooses a host. Verto >>>> reports the remote stream arrived but getStats shows zero audio bytes in or >>>> out, and it's silent to our user. I applied a display filter using our >>>> Freeswitch's public IP ip.src == 52.xx.yy.zz or ip.dst == ip.src == >>>> 52.xx.yy.zz and in addition to TLS I see some TCP ACKs from our user's >>>> 10.0.0.189 but I don't know what else to look for. >>>> >>>> >>>> ---------- Forwarded message ---------- >>>>> From: Mike Jerris <mike(a)freeswitch.org> >>>>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>>>> Cc: >>>>> Bcc: >>>>> Date: Wed, 26 Aug 2020 16:19:56 -0400 >>>>> Subject: Re: [Freeswitch-users] One-way audio but not video on 10.4 >>>>> >>>> >>>> >>>>> Configure stun in your client so you get proper candidates that are >>>>> reachable. >>>>> >>>>> On Aug 19, 2020, at 6:18 PM, David P <davidswalkabout(a)gmail.com> >>>>> wrote: >>>>> >>>>> Yesterday I had a verto call in which the <video> contained a remote >>>>> echo (which is how we configure it) but there was no audio out nor in >>>>> (confirmed by getStats showing zero audio bytes out or in). >>>>> >>>>> I was using: >>>>> FreeSWITCH Version >>>>> 1.10.4-release+git~20200805T110119Z~133fc2c870~64bit (git 133fc2c >>>>> 2020-08-05 11:01:19Z 64bit) >>>>> from Chrome84/Win10 >>>>> >>>>> I noticed the video candidates included a relay but the audio ones >>>>> didn't: >>>>> >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e m=audio 62058 UDP/TLS/RTP/SAVPF >>>>> 111 103 104 9 0 8 106 105 13 110 112 113 126 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e c=IN IP4 192.168.208.1 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=rtcp:9 IN IP4 0.0.0.0 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:1252570982 1 udp >>>>> 2122260223 192.168.208.1 62058 typ host generation 0 network-id 1 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:854729925 1 udp >>>>> 2122194687 172.20.4.169 62059 typ host generation 0 network-id 2 >>>>> network-cost 10 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:69946262 1 tcp >>>>> 1518280447 192.168.208.1 9 typ host tcptype active generation 0 network-id 1 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:2087835701 1 tcp >>>>> 1518214911 172.20.4.169 9 typ host tcptype active generation 0 network-id 2 >>>>> network-cost 10 >>>>> >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e m=video 13143 UDP/TLS/RTP/SAVPF >>>>> 96 97 98 99 100 101 102 122 127 121 125 107 108 109 124 120 123 119 114 115 >>>>> 116 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e c=IN IP4 35.160.47.89 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=rtcp:9 IN IP4 0.0.0.0 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:1252570982 1 udp >>>>> 2122260223 192.168.208.1 62060 typ host generation 0 network-id 1 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:854729925 1 udp >>>>> 2122194687 172.20.4.169 62061 typ host generation 0 network-id 2 >>>>> network-cost 10 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:69946262 1 tcp >>>>> 1518280447 192.168.208.1 9 typ host tcptype active generation 0 network-id 1 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:2087835701 1 tcp >>>>> 1518214911 172.20.4.169 9 typ host tcptype active generation 0 network-id 2 >>>>> network-cost 10 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:3112167340 1 udp >>>>> 25042943 35.jj.kk.mm 13143 typ relay raddr 210.55.134.98 rport 53504 >>>>> generation 0 network-id 2 network-cost 10 >>>>> >>>>> The relay was chosen for video, and it appeared that the same >>>>> candidate was going to be chosen for audio, but then an "AUDIO RTP" >>>>> suggests it chose a private IPv4. And it seems that private IPv4 wasn't >>>>> treated as a peer-reflexive candidate, so the media couldn't be routed... >>>>> >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>> [DEBUG] switch_core_media.c:4338 Choose rtp candidate, index 2, >>>>> 35.jj.kk.mm:13143 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>> [DEBUG] switch_core_media.c:4104 >>>>> verto.rtc/1200_a6dd33a4-1def-447c-a62a-ae807bb92da6_v-1*s-4 choosing family >>>>> v4 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>> [DEBUG] switch_core_media.c:4349 Choose same candidate, index 0, for rtcp >>>>> based on rtcp-mux attribute 35.jj.kk.mm:13143 >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>> [DEBUG] switch_core_media.c:4401 setting remote video ice addr to index 2 >>>>> 35.jj.kk.mm:13143 based on candidate >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>> [DEBUG] switch_core_media.c:4436 Setting remote rtcp video addr to >>>>> 35.jj.kk.mm:13143 based on candidate >>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>> [DEBUG] switch_core_media.c:8663 AUDIO RTP >>>>> [verto.rtc/1200_a6dd33a4-1def-447c-a62a-ae807bb92da6_v-1*s-4] 10.0.0.100 >>>>> port 26610 -> 192.168.208.1 port 62058 codec: 111 ms: 20 >>>>> >>>>> Am I interpreting the log correctly? >>>>> >>>>> Is there a way to avoid this? >>>>> >>>>> Note that in /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml >>>>> we already deny private IPv4 (although I think we should allow them so they >>>>> could be used peer-reflexively): >>>>> <list name="disallow-privateIPv4" default="allow"> >>>>> <node type="deny" host="10.0.0.0" mask="255.0.0.0"/> >>>>> <node type="deny" host="172.16.0.0" mask="255.240.0.0"/> >>>>> <node type="deny" host="192.168.0.0" mask="255.255.0.0"/> >>>>> </list> >>>>> >>>>> >>>> >>> _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com

2 1

Add a=sendrecv onto sdp
by Oleg Belousov 17 Sep '20

17 Sep '20

Hi. Please advise how I can add a=sendrecv parameter onto the SDP part of 183 Session progress. I'm using preAnswer in lua script to respond 183 with SDP to an invite. -- obelousov.tel

3 3

HA keep alive not working for me
by Joli Martinez 17 Sep '20

17 Sep '20

Hello, I am following the keepalive instructions below. Both sites appear to have the same config setup as far as the keepalive is concerned. I am running this on Debian 10 and using the latest version of freeswitch. I have setup both the primary and backup nodes as described, but when I attempt to restart freeswitch it fails. Looking at the logs I keep getting the output below. It appears some parameters are not being passed. Attached is my sanitized files. Sep 16 21:21:52 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:21:53 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:21:54 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:21:55 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:21:56 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:21:57 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:21:58 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:21:59 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:22:00 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:22:01 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:22:02 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:22:03 dev-sca02a ka-notify: what() id() state() prio() Sep 16 21:22:04 dev-sca02a ka-notify: what() id() state() prio()

1 0

Re: [Freeswitch-users] Experience with FreeSWITCH on capable servers?
by Sam van Herwaarden 17 Sep '20

17 Sep '20

That sounds very encouraging. Is the way that FreeSWITCH loads the system when bridging calls similar to when it plays WAVs? I could imagine that the latter would do more IO (likely cached so from RAM). I'll see what I can test - to some extent we will have to make the server purchase decision without knowing how that exact hardware will perform. Thanks! Sam On Thu, Sep 17, 2020 at 12:06 AM Alexis <ap(a)gen-ip.fr> wrote: > Hi, > Some of our servers worked (in production) without major issues at 3k > channels, on a E3-1230 v6 (4 cores, 8 threads, 3.5GHz, turbo at 3.9GHz, > 2017) under freeswitch 1.6 on CentOS 7. Those servers acted as SBC, they > just did some DB lookup and bridged the calls. No transcoding at all, but > they handled RTP. The CPU was quite full, I don't recommend at all to put > this load on such a small CPU, but it worked. Each CPU core was used at > 80-85% so that was an heavy load. > We just had some memory leaks, but it may be related to some specific > unoptimized perl scripts executed by freeswitch. > > Soooo yes, FreeSWITCH can handle 3k+ channels easily. That just depend on > what you want to do with your calls (and how you do it ;)). > There is no secret here, you'll have to test what you want in real the > world to see if your server can handle what you want. > > Alexis > Le 16/09/2020 à 19:10, Sam van Herwaarden a écrit : > > The server is going to be streaming audio on each of these channels > (8000Hz mono wav). I’m not so familiar with all the telephony terminology > yet so not sure if that’s similar to what it would do when bridging calls? > > If I run into software limitations I can probably also containerize or use > KVM or so. > > What kind of channel counts did you manage on what kind of machine? I > think the FreeSWITCH benchmarks page is a little bit outdated and doesn’t > really have examples of 1000+ calls and modern hardware. Would be great to > have some reference points. > > Thanks! > > ons 16 sep. 2020 kl. 18:57 skrev Abaci B <abaci64(a)gmail.com>: > >> The machine you describe shouldn't have an issue handling more than 3000 >> calls if all you do is bridging calls, if you do more complex stuff you >> would probably have to test. >> I personally used more that on a single machine with much lower specs but >> I split them in multiple containers (lxc) as I was getting audio problem >> when somewhere between 2000 & 3000 channels doing audio conferencing on a >> single instance (same limits with 4, 8 or 16 cores) that I never was able >> to figure out. >> >> On Wed, Sep 16, 2020 at 11:12 AM Sam van Herwaarden < >> svanherwaarden(a)precisionag.org> wrote: >> >>> Thanks! >>> >>> Not sure I understand - FreeSWITCH itself can easily make use of >>> multiple cores, right? Is there a benefit to running multiple instances of >>> FreeSWITCH on the same node? >>> >>> To clarify - ideally I'd handle as many channels as possible on a single >>> server. I'm wondering how many channels on one server is possible on modern >>> hardware, and I've been assuming that core count is the most important >>> factor there but if there are other bottlenecks to be aware of (e.g. RAM >>> bandwidth) that would be really useful to hear about as well. >>> >>> Best, >>> Sam >>> >>> >>> >>> On Wed, Sep 16, 2020 at 5:05 PM Sergey Safarov <s.safarov(a)gmail.com> >>> wrote: >>> >>>> You can use kamailio as frontend and multiple freeswitch on backend. >>>> Need to use kamailio dispatcher module. >>>> >>>> On Wed, Sep 16, 2020, 12:46 PM Sam van Herwaarden < >>>> svanherwaarden(a)precisionag.org> wrote: >>>> >>>>> Hi all, >>>>> >>>>> I'm working on a FreeSWITCH setup that places phone calls over a SIP >>>>> line where we have a capacity of 3000 channels. What I'm not sure about yet >>>>> is if this can be done on a single server or if we will need to run >>>>> multiple servers in parallel. From what I've read FreeSWITCH parallelizes >>>>> well. >>>>> >>>>> Does anyone here have experience running FreeSWITCH on powerful >>>>> servers and how far can you get with a single host? >>>>> >>>>> I'm thinking about something like a Threadripper setup with 32 cores >>>>> or so and let's say 64GB of RAM (but if we would need more that's probably >>>>> possible as well). >>>>> >>>>> Any other advice about hardware requirements for FreeSWITCH setups on >>>>> high capacity systems would be welcome as well! >>>>> >>>>> Thank you, >>>>> Sam >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> The FreeSWITCH project is sponsored by SignalWire >>>>> https://signalwire.com >>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>> services. >>>>> Build your next product on our scalable cloud platform. >>>>> >>>>> Join our online community to chat in real time >>>>> https://signalwire.community >>>>> >>>>> Professional FreeSWITCH Services >>>>> sales(a)freeswitch.com >>>>> https://freeswitch.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> https://freeswitch.com/oss >>>>> https://freeswitch.org/confluence >>>>> https://cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users(a)lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://freeswitch.com >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> Professional FreeSWITCH Services >>>> sales(a)freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales(a)freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales(a)freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Alexis Prodhomme > Support Technique Gen-IP > email : support(a)gen-ip.fr > tel : 02.90.75.30.50 > >

1 0

Find out answered agent in fifo
by Morteza Bashsiz 16 Sep '20

16 Sep '20

Hi Hope you are well I want to find out which agent in fifo has answered the call and also when answered the call How can i find it Thank you very much for your contribution

1 0

Problem with the "file-only" Conference Layout Option
by Lyle Pratt 15 Sep '20

15 Sep '20

Howdy Folks, I'm experimenting with some video capabilities and am trying to play a video file in a specific layout position in a conference. From the documentation, it seemed like I could use the "file-only" option in a layout configuration to achieve this, however, even with the layout applied on the conference, video files always play full screen. >From the docs: file-only with file-only="true" this box will only show video files that are played via the conference play api. Does the "file-only" option no longer work? Any other tips? Thanks! Lyle Pratt

2 1

[mod_amqp][golang]
by Bilal Abbasi 15 Sep '20

15 Sep '20

Hi Users, I am using mod_rabbitmq to send events on rabbitmq and trying to capture/consume using golang. I saw that there are certain things we need from mod_amqp, like queue name,binding key, consumer tag. I cant see them under amqp.conf and they are required by golang amqp script https://github.com/streadway/amqp /_examples/simple-consumer/consumer.go Need help Regards Abbasi

3 4

issue with mod_python
by David Villasmil 12 Sep '20

12 Sep '20

Hello guys, i¡m not sure what's going on. I just installed a fresh 1.10.5 and i've got a script in # ls -l /usr/share/freeswitch/scripts/foo-outbound.py -rw-r--r-- 1 root root 3804 Sep 8 23:44 /usr/share/freeswitch/scripts/foo-outbound.py but FS either can't find it or something's changed in the architecture: 2020-09-09 00:05:08.548244 [NOTICE] mod_python.c:213 Invoking py module: foo-outbound 2020-09-09 00:05:08.548244 [ERR] mod_python.c:253 Error importing module 2020-09-09 00:05:08.548244 [ERR] mod_python.c:165 Python Error by calling script "foo-outbound": <type 'exceptions.ImportError'> Message: No module named foo-outbound l> status UP 0 years, 0 days, 0 hours, 26 minutes, 27 seconds, 2 milliseconds, 76 microseconds FreeSWITCH (Version 1.10.5 -release-17-25569c1631 64bit) is ready 6 session(s) since startup 0 session(s) - peak 1, last 5min 0 0 session(s) per Sec out of max 30, peak 1, last 5min 0 1000 session(s) max min idle cpu 0.00/90.07 Current Stack Size/Max 240K/8192K Anyone knows anything about this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 3

Early media in the "wrong" direction
by Olle Frimanson 11 Sep '20

11 Sep '20

Hi, We have a use case where we need to send early media from A-leg and play it before 200 OK on multiple B-legs. This the scenario in principle: A sends an INVITE to Freeswitch Freeswitch responds with 183 A starts sending media to Freeswitch Freeswitch does parallell bridging to B,C and D B,C and D responds with 183 Now I want Freeswitch to relay the media transmitted from A to B,C and D Is this scenario supported by Freeswitch I have various options like audio_media_flow and bridge_early_media but I can't get it to work. Any help or hints are appreciated BR/Olle

2 3

[In Progress/Ringing Calls][FreeSWITCH Event]
by Bilal Abbasi 10 Sep '20

10 Sep '20

Hi Users, I am trying to get in progress call event from freeswitch, i saw that we have channel_progress and channel_progress_media events, but those are not getting executed all the time. Can someone guide me what event i need to capture to show current running calls, basically it would be a simple counter getting increment on new channel in progress/ringing and decrement on channel_hangup_complete event. Regards Abbasi

2 2