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Understanding XML CDR's <call-stats>
by Simon Ditner 20 Apr '17

20 Apr '17

I'm having a bit of trouble interpreting the jitter_ related data that I am getting out of XML CDR's in an attempt to measure how well a system is performing. For example, I place a G.711 call over a private LAN between two idle systems using PCMU, the originator being static in a continuous loop, and the destination being "echo", with jitterbuffer_ms=60:200:20 At the 'echo' end, I examine the call-stats section of the XML CDR (below), and I am not sure what to make of the jitter_ values -- for example max_variance of 7076.55, and a loss_rate of 116.67 -- as I haven't been able to determine from the source code thus far what they are measuring, and what units they are working in. Any pointers would be appreciated. Cheers, Simon ---- <call-stats> <audio> <inbound> <raw_bytes>817744</raw_bytes> <media_bytes>817744</media_bytes> <packet_count>4772</packet_count> <media_packet_count>4772</media_packet_count> <skip_packet_count>51</skip_packet_count> <jitter_packet_count>4774</jitter_packet_count> <dtmf_packet_count>0</dtmf_packet_count> <cng_packet_count>0</cng_packet_count> <flush_packet_count>0</flush_packet_count> <largest_jb_size>0</largest_jb_size> <jitter_min_variance>0.10</jitter_min_variance> <jitter_max_variance>7076.55</jitter_max_variance> <jitter_loss_rate>116.67</jitter_loss_rate> <jitter_burst_rate>115.67</jitter_burst_rate> <mean_interval>20.06</mean_interval> <flaw_total>0</flaw_total> <quality_percentage>100.00</quality_percentage> <mos>4.50</mos> </inbound> <outbound> <raw_bytes>812700</raw_bytes> <media_bytes>812700</media_bytes> <packet_count>4725</packet_count> <media_packet_count>4725</media_packet_count> <skip_packet_count>0</skip_packet_count> <dtmf_packet_count>0</dtmf_packet_count> <cng_packet_count>0</cng_packet_count> <rtcp_packet_count>0</rtcp_packet_count> <rtcp_octet_count>0</rtcp_octet_count> </outbound> <error-log> <error-period> <start>1492563921959035</start> <stop>1492563927519075</stop> <flaws>511194</flaws> <consecutive-flaws>0</consecutive-flaws> <duration-msec>5560</duration-msec> </error-period> </error-log> </audio> </call-stats>

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fs sending from the wrong ip?
by David Villasmil 20 Apr '17

20 Apr '17

Hello guys, i've got this weird situation where I have multiple interfaces on my FS. I have one default gateway with ip i.e. 1.2.3.245 and another with ip 172.16.110.6, on my dialplan i'm sending the bridge to 10.71.3.100 which is supposed to go out via 172.16.110.6, but i see fs sending on the correct interface but with a wrong ip: U 1.2.3.245:5080 -> 10.71.3.100:5060 INVITE sip:2222222@10.71.3.100 SIP/2.0. Via: SIP/2.0/UDP 1.2.3.245:5080;rport;branch=z9hG4bKXpZN0ZQQN6B9j. Max-Forwards: 69. From: "1111111" <sip:1111111@1.2.3.245>;tag=Zv4XUZNaDamHc. To: <sip:2222222@10.71.3.100>. Call-ID: 1eca3592-9ef9-1235-63ae-0026b955dc81. CSeq: 105930868 INVITE. Contact: <sip:gw+gw_int2expresso@1.2.3.245:5080 ;transport=udp;gw=gw_int2expresso>. User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ca9207aa32~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 275. X-cid: MGNiZjNlZTA1NmY0YThlNjAyZTZkNjNjN2NmYTViNTM.. X-FS-Support: update_display,send_info. Remote-Party-ID: "1111111" <sip:1111111@1.2.3.245 >;party=calling;screen=yes;privacy=off. my routes: root@HayoSW:~# ip route default via 1.2.3.241 dev eth0 metric 100 10.10.0.0/24 dev eth0 proto kernel scope link src 10.10.0.112 10.71.0.0/24 via 172.16.110.5 dev eth1 172.16.110.4/30 dev eth1 proto kernel scope link src 172.16.110.6 1.2.3.240/29 dev eth0 proto kernel scope link src 1.2.3.245 Help is greatly appreciated! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Clarification on presence_data_cols
by Colin Morelli 19 Apr '17

19 Apr '17

Hey list, I'm trying to use presence_data_cols to get information from the B-leg of a call when a bridge starts. I've added presence_data_cols=chanvar1:chanvar2 The variables show up under PD- in the bridge request, which is *great*, but - I was looking for some clarification on the docs. The docs seem to indicate that presence_data_cols can be used to specify additional columns in the channel table. I've disabled all sql mods and it appears it's just pulling variables off of the channel. Is this expected, or am I going to run into issues with this? Thanks, Colin

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Data channel on WebRTC without Verto
by Agustí Ubalde 19 Apr '17

19 Apr '17

Hi all, Is possible to establish a call between 2 peers and send info througth data channel without Verto module? Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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simple portal for CDR along with call recording
by Bipin Patel 19 Apr '17

19 Apr '17

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Re: [Freeswitch-users] Update SDP on 183
by jaz 19 Apr '17

19 Apr '17

... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Update-SDP-on-183-tp7596303p7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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How to pause/resume conference recording
by Chad Phillips 18 Apr '17

18 Apr '17

I’m using the following forms to attempt pausing/resuming a conference recording that has been started: conference <confname> recording pause <file-path> conference <confname> recording resume <file-path> I’ve verified the file in question has started recording successfully, and I’m getting a success response of “Pause/Resume recording file” from my API command. However, when inspecting the file after pausing it, I see that it’s still being written to. Looking at the conference_record_action() function, it appears what it does is set a flag on each individual member, MFLAG_PAUSE_RECORDING, when a recording is to be paused. At this point I’m not sure I’m clear what the result of the above API commands are — what are they intended to do? And is there a way to completely stop writing additional data to a recording file when the recording is paused?

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inbound-late-negotiation with pickup channels
by Colin Morelli 18 Apr '17

18 Apr '17

Hey all, So I'm using inbound-late-negotiation and inherit_codec on calls to try to get the A and B legs to agree on a codec and reduce the frequency of transcoding. For direct calls, this seems to work fine. I can't say the same about pickup channels, though. I expected that the pickup application would work with inherit_codec the same way, since codec negotiation still doesn't start until the channel is picked up, but it doesn't seem to be happening that way. Instead, the channel that is picking up the parked channel seems to pick a codec based on its own preference. Is there something I'm missing here - or is this possibly a bug? Best, Colin

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Frequent Call Drops in Freeswitch conference call in GSM network
by Deepika Yadav 18 Apr '17

18 Apr '17

Hi, I involved in a research project where we have developed a mobile based training system. Our server is Freeswitch based. The server on receiving request from an android app establishs conference call amongst 21 participants over the cellular network by calling on their mobile numbers. To achieve this, a VOIP-GSm gateway service is used offered by Doorvaani. The participants locations are rural areas of India. We need to maintain the conference call for atleast 1 hour as there is some kind of training program. We have tested our training platform by conducting 10 such conference calls. However, one persistent problem that we are facing is frequent call drops. In between of an on-going conference, some of the participants faces call drops, to which we have adopted a workaround of redialing them. One of the potential cause is the poor signal state in some areas, but there are many people who have decent signal strength and still face call drops. We are not sure how can we investigate the matter as neither the VOIP-GSM logs are available for us nor can we setup conference call with more than 5-6 participants using existing service providers. Is this a problem from Freeswitch, can I debug? or should we setup our own VOIP-GSM gateway as currently we are using web service from a company we can not physically access the logs. I would be happy to get suggestions on how to deal with this. Regards Deepika https://www.iiitd.edu.in/~deepikay/

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FS 10061 - send custom variables on incoming call via verto
by Gregor Nanger 18 Apr '17

18 Apr '17

Can someone please elaborate little on this functionality, what is meant with this feature? -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si

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