April 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Check if media is ready on originate response
by Oleg Stolyar 24 Apr '17

24 Apr '17

Hi guys, I use originate with multiple providers. In some cases I get 183 but in others I only get 180 without SDP. On originate return I execute a dialplan. I want to use return_ring_ready, so the dialplan starts early and I can process the call. I am looking for a way to check in the dialplan if the media is ready. If it is, I will process the call, if it's not, I will use execute_on_media to process the call. Is there a way to do that in the dialplan?

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Internal Profile not setting EXTERN IP
by Andrew Paul 24 Apr '17

24 Apr '17

Hi all, we have FREESWITCH behind opensips and freeswitch is running in AWS. Both Opensips and Freeswitch is in same network + Softphone SDP have public IP address. SOFTPHONE --> OPENSIPS --> ( profile : internal) FREESWITCH ( external) --> CARRIER --> CUSTOMER Freeswitch is action as mediaserver here , I can see the extern-rtp-ip is setting on the calls to CARRIER, But when freeswitch 200Ok to Opensips, it is not putting external IP. It is setting the freeswitch local ip. I have followed the instruction in https://freeswitch.org/ confluence/display/FREESWITCH/Amazon+EC2 , but not sure why FS is not setting external IP for internal profile. Is there any way to force the EIP ( public IP ) to SDP to opensips?. I can see the Contact address have Public IP but not the SDP Connection IP. Internal.xml <param name="inbound-zrtp-passthru" value="false"/> <param name="nonce-ttl" value="60"/> <param name="disable-register" value="true"/> <param name="auth-calls" value="$${internal_auth_calls}"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="sip-ip" value="192.168.0.173"/> <param name="rtp-ip" value="192.168.0.173"/> <param name="ext-sip-ip" value="54.81.198.142"/> <param name="ext-rtp-ip" value="54.81.198.142"/> <param name="aggressive-nat-detection" value="true"/> send 1379 bytes to udp/[192.168.0.132]:5060 at 08:53:33.106380: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.132:5060;branch=z9hG4bKe04b.c7469955.0;rport= 5060 Via: SIP/2.0/UDP 182.75.26.194:58618;rport=58618;received=182.75.26.194; branch=z9hG4bK-524287-1---8cfb5fbe8c5d0f23 Record-Route: <sip:192.168.0.132;r2=on;lr;ftag=7325490b;vsf= AAAAAH1uFG18am1hcXBAIRUIFgwREU0YRxoMAAoOQRtBDlQZcmFuc3BvcnQ9 VURQ;did=ad3.3f713b01> Record-Route: <sip:54.175.200.193:5060;r2=on;lr;ftag=7325490b;vsf= AAAAAH1uFG18am1hcXBAIRUIFgwREU0YRxoMAAoOQRtBDlQZcmFuc3BvcnQ9 VURQ;did=ad3.3f713b01> From: "14154839556" <sip:14154839556@calldirect.piba.com>;tag=7325490b To: <sip:14154839556@calldirect.piba.com;transport=UDP>;tag=7Ury0X9HXp68j Call-ID: 7sgGUmlws7Bg64kCbksywQ.. CSeq: 2 INVITE Contact: <sip:14154839556@54.81.198.142:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 222 Remote-Party-ID: "Outbound Call" <sip:14154839556@calldirect.piba.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1492664761 1492664762 IN IP4 192.168.0.173 s=FreeSWITCH c=IN IP4 192.168.0.173 t=0 0 m=audio 28052 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Attaching the logs for reference. Regards Andrew

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[mod_conference] limit the number of conferences
by Mititelu Stefan 23 Apr '17

23 Apr '17

Hello, I've been wondering... Is there a way to limit the maximum number of conferences spawned on a Freeswitch machine? One can definitely limit the number of conference members via 'max-members', but I couldn't find anything related to 'max-conferences'. Cheers, Stefan

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Anything like Asterisk's app_jack in FS?
by Oivvio Polite 23 Apr '17

23 Apr '17

In Asterisk it's possible to route call audio in an out of a jack audio server and do arbitrary audio processing in jack. [1] Is anything similar available in FS? oivvio [1] https://www.voip-info.org/wiki/view/Asterisk+cmd+jack

3 4

Use inherit_codec Codec on Re-invite
by Colin Morelli 22 Apr '17

22 Apr '17

Hey all, back again Is there any option in FS to either: - Keep inherit_codec working after the first invite - Try to reduce codec changes by attempting to use the existing codec on a re-invite Unless I'm missing something, FS seems to ignore both the current codec being used, and the inherit_codec setting on re-invites, so it defaults to essentially restarting codec negotiation. Ideally I could have it set to preserve inherit_codec if the call is still bridged (i.e. try to negotiate the B-leg's codec onto the A-leg if it's supported by the A-leg's new SDP). Consider the following call: INVITE (this is good and what I'd expect): A --- (opus, PCMU) ---> FS --- (opus, PCMU) ---> B A <-- (PCMU) --- FS <--- (PCMU) --- B Later Re-INVITE: A --- (opus, PCMU) ---> FS A <-- (opus) --- FS Ideally, I could get something like: Later Re-INVITE: A --- (opus, PCMU) ---> FS (checks codec used on B-leg of call, if inherit_codec=true) A <-- (PCMU) --- FS Thanks in advance, Colin

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GSM Codec
by John Nash 22 Apr '17

22 Apr '17

Sorry may be a dumb question but I do not see GSM codec when I do show codec. Do I need to compile some module in order to have gsmm codec?

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Post-processing conference recording
by Chad Phillips 21 Apr '17

21 Apr '17

I’m looking to do some simple post-processing on a recording made via “conference foo recording start /tmp/bar.mp4” — in this case, just fix up the file permissions and move it to a web-accessible directory. I’m curious how others have approached this. My current plan is: - Fire up a simple event listener (probably a node.js service, running as the same user that runs the FreeSWITCH process) - Listen for ‘del-member’ conference event, where Conference-Size is 0 (I'd prefer some kind of 'conference-end' event, but didn't find any documentation saying something like that exists) - Locate the recording based on conference name, then handle the perms/moving Would love to hear a better/cleaner approach if anyone has one :)

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New "console filter feature"
by David Villasmil 20 Apr '17

20 Apr '17

Hello all, When you have some traffic on freeswitch, it's nearly impossible to use the cli for debugging... I've made a patch on mod_console to add a feature for filtering what's shown on the console. You provide a regex, and nothing is shown until that regex is found, when found, the filter will take the "UUID" of the call and only show the messages coming from that UUID. "console uuid" must be enabled for this to work, so by enabling the filter, the uuid feature is enabled automatically. I will do a PR this afternoon or tomorrow. Hope it's helpful. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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outbound proxy and fs_path
by Slava Bendersky 20 Apr '17

20 Apr '17

Hello Everyone, My setup FS1 ------| | Kamailio (Proxy) --- internet --- soft client FS2------ | fs_path ignored when trying originate and cal just failing freeswitch(a)casip00.networklab.prod> sofia status profile internal-proxy ================================================================================================= Name internal-proxy Domain Name N/A Auto-NAT false DBName sofia_reg_internal-proxy Pres Hosts $${domain},10.18.130.24 Dialplan XML Context default Challenge Realm auto_to RTP-IP 10.18.130.24 SIP-IP 10.18.130.24 URL sip:mod_sofia@10.18.130.24:5160 BIND-URL sip:mod_sofia@10.18.130.24:5160;transport=udp HOLD-MUSIC N/A OUTBOUND-PROXY N/A CODECS IN CODECS OUT PCMU,G729 TEL-EVENT 101 DTMF-MODE info CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA true LATE-NEG true PROXY-MEDIA true ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 4 FAILED-CALLS-OUT 4 REGISTRATIONS 1 freeswitch(a)casip00.networklab.prod> sofia_contact 4310(a)sip.company.tld sofia/internal-proxy/sip:4310@client_public_ip:40368;rinstance=8344944f63ac4573;transport=tcp;fs_path=sip%3A10.18.130.27%3Blr%3Breceived%3Dsip%3Aclient_public_ip%3A40368%3Btransport%3Dtcp freeswitch(a)casip00.networklab.prod> originate sofia/internal-proxy/sip:4310@client_public_ip:40368;rinstance=8344944f63ac4573;transport=tcp;fs_path=sip%3A10.18.130.27%3Blr%3Breceived%3Dsip%3Aclient_public_ip%3A40368%3Btransport%3Dtcp inline -ERR NORMAL_TEMPORARY_FAILURE 2017-04-20 16:33:48.568255 [DEBUG] switch_ivr_originate.c:2138 Parsing global variables 2017-04-20 16:33:48.568255 [NOTICE] switch_channel.c:1104 New Channel sofia/internal-proxy/4310@client_public_ip:40368 [d3f8d570-a829-4a89-af71-382dc2b37af2] 2017-04-20 16:33:48.568255 [DEBUG] mod_sofia.c:4753 (sofia/internal-proxy/4310@client_public_ip:40368) State Change CS_NEW -> CS_INIT 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:584 (sofia/internal-proxy/4310@client_public_ip:40368) Running State Change CS_INIT 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:627 (sofia/internal-proxy/4310@client_public_ip:40368) State INIT 2017-04-20 16:33:48.568255 [DEBUG] mod_sofia.c:90 sofia/internal-proxy/4310@client_public_ip:40368 SOFIA INIT 2017-04-20 16:33:48.568255 [DEBUG] sofia_glue.c:1253 sip:10.18.130.27;lr;received=sip:client_public_ip:40368;transport=tcp Setting proxy route to sofia/internal-proxy/4310@client_public_ip:40368 2017-04-20 16:33:48.568255 [DEBUG] sofia_glue.c:1282 sofia/internal-proxy/4310@client_public_ip:40368 sending invite version: 1.6.12 64bit Local SDP: v=0 o=FreeSWITCH 1492703274 1492703275 IN IP4 10.18.130.24 s=FreeSWITCH c=IN IP4 10.18.130.24 t=0 0 m=audio 17154 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:13 CN/8000 a=ptime:20 a=sendrecv 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:40 sofia/internal-proxy/4310@client_public_ip:40368 Standard INIT 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:48 (sofia/internal-proxy/4310@client_public_ip:40368) State Change CS_INIT -> CS_ROUTING 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:627 (sofia/internal-proxy/4310@client_public_ip:40368) State INIT going to sleep 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:584 (sofia/internal-proxy/4310@client_public_ip:40368) Running State Change CS_ROUTING 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:643 (sofia/internal-proxy/4310@client_public_ip:40368) State ROUTING 2017-04-20 16:33:48.568255 [DEBUG] mod_sofia.c:143 sofia/internal-proxy/4310@client_public_ip:40368 SOFIA ROUTING 2017-04-20 16:33:48.568255 [DEBUG] switch_ivr_originate.c:67 (sofia/internal-proxy/4310@client_public_ip:40368) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:643 (sofia/internal-proxy/4310@client_public_ip:40368) State ROUTING going to sleep 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:584 (sofia/internal-proxy/4310@client_public_ip:40368) Running State Change CS_CONSUME_MEDIA 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:662 (sofia/internal-proxy/4310@client_public_ip:40368) State CONSUME_MEDIA 2017-04-20 16:33:48.568255 [DEBUG] switch_core_state_machine.c:662 (sofia/internal-proxy/4310@client_public_ip:40368) State CONSUME_MEDIA going to sleep 2017-04-20 16:33:48.568255 [DEBUG] sofia.c:6965 Channel sofia/internal-proxy/4310@client_public_ip:40368 entering state [calling][0] 2017-04-20 16:33:48.568255 [DEBUG] sofia.c:6965 Channel sofia/internal-proxy/4310@client_public_ip:40368 entering state [terminated][503] 2017-04-20 16:33:48.568255 [NOTICE] sofia.c:8078 Hangup sofia/internal-proxy/4310@client_public_ip:40368 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] Any help thank you. Slava

1 0

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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