August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

question on file string in playback
by Dave Horton 09 Aug '16

09 Aug '16

When using the playback application (or play_and_get_digits etc) do I need to specify the full path to the file under the base sounds directory, including the sample rate (e.g. 8000). It’s a bit cumbersome to have to put the sample rate into the file string since that is dynamically determined at call set up, and I _thought_ that freeswitch had the ability to automatically factor that in when locating the file. However, I can’t find examples that do that — am I wrong, and must I somehow concoct a file string that will be different per call if different codecs are chosen?

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Empty Database of freeswitch and fusionpbx
by uzair iltifat 09 Aug '16

09 Aug '16

Hi, I have success fully installed the freeswitch and registered 2 SIP user, now i have installed the fusion pbx and it has given me the successfully installed message. Now when i try check its databases from mysql console they are empty, why is it so. Even it didnt have the v_default_setting , etc tables in it. waiting for the response. http://uzair.ahmed.netcv.com/http://uzair.ahmed.netcv.com/ Uzair Ahmed [image: http://]about.me/uzairIltifat <http://about.me/uzairIltifat?promo=email_sig> <http://about.me/uzairIltifat?promo=email_sig> http://uzair.ahmed.netcv.com/

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Sounds and moh format.
by Mimiko 09 Aug '16

09 Aug '16

Hello. In asterisk sounds and moh a delivered as files encoded to different codec like ulaw, alaw, gsm and so on. FS deliver sounds in .wav file with different discretisation frequency like 8k, 16k, 32k, 48k. If asterisk has the leg in one format, it will fetch the corresponding file to other party not doing any decode/encode of the sound. FS needs to encode every time into corresponding codec of the client. Which approach is better from performance view? -- Mimiko desu.

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No tone detection in AWS
by Gonzalo Gasca Meza 09 Aug '16

09 Aug '16

I'm troubleshooting Tone detection in Freeswitch installed in AWS (Debian) I have 2 servers one works the other doesn't: *FreeSWITCH (Version 1.7.0 -928-db61f4d 64bit)* DigitalOcean WORKS http://pastebin.com/bPx4xGCP *FreeSWITCH (Version 1.6.9 -16-d574870 64bit)* is ready AWS FAILS http://pastebin.com/ktnFYJ05 *Troubleshooting* Seems to be that Freeswitch in AWS EC2 because of NAT is not able to get the RTP from remote side. When I do a packet capture I don't get any RTP from remote end and in initial SIP SDP from Freeswitch I do see local IP address from AWS even though ext-X params are configured. 1. One instance is installed in AWS, the other one in DigitalOcean, 2. I'm originating my calls using ESL and my Gateway: *originate {ignore_early_media=true}sofia/gateway/asterisk/+14171111111 handle_calls* 3. I already configured* ext-rtp-ip and ext-sip-ip* fields in external.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> https://www.twilio.com/resources/images/docs/Twilio-Freeswitch.pdf Any ideas? *Logs* sofia status gateway asterisk ================================================================================================= Name asterisk Profile external Scheme Digest Realm 192.241.203.21:5090 Username FreeSWITCH Password no >From <sip:FreeSWITCH@192.241.203.21:5090> Contact <sip:gw+asterisk@52.67.78.120:5090;transport=udp;gw=asterisk> Exten FreeSWITCH To sip:FreeSWITCH@192.241.203.21:5090 Proxy sip:192.241.203.21:5090 Context public Expires 3600 Freq 3600 Ping 0 PingFreq 0 PingTime 0.00 PingState 0/0/0 State NOREG Status UP Uptime 290s CallsIN 0 CallsOUT 2 FailedCallsIN 0 FailedCallsOUT 0 sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 172.31.3.166 Ext-RTP-IP 52.67.78.120 SIP-IP 172.31.3.166 Ext-SIP-IP 52.67.78.120 URL sip:mod_sofia@52.67.78.120:5090 BIND-URL sip:mod_sofia@52.67.78.120:5090 ;maddr=172.31.3.166;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,VP8 CODECS OUT OPUS,G722,PCMU,PCMA,VP8 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT true CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 1 REGISTRATIONS 0 Mod_spandsp >module_exists mod_spandsp true >show applications spandsp_detect_tdd,Detect TDD data,,mod_spandsp spandsp_inject_tdd,Send TDD data,,mod_spandsp spandsp_send_tdd,Send TDD data,,mod_spandsp spandsp_start_dtmf,Detect dtmf,,mod_spandsp spandsp_start_fax_detect,start fax detect,<app>[ <arg>][ <timeout>][ <tone_type>],mod_spandsp spandsp_start_tone_detect,Start background tone detection with cadence,<name>,mod_spandsp spandsp_stop_detect_tdd,stop sending tdd,,mod_spandsp spandsp_stop_dtmf,stop inband dtmf,,mod_spandsp spandsp_stop_fax_detect,stop fax detect,,mod_spandsp spandsp_stop_inject_tdd,stop sending tdd,,mod_spandsp spandsp_stop_tone_detect,Stop background tone detection with cadence,,mod_spandsp Thanks

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trying renegotiate video mid-call (renegotiate codec on reinvite not working)
by Dave Horton 08 Aug '16

08 Aug '16

I am establishing a bridged call with audio only: Command Execute bridge({sip_renegotiate_codec_on_reinvite=true}sofia/gateway/${asterisk}/110) and then receiving a re-INVITE on the A leg that includes video. Freeswitch sends 200 OK on the A leg but never sends a re-INVITE on the B leg. I am running version FreeSWITCH Version 1.6.8+git~20160505T153832Z~99de0ad502~64bit (git 99de0ad 2016-05-05 15:38:32Z 64bit) I have tried with late negotiation both true and false, neither works (in terms of getting FS to generate a re-INVITE on the B leg). I can provide full logs, etc and open a JIRA if this is a bug; just wanted to check if there was any possible explanation / collision of settings etc. that could explain it

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Replace One Session with Another
by Colin Morelli 08 Aug '16

08 Aug '16

Hey all, I asked this question a little while back and didn't get a response but I worked around the issue for a while so I had never followed up. Now I'm facing some other issues and would like to see if there's an answer to this out there (or if it needs to be built). I'm looking to get behavior like the "pickup" channel, but at any point during a call. Say I have channel A on a call with channel B. Channel C calls, and I want it to take the place of A such that C is now talking to B. In this particular case, intercept could work, but I have two issues with that: 1. I would like to copy all channel vars from A to C 2. I would like to be able to do the same thing *at any point* during a call. Even if A is not currently bridged to another endpoint. As I understand, intercept connects you with an existing call, it doesn't replace anything. So logically it would only work when there's another endpoint to bridge you to. Essentially, I want to swap sessions. Have everything that A is doing replaced with C, and drop A. This actually appears to be what the pickup channel does (I have noticed all channel vars are copied), but pickup only works during bridging. Is this possible to do? Best, Colin

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PosgresSQL in core
by Murugan Pandian 08 Aug '16

08 Aug '16

Hi, I have two FS node with single PosgresSQL,I can able to register extension in table, but when i make call between two extensions it always fails. 1) 1000(FS1)--->1001(FS1) Working 2)1000(FS1)--->1001(FS2) Not working Regard's Murugan Pandian

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mod_flite not loading on Ubuntu 14.04 at least
by Caragea Silviu 08 Aug '16

08 Aug '16

Hello, I successfully compiled and started FS 1.6 (last commit from that branch) but seems there is a problem with mod_flite which is not startign because of : ib/freeswitch/mod/mod_flite.so **/usr/lib/libflite_cmu_us_slt.so.1: undefined symbol: cmu_lex_init** I was looking to FS 1.4 which works fine and I see there the flite is compiled from sources. I have a feeling that libflite-dev is broken on ubuntu . anyone has the same problem ? There is any workaround to this problem ? Silviu

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Ad-hoc conferences with RFC 4579
by Lassi Romanainen 08 Aug '16

08 Aug '16

Hi, I've been trying to find an example how to configure (or use, if no extra configuration is needed) Freeswitch for creating ad-hoc conferences described in RFC 4579. Has anyone bumped into one recently or maybe used Freeswitch in such manner? I'm trying to create conference rooms as described in this chapter: https://tools.ietf.org/html/rfc4579#page-16 (5.4. INVITE: Creating a Conference Using Ad-Hoc SIP Methods) Thnks beforehand for any pointers! Best regards, -Lassi Romanainen

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First Inbound DID
by Don Hawkins 07 Aug '16

07 Aug '16

I feel like the directions on setting up a DID for inbound couldn't be clearer but yet I've been unable to get it working. * 1. SIP Invite: ------------------------------------------------------------------------ ACK sip:19722364002@mymyserver.com SIP/2.0 Via: SIP/2.0/UDP 162.217.100.10;branch=z9hG4bK1bf3.34c008d0cd01ff7b9bec681d865ae587.0 Max-Forwards: 69 To: <sip:19722364002@myserver.com>;tag=cZryjBv7B58aB From: <sip:+12144630000@162.217.100.10>;tag=as74def9f2 Call-ID: 0a7173f94045e4985f82b31525b0d93f(a)192.170.152.170 CSeq: 102 ACK Content-Length: 0 ------------------------------------------------------------------------ * 2. In public.xml AND in /public/inboundtest.xml <include> <extension name="public_did"> <condition field="destination_number" expression="^(19722364002)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> </include> also tried... <condition field="${sip_to_user}" expression="^(19722364002)$"> * 3. Either way when I call the DID: 2016-08-06 23:05:04.089527 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/+14693380000(a)162.217.100.10 [ca1b456f-e9fc-4c99-a72d-d594835feb10] 2016-08-06 23:05:04.089527 [DEBUG] switch_core_state_machine.c:543 (sofia/internal/+14693380000(a)162.217.100.10) Running State Change CS_NEW 2016-08-06 23:05:04.089527 [DEBUG] sofia.c:9456 sofia/internal/+ 14693380000(a)162.217.100.10 receiving invite from 162.217.100.10:5060 version: 1.6.9 -16-d574870 64bit 2016-08-06 23:05:04.089527 [DEBUG] sofia.c:9623 IP 162.217.100.10 Rejected by acl "domains". Falling back to Digest auth. NOTE: 4693380000 is the number I am calling from, not the DID number. The DID number is 9722364002. Any help is appreciated, thanks!

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