March 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_skypopen - prereqs for Jessie?
by Rick Jarvis 08 Mar '16

08 Mar '16

Following this: https://freeswitch.org/confluence/display/FREESWITCH/mod_skypopen <https://freeswitch.org/confluence/display/FREESWITCH/mod_skypopen> Getting errors such as: E: Package 'gcc-4.7-base:i386' has no installation candidate E: Unable to locate package libavcodec53 E: Unable to locate package libavutil51 Can anyone tell me how to install the prereqs on Jessie? Or am I missing a repo or something? deb http://ftp.uk.debian.org/debian/ jessie main contrib non-free deb-src http://ftp.uk.debian.org/debian/ jessie main contrib non-free deb http://security.debian.org/ jessie/updates main contrib deb-src http://security.debian.org/ jessie/updates main contrib deb http://files.freeswitch.org/repo/deb/debian/ jessie main

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BLF of dialing party
by Naveen Khanna 08 Mar '16

08 Mar '16

HI, I have come across a situation where BLF of dialing party is not lit. However, if the same user is receiving the Calls then we get proper BLF. Regards, Naveen Khanna M : +91-9911393060 | Skype : naveen.khanna.bm

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Beginners Question - Bridge vs. Transfer
by William Colls 08 Mar '16

08 Mar '16

Where might I find guidance on the difference between these two and best practice for using each? Thanks for your time. William.

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Stutter on ringtone - incoming calls only
by Karl Schmidt 07 Mar '16

07 Mar '16

When people call in (via callcentric) - they hear a stutter in the ringtone. My understanding is that the ringtone is generated by freeswitch - looks like here: <action application="set" data="transfer_ringback=${us-ring}"/> We are running the Debian release - 1.4.26~37-1~jessie+1 Could be this is a new 'feature'? Never sure. Anyone else seeing this? I'm thinking the ringtone might be using a different CODEX? -- -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB https://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-3089 Teaching people to think of themselves as victims is close to pure evil. kps --------------------------------------------------------------------------------

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Joining several FS servers in different regions
by Oivvio Polite 07 Mar '16

07 Mar '16

I'm working on a Saas service that will connect WebRTC (with SIP or Verto for signaling) clients to one another and to PSTN. I'm aware of other such services, but this is tailored to a specific niche. I also want WebRTC media to flow through FS, rather than P2P, so that I can record it. The primary markets are Europe and North America. To keep latency low I need to have servers on both continents. Later on I might want to add more regions. I'm not too worried about high availability or high loads. The clients can keep track of which FS is closest to them and register with that. But this still leaves me with a couple of questions. 1. Maintaining a single user database ===================================== What's the simplest way of maintaining one joint user database? (User directory in FS parlance). The first thing that comes to mind is to generate the xml dynamically from a central database, and force as `reloadxml` on all servers every time there's an update. The communication between central database and FS servers could be via RabbitMQ. mod_xml_curl could also be a part of this I guess. How does that sound? Is there an other simpler way of achiving my goal that I'm not seeing? 2. Keeping track of where a user i currently registered ======================================================= Let's say I have a client Alice who's registered with a FS server in North America and a client Bob who has registered wit a FS server in Europe. Now Alice tries to call Bob. How does the North America server know that Bob is currently registered with the Europe server and that the call should be routed through there? For this second problem I don't even have a tentative solution so I'm curious to hear any ideas. 3. SIP Trunks and geography =========================== If Alice is in North America and wants to call a PSTN endpoint in North America I should obviously be using a SIP trunk in North America, since going over the Atlantic twice will add a lot of latency. But what about when Alice want to call a PSTN endpoint in Europe? Is it better to use a SIP endpoint in Europe or in North America? Thanks in advance Oivvio

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SSL certificate and webrtc
by Gregor Nanger 06 Mar '16

06 Mar '16

Hi! I bought wildcard certificate for using it with freeswitch. Now I have hard time to install it into Freeswitch and would appreciate any help. I have private key (generated at CSR request), certificate issued by authority and root certificate from authority. Now I do not know where to enter certficate. According to documentation I assume that this is correct: Cert, Key and Chain(s) are all contained in a single file in this order: -----BEGIN CERTIFICATE----- <cert> -----END CERTIFICATE----- -----BEGIN RSA PRIVATE KEY----- <key> -----END RSA PRIVATE KEY----- -----BEGIN CERTIFICATE----- <chain> -----END CERTIFICATE----- What is puzzling me is if this is correct: <cert> = root certificate from authority <key> = private key <chain> = public certificate issued by authority Would this work?

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Codec negotiation. Totally confused
by Igor Olhovskiy 06 Mar '16

06 Mar '16

Hi! I’m getting really strange things, or I’m just missed something. My phone is dials to freeswitch with this this line in log 2016-03-06 08:31:56.681036 [DEBUG] sofia.c:6770 Remote SDP: v=0 o=root 1697549695 1697549695 IN IP4 <EXTERNAL IP HERE> s=call c=IN IP4 <EXTERNAL IP HERE> t=0 0 m=audio 26894 RTP/AVP 9 0 8 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=nortpproxy:yes 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/10(a)consertis.securenetvox.net G722/8000 20 ms 160 samples 64000 bits 1 channels And when switches to external profile, I see 2016-03-06 08:31:56.721010 [DEBUG] sofia_glue.c:1257 sofia/external/00972543279009 sending invite version: 1.6.6 git d2d0b32 2016-01-11 20:16:12Z 64bit Local SDP: v=0 o=FreeSWITCH 1457219712 1457219713 IN IP4 10.0.20.71 s=FreeSWITCH c=IN IP4 10.0.20.71 t=0 0 m=audio 29804 RTP/AVP 9 101 13 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=sendrecv The question is - why only G722 left? Across dialplan there is no things like inherit_codec, bypass media or other codec-related stuff Profiles external ext-sip-ip [10.0.20.71] rtp-timeout-sec [300] rtp-hold-timeout-sec [1800] tls [false] tls-only [false] tls-bind-params [transport=tls] tls-sip-port [5081] tls-cert-dir [/usr/local/freeswitch/conf/ssl] tls-passphrase [] tls-verify-date [true] tls-verify-depth [2] tls-verify-in-subjects [] tls-version [tlsv1] tls-verify-policy [all] odbc-dsn [pgsql://hostaddr=127.0.0.1 port=5432 dbname=freeswitch user=fusionpbx password=pass options='' application_name='freeswitch'] track-calls [true] inbound-codec-negotiation [greedy] debug [0] user-agent-string [FreeSWITCH] sip-trace [no] sip-capture [no] rfc2833-pt [101] sip-port [5080] dialplan [XML] context [public] dtmf-duration [2000] inbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] outbound-codec-prefs [PCMU,PCMA,GSM] hold-music [local_stream://default] zrtp-passthru [true] rtp-timer-name [soft] local-network-acl [localnet.auto] manage-presence [false] nonce-ttl [60] auth-calls [false] rtp-ip [10.0.20.71] sip-ip [10.0.20.71] ext-rtp-ip [10.0.20.71] internal tls-cert-dir [/usr/local/freeswitch/conf/ssl] tls-passphrase [] tls-verify-date [true] tls-verify-depth [2] tls-verify-in-subjects [] tls-version [tlsv1] nonce-ttl [60] auth-calls [true] inbound-reg-force-matching-username [true] auth-all-packets [false] ext-rtp-ip [10.0.20.71] ext-sip-ip [10.0.20.71] rtp-timeout-sec [300] rtp-hold-timeout-sec [1800] tls-verify-policy [all] multiple-registrations [contact] enable-timer [false] dbname [share_presence] send-presence-on-register [true] inbound-codec-negotiation [greedy] NDLB-force-rport [safe] challenge-realm [auto_to] outbound-proxy [10.0.20.70] track-calls [true] odbc-dsn [pgsql://hostaddr=127.0.0.1 port=5432 dbname=freeswitch user=fusionpbx password=btgJek49 options='' application_name='freeswitch'] nat-options-ping [true] liberal-dtmf [true] all-reg-options-ping [true] force-publish-expires [true] unregister-on-options-fail [true] user-agent-string [FreeSWITCH] debug [0] sip-trace [no] sip-capture [no] watchdog-enabled [no] watchdog-step-timeout [30000] watchdog-event-timeout [30000] log-auth-failures [true] forward-unsolicited-mwi-notify [false] context [public] rfc2833-pt [101] sip-port [5060] dialplan [XML] dtmf-duration [2000] inbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] outbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] rtp-timer-name [soft] rtp-ip [10.0.20.71] sip-ip [10.0.20.71] hold-music [local_stream://default] apply-nat-acl [nat.auto] aggressive-nat-detection [true] apply-inbound-acl [domains] local-network-acl [localnet.auto] record-path [/usr/local/freeswitch/recordings] record-template [${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}] manage-presence [true] presence-probe-on-register [true] manage-shared-appearance [true] tls [false] tls-only [false] tls-bind-params [transport=tls] tls-sip-port [5061] Tried with indbound-late-negotiation=false, also not helps… Can you please, point, what is missing? Thanks -- Best regards, Igor

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Organizations that use FreeSwitch with over 5, 000 extensions
by Yehavi Bourvine 06 Mar '16

06 Mar '16

Hi, We are starting here again the debate of open cource vs. commercial software. Our current PBX holds more than 8,000 extensions. Is there an organization that uses Freeswitch having a few thousands of extensions? I need examples... Thanks! __Yehavi:

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mod_local_stream strange behavior
by Igor Olhovskiy 04 Mar '16

04 Mar '16

Hello, Having a strange issue with a MoH and ringback tones defined with local_stream. Idea is I’m having 1 wav file for MoH. And when I’m playing it, it starts not from beginning, but nearly 3-7 seconds before file end and than - starts over. Can’t get why it not just playing file from start. Yes, I know there is a workaround to use file_stream, but may be there is way to make local_stream with a same behavior? System: Debian 8, VmWare ESXi, FS 1.6.6 (FusionPBX) <directory name=«Domain_name/TestingMoH/8000» path=«$${sounds_dir}/Domain_name/TestingMoH/8000»> <param name="rate" value="8000"/> <param name="shuffle" value="true"/> <param name="channels" value="1"/> <param name="interval" value="20"/> <param name="timer-name" value="soft"/> </directory> Or I’m missing some parameter? -- Best regards, Igor

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freeswwitch database
by jacek burghardt 03 Mar '16

03 Mar '16

How do I use freeswitch with fusionpbx and mysql ? I get this error switch_odbc.c:368 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified Freeswitch log show cant logon to datbase with user name and password.

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