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ESL and Event Socket for .NET
by Tony Bourdeaux 24 Feb '15

24 Feb '15

Could anyone help guide me to the correct Event Socket Libraries for Freeswitch with .NET (C#)? I see several projects that are incomplete and Visual Studio project in the source under /libs/esl/managed. But I cannot get this to compile. Is there any documentation on ESL for .NET that someone could share ? Thank You. -- Tony Bourdeaux

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Active channels on gateways
by Zoltán Szabó 24 Feb '15

24 Feb '15

Is there any way to tell how much channels are on a gateway? If I run show channels or show calls, these return the channels but the entries do not contain the gateway name, so I can not count by gw. Thanks, Zoltan

2 1

Freeswitch + XML_CURL
by Antony Aguirre Morales 24 Feb '15

24 Feb '15

Hi, Now, I have a freeswitch server and xml_curl are working correctly. I would like add a second freeswitch server but this second freeswitch (FS) should make the queries to the DB which are linked with the first freeswitch. How can I configure both FS for making the queries to the same DB or I have to configure DB for multiple domanins? FS1 -> 192.168.15.1 FS2 -> 192.168.15.2 BD xml_curl -> 192.168.15.3 extension en xml_curl : <document type="freeswitch/xml"> <section name="directory" description="FreeSWITCH Directory"> <domain name="192.168.15.1"> <params> <param name="default_gateway" value="errors"/> <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/> </params> <variables/> <groups> <group name="default"> <users> <user id="12345678ip" cacheable="0" mailbox="12345678ip"> <params> <param name="password" value="6389"/> <param name="vm-password" value="12345678ip"/> <param name="register" value="true"/> </params> <variables> <variable name="accountcode" value="12345678ip"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension12345678ip"/> <variable name="effective_caller_id_number" value="12345678ip"/> </variables> <gateways> <gateway name="12345678ip"> <param name="username" value="12345678ip"/> <param name="from-user" value="12345678ip"/> <param name="from-domain" value="empresa1"/> <param name="password" value="6389"/> <param name="extension-in-contact" value="true"/> <param name="proxy" value="0.0.0.0"/> <param name="register-proxy" value="0.0.0.0"/> <param name="register-transport" value="udp"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways> </user> </users> </group> </groups> </domain> </section>     </document>

1 0

Dealing with rtp_audio_in and out variables
by Florent Krieg 24 Feb '15

24 Feb '15

Hi there, Does anybody know if we can access rtp_audio_* variables from the dialplan (after a bridge is terminated for instance)? Or are they only available in the CDR (through mod_xml_cdr)? I'm talking about: [...] <rtp_audio_in_raw_bytes>169076</rtp_audio_in_raw_bytes> <rtp_audio_in_media_bytes>168732</rtp_audio_in_media_bytes> <rtp_audio_in_packet_count>983</rtp_audio_in_packet_count> <rtp_audio_in_media_packet_count>981</rtp_audio_in_media_packet_count> <rtp_audio_in_skip_packet_count>72</rtp_audio_in_skip_packet_count> <rtp_audio_in_jitter_packet_count>0</rtp_audio_in_jitter_packet_count> <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count> <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count> <rtp_audio_in_flush_packet_count>2</rtp_audio_in_flush_packet_count> <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size> [...] If it's possible to read them, could you tell me how? Thanks in advance Florent

2 2

"Only 1 of the same file per channel please" Error message
by Ali Pey 24 Feb '15

24 Feb '15

Hello, Can someone tell me why I get this error message time to time? I have no way to duplicate it in the lab but I see these errors time to time in production and I'm not sure what causes them. Thanks, Ali Pey

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WebRTC connection delay (Auto Changing stun/rtp/dtls port from)
by Adam Ben-Ayoun 24 Feb '15

24 Feb '15

Hi guys, We are using Freeswitch as an MCU for WebRTC (only for audio conferences), most of the times connecting from our WebRTC client is pretty quick, however, on some WiFi's (probably with a certain NAT/topology), the audio connection can take up to 10sec to initialize when using STUN and TURN, we also see there are differences on the Freeswitch logs when it happens: 2015-02-22 21:06:25.920612 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 67.180.21.132:59854 to 166.170.42.141:40334 2015-02-22 21:06:25.960618 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 166.170.42.141:40334 to 67.180.21.132:59854 2015-02-22 21:06:26.080616 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 67.180.21.132:59854 to 54.187.44.182:53379 2015-02-22 21:06:26.680617 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 54.187.44.182:53379 to 166.170.42.141:40334 2015-02-22 21:06:27.040614 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 166.170.42.141:40334 to 67.180.21.132:59854 2015-02-22 21:06:27.880612 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 67.180.21.132:59854 to 166.170.42.141:40334 2015-02-22 21:06:28.020610 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 166.170.42.141:40334 to 67.180.21.132:59854 2015-02-22 21:06:28.600614 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 67.180.21.132:59854 to 166.170.42.141:40334 2015-02-22 21:06:28.980603 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 166.170.42.141:40334 to 67.180.21.132:59854 2015-02-22 21:06:29.560612 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 67.180.21.132:59854 to 166.170.42.141:40334 2015-02-22 21:06:29.940602 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 166.170.42.141:40334 to 67.180.21.132:59854 2015-02-22 21:06:30.540561 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 67.180.21.132:59854 to 166.170.42.141:40334 2015-02-22 21:06:30.940606 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 166.170.42.141:40334 to 67.180.21.132:59854 2015-02-22 21:06:31.500595 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 67.180.21.132:59854 to 166.170.42.141:40334 2015-02-22 21:06:31.900613 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 166.170.42.141:40334 to 67.180.21.132:59854 2015-02-22 21:06:32.460602 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 67.180.21.132:59854 to 166.170.42.141:40334 2015-02-22 21:06:32.860614 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 166.170.42.141:40334 to 67.180.21.132:59854 2015-02-22 21:06:33.040612 [INFO] switch_rtp.c:2926 Changing audio DTLS state from HANDSHAKE to SETUP 2015-02-22 21:06:33.040612 [INFO] switch_rtp.c:2834 audio Fingerprint Verified. 2015-02-22 21:06:33.040612 [INFO] switch_rtp.c:3376 Activating Audio Secure RTP SEND 2015-02-22 21:06:33.040612 [INFO] switch_rtp.c:3354 Activating Audio Secure RTP RECV 2015-02-22 21:06:33.040612 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-02-22 21:06:33.040612 [INFO] switch_rtp.c:2874 Changing audio DTLS state from SETUP to READY 2015-02-22 21:06:33.040612 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-02-22 21:06:33.100606 [DEBUG] switch_rtp.c:1954 rtcp_stats_init: ssrc[-1706748029] base_seq[28893] X.X.X.X is the IP of our TURN server, now, it seems like although we have a good reflexive candidate, it takes some time for Freeswitch to pick it up and it actually tries to use the relay (TURN)... Any idea what is happening and what we can do to solve that? Here are the ICE candidates from the client SDP: a=candidate:1780409101 1 udp 2122063615 192.168.15.3 56494 typ host generation 0 a=candidate:1780409101 2 udp 2122063615 192.168.15.3 56494 typ host generation 0 a=candidate:4043381328 1 udp 2122129151 10.159.103.156 51485 typ host generation 0 a=candidate:4043381328 2 udp 2122129151 10.159.103.156 51485 typ host generation 0 a=candidate:2618202841 1 udp 1685855999 67.180.21.132 56494 typ srflx raddr 192.168.15.3 rport 56494 generation 0 a=candidate:2618202841 2 udp 1685855999 67.180.21.132 56494 typ srflx raddr 192.168.15.3 rport 56494 generation 0 a=candidate:614277117 1 tcp 1518083839 192.168.15.3 54312 typ host tcptype passive generation 0 a=candidate:614277117 2 tcp 1518083839 192.168.15.3 54312 typ host tcptype passive generation 0 a=candidate:3213026976 1 tcp 1518149375 10.159.103.156 54313 typ host tcptype passive generation 0 a=candidate:3213026976 2 tcp 1518149375 10.159.103.156 54313 typ host tcptype passive generation 0 a=candidate:3054806047 1 udp 24911615 X.X.X.X 65452 typ relay raddr 67.180.21.132 rport 54310 generation 0 a=candidate:3054806047 2 udp 24911615 X.X.X.X 65452 typ relay raddr 67.180.21.132 rport 54310 generation 0 a=candidate:1032594883 1 udp 1685921535 166.170.42.141 64296 typ srflx raddr 10.159.103.156 rport 51485 generation 0 a=candidate:1032594883 2 udp 1685921535 166.170.42.141 64296 typ srflx raddr 10.159.103.156 rport 51485 generation 0 a=candidate:3054806047 1 udp 24977151 X.X.X.X 49737 typ relay raddr 166.170.42.141 rport 56009 generation 0 a=candidate:3054806047 2 udp 24977151 X.X.X.X 49737 typ relay raddr 166.170.42.141 rport 56009 generation 0 Thanks, Adam

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To and From fields in SIP OPTIONS
by Stanislav Sinyagin 24 Feb '15

24 Feb '15

It appears that in the OPTIONS ping packet, To and From fields contain only the domain name. In sofia.c: if (ping_freq) { if (ping_freq >= 5) { gateway->ping_freq = ping_freq; gateway->ping_max = ping_max; gateway->ping_min = ping_min; gateway->ping_monitoring = ping_monitoring; gateway->ping = switch_epoch_time_now(NULL) + ping_freq; gateway->options_to_uri = switch_core_sprintf(gateway->pool, "<sip:%s>", !zstr(from_domain) ? from_domain : proxy); gateway->options_from_uri = gateway->options_to_uri; } else { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ERROR: invalid ping!\n"); } } Is there a reason why To and From fields don;t contain the register username? This seems to break some other vendor gateways. thanks, stan

4 4

Data modem (V32, V92) with SpanDSP
by François 24 Feb '15

24 Feb '15

Hi all, As seen in the HylaFax wiki entry, mod_spandsp can be used as a soft-modem. https://wiki.freeswitch.org/wiki/HylaFax Can it also be used as a data modem (not just fax modem) with V32, V92 or similar protocols? If not, does anyone know of any data-modem software that could work along with FreeSWITCH? Thanks, François.

5 11

Nibblebill issues on centos 6?
by Chris Martineau 24 Feb '15

24 Feb '15

Hi, Anyone come across any issues using nibblebill on centos 6 to a remote Mysql DB on Centos 5? Doesn't matter what version of Freeswitch I am using, if I deploy it on our existing centos 5 servers to a mysql db deployed also on a centos 5 server it works fine. If I deploy on a centos 6 platform I get strange issues with the odbc interface. ISQL connection works fine but when I plug the details into the dsn in the nibblebill configuration it doesn't work? If I try dsn freeswitch:user:passwd then the module fails to initialise the database and complains that it cannot connect to odbc. The only combination that works is just setting dsn to freeswitch which seems to get by the db handle code but then fails with a 'cannot find table' error. The first configuration works fine on centos 5 with the same software and config and to the same remote db. The only difference I can see is the odbc drivers which are later versions on centos6. Could this be something to do with new passwords formats on mysql? The versions of mysql being used are using the old format but setting the old_passwords entry in my.cnf makes no difference, set or unset. Any ideas? Regards Chris

4 8

Where to go from here?
by T Fred Farmington 24 Feb '15

24 Feb '15

Thank you for your replies I just now copied my FreeSWITCH.log data into the pastebin. Since I had just done a full Re-Install of my FreeSWITCH thereby removing any previous modification attempts, I did not include my dialplan, internal & external xml files. They are now just the after-install defaults. I did that Re-Install out of frustration due to my previous xml file modifications only introducing more problems and solving none. I need to understand: 1. Which xml files to modify and which ones to leave alone 2. How/Where to modify these xml files I sincerely thank you for your assistance. TF

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