February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

disabling DTMF info packets via dtmf_type or dtmf-type
by Robert Stevens 24 Feb '15

24 Feb '15

Hello, During my testing, I have noticed, that if you do not set dtmf_type as a channel variable or dtmf-type as a sofia profile parameter, and if the SDP does not support rfc2833, then FS will failover to INFO packets. This of course is fine on the face of it. According to https://freeswitch.org/confluence/display/FREESWITCH/Variables, you can set dtmf_type to either info, rfc2833, or none as a channel variable. I have noticed however, that if dtmf_type is set to 'none', FS will still failover to INFO packets when the SDP does not contain a rfc2833 media attribute. The dtmf-type sofia profile parameter does work as expected when I set it as a profile parameter, but I would like to set it in the dialplan and the only way to do this, I would presume, is via dtmf_type. However, it currently is not working (I always get the 'no rfc2833 found in sdp switching to info' log entry) and see info packets when the sofia siptrace is enabled). I would think this is a bug. I tried several workarounds to mitigate but was not successful. Specifically I tried setting dtmf-type in the sofia profile params to 'rfc2833' thinking that it would be hard coded to be rfc2833 and not do the switchover to INFO, but alas, it still did the switchover. Does anyone have any insight on this? I think it is a bug and will file a JIRA on it shortly. Best regards, Robert Stevens rstevens(a)robcoit.com

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

24 Feb '15

It appears to me that the vanilla configuration files that come with FreeSWITCH allow you to run in a very limited mode. As I indicated before I can get my in-house softphones to successfully call into the FreeSWITCH and reach one of the default extensions. When I do this I hear the associated FreeSWITCH default voice recordings But when I try to call in via my outside SIP line, I cannot get the connection. I see the call arrive in the FS_CLI and in the log files, but I only get a Busy signal and no connection is made. Editing my sip_profiles - internal & external xml files did not change that behavior. Either: 1. I made the wrong edits in those files 2. I did not edit enough files 3. or I have no idea what I am doing. I would rather understand what should be done than just 'groping in the dark' making edits here and there based on a partial understanding (or lack of understanding) of some web reference. I thought that the FreeSWITCH 1.2. book would help, but nowhere does it say: If you want to receive 'outside' calls, do this...... My current SIP line is configured very simply. * No SIP authentication required * A single IP address on Port 5060 upd * DTMF type is set to RFC2833 * Rtp will be from 208.85.184.41 udp ports 10,000 to 59,000 (although we are doing voice only at this time) I have a number of things that I would like the FreeSWITCH phone switch to do, but I feel that I need to take 'baby steps'. So my First goal is to just successfully receive an 'outside' call to one of the existing default extensions - similar to how the softphones work now. My second 'baby step' goal would be to to use one of my softphones to place a call OUT on the same SIP line. After that I can look at beginning to understand how to configure the various internal extensions to do various tasks. Thanks for your continued patience and support.

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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

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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>

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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

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"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

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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.

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