February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Help xml_curl
by antony aguirre 19 Feb '15

19 Feb '15

Good day! Could support me with a question: Currently have 1 freeswitch running directory XML_CURL smoothly, my question is I want to add 2 freeswitch more and consult the same directory, is there any way for a single extension of the line can have 3 different domains or should I create 3 extensions with 3 domain of each freeswitch. Regards!

1 0

Prioritize bandwidth by uuid
by Andrew V 19 Feb '15

19 Feb '15

Let's say I have a 1000 concurrent calls and 800 of them are on hold with music and 200 of them are live conversations. The music cannot be turned off. Can I prioritize the bandwidth for the 200 live calls (by uuid) over the on hold calls?

3 2

TLS/TCP Contact Issue
by Victor Medina 19 Feb '15

19 Feb '15

HI guys. Even thou my server is configure to use TLS, is responding with non-TLS contact info. 10.0.1.16:2706 <- my.server.name/xx.xx.xx.xx:5061 [my.server.name; xx.xx.xx.xx:5061] 826 SIP/2.0 202 Accepted Via: SIP/2.0/TLS 186.188.115.66:48907 ;branch=z9hG4bKP0yxkkdDNcnyKoyR;rport=48907 From: <sips:1000@my.server.name>;tag=E29270FC8BF4BA92CFBCAE952042F542 To: <sips:1000@my.server.name>;tag=MyMIe4c0faPo Call-ID: 568EFE0E3A65BEABAEC6BD50E56CE667A4B2744B CSeq: 1 SUBSCRIBE Contact: <sip:1000@xx.xx.xx.xx:5060;transport=tcp> <-! HOW DO I FIX THIS? Expires: 600 User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=600 Content-Length: 0 This line in particular Contact: <sip:1000@xx.xx.xx.xx:5060;transport=tcp> <-! Where am i wrong? -- Víctor E. Medina M. Software [image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image: ve] BB #79A8AFA2 /@VMCibersys

3 2

Call decline with 603
by Rahul MathuR 18 Feb '15

18 Feb '15

Hi all, I was doing a POC of WebRTC based audio call to PSTN, routed via kamailio (for protocol translation & proxy) and FS as SIP server. I get 100 trying from FS but after that 603 decline with reason - "Reason: Q.850;cause=16;text="NORMAL_CLEARING"" Also, my phone didn't ring whereas, ITU T Q850 codes say, 16 NORMAL_CLEARING normal call clearing [Q.850] This cause indicates that the call is being cleared because one of the users involved in the call has requested that the call be cleared. Under normal situations, the source of this cause is not the network. Could you please help me in resolving this issue. The messages are as under - 1. INVITE from JsSIP to Kamailio INVITE sip:00919650926333@<FreeSwitch_IP> SIP/2.0 Route: <sip:<Kamailio_IP>:10080;transport=ws;lr> Via: SIP/2.0/TCP amadf8lur89p.invalid;branch=z9hG4bK1158107 Max-Forwards: 69 To: <sip:00919650926333@<FreeSwitch_IP>> From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=3dp7hdgg6j Call-ID: n41t8s01dnclcbodd3il CSeq: 352 INVITE Proxy-Authorization: Digest algorithm=MD5, username="55555", realm="<FreeSwitch_IP>", nonce="d0593b9e-b772-11e4-aeec-2965abc3007e", uri="sip:00919650926333@<FreeSwitch_IP>", response="12e4164f793098007d8bf09f7b50815f", qop=auth, cnonce="l02s3ec50q1g", nc=00000001 Contact: <sip:220bscel@amadf8lur89p.invalid;transport=ws;ob> Content-Type: application/sdp Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: ice,outbound User-Agent: JsSIP 0.6.4 Content-Length: 1769 v=0 o=- 9056480915531460217 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS dcrNk6emj9gTfynUdLaYmsTVbZwnKc9iEwCG m=audio 13228 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 c=IN IP4 59.178.158.4 a=rtcp:13228 IN IP4 59.178.158.4 a=candidate:2437072876 1 udp 2122260223 192.168.1.2 64540 typ host generation 0 a=candidate:2437072876 2 udp 2122260223 192.168.1.2 64540 typ host generation 0 a=candidate:3753982748 1 tcp 1518280447 192.168.1.2 0 typ host tcptype active generation 0 a=candidate:3753982748 2 tcp 1518280447 192.168.1.2 0 typ host tcptype active generation 0 a=candidate:941443129 1 udp 1686052607 59.178.158.4 13228 typ srflx raddr 192.168.1.2 rport 64540 generation 0 a=candidate:941443129 2 udp 1686052607 59.178.158.4 13228 typ srflx raddr 192.168.1.2 rport 64540 generation 0 a=ice-ufrag:9Yhi1W5j+XrHKQKQ a=ice-pwd:fVw9fkXXv1bteXnWh3B/694c a=ice-options:google-ice a=fingerprint:sha-256 C1:96:B0:69:7A:4C:D6:3B:DD:6C:4B:83:BF:F6:45:56:43:95:B4:46:E0:11:BF:AB:2A:42:8D:47:F7:DA:A8:66 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:1011570965 cname:m/wi9qp3sxYVEQLV a=ssrc:1011570965 msid:dcrNk6emj9gTfynUdLaYmsTVbZwnKc9iEwCG 9fd4df2c-5c5d-4dec-a1a6-e181d5da35c4 a=ssrc:1011570965 mslabel:dcrNk6emj9gTfynUdLaYmsTVbZwnKc9iEwCG a=ssrc:1011570965 label:9fd4df2c-5c5d-4dec-a1a6-e181d5da35c4 2. INVITE from Kamailio to FS INVITE sip:00919560509733@<FreeSwitch_IP> SIP/2.0 Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:5090;r2=on;lr=on> Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:10080;transport=ws;r2=on;lr=on> Via: SIP/2.0/UDP <Kamailio_IP>:5090;branch=z9hG4bKb937.6279f7ec5448a6c7867563e9b50fbe39.0 Via: SIP/2.0/TCP 4dcddrn8nrh8.invalid;rport=12016;received=59.178.154.62;branch=z9hG4bK4195437 Max-Forwards: 68 To: <sip:00919560509733@<FreeSwitch_IP>> From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=i5so9j18ej Call-ID: pk2gvvb0qancp1ki8v5t CSeq: 5489 INVITE Proxy-Authorization: Digest algorithm=MD5, username="55555", realm="<FreeSwitch_IP>", nonce="85496252-b74d-11e4-ae0e-2965abc3007e", uri="sip:00919560509733@<FreeSwitch_IP>", response="a821425e292b05c4b9480cb66336cb93", qop=auth, cnonce="h8i58dtuuc0p", nc=00000001 Contact: <sip:g5uvomah@4dcddrn8nrh8.invalid ;transport=ws;ob;alias=59.178.154.62~12016~5> Content-Type: application/sdp Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: ice,outbound User-Agent: JsSIP 0.6.4 Content-Length: 782 v=0 o=- 4750671857217657811 2 IN IP4 <Kamailio_IP> s=- t=0 0 a=msid-semantic: WMS IMbls497PsJs5HOf2sY8de0vWcfW3Vjim2Te m=audio 32838 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 <Kamailio_IP> a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2268746120 cname:Bk0e7lSPEkClAxNy a=ssrc:2268746120 msid:IMbls497PsJs5HOf2sY8de0vWcfW3Vjim2Te 291b0246-4d75-4347-8b2d-e8e42171a36e a=ssrc:2268746120 mslabel:IMbls497PsJs5HOf2sY8de0vWcfW3Vjim2Te a=ssrc:2268746120 label:291b0246-4d75-4347-8b2d-e8e42171a36e a=sendrecv a=rtcp:32839 3. 100 trying from FS to Kamailio SIP/2.0 100 Trying Via: SIP/2.0/UDP <Kamailio_IP>:5090;branch=z9hG4bKb937.6279f7ec5448a6c7867563e9b50fbe39.0 Via: SIP/2.0/TCP 4dcddrn8nrh8.invalid;rport=12016;received=59.178.154.62;branch=z9hG4bK4195437 Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:5090;r2=on;lr=on> Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:10080;transport=ws;r2=on;lr=on> From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=i5so9j18ej To: <sip:00919560509733@<FreeSwitch_IP>> Call-ID: pk2gvvb0qancp1ki8v5t CSeq: 5489 INVITE User-Agent: ASTPP Content-Length: 0 4. 603 from FS to Kamailio SIP/2.0 603 Decline Via: SIP/2.0/UDP <Kamailio_IP>:5090;branch=z9hG4bKb937.6279f7ec5448a6c7867563e9b50fbe39.0 Via: SIP/2.0/TCP 4dcddrn8nrh8.invalid;rport=12016;received=59.178.154.62;branch=z9hG4bK4195437 Max-Forwards: 68 From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=i5so9j18ej To: <sip:00919560509733@<FreeSwitch_IP>>;tag=y4D44m427U2Kj Call-ID: pk2gvvb0qancp1ki8v5t CSeq: 5489 INVITE User-Agent: ASTPP Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "00919560509733" <sip:00919560509733@ <FreeSwitch_IP>>;party=calling;privacy=off;screen=no -- Warm Regds. MathuRahul

3 4

How to use mod_rtmp to play a stream to users.
by Aqs Younas 18 Feb '15

18 Feb '15

Hi, After facing multiple issues with mod_vlc while playing stream to users. I have decided to use mod_rtmp for this purpose. But does not know how i can use it to play a stream, as no example is example on wiki. Correct me if i am wrong, can i use this module for streaming? Thanks in Advance.

3 3

what is realm must be used for proxied calls
by Sergey Safarov 18 Feb '15

18 Feb '15

I has configured gateway with attributes "realm" and "proxy". I thinking that registration request will be sent like "user@realm" in from field. But really is sent user@proxy in from field. It is bug or feature? :o)

2 2

RECOVERY_ON_TIMER_EXPIRE only when options-ping is enabled / FS TLS roundup
by Emrah 17 Feb '15

17 Feb '15

Hi, I am experiencing call drops when using TLS. Calls terminates with a 408 RECOVERY_ON_TIMER_EXPIRE. this seems to only happen when I use options-ping. It never occurred when I disabled it on the profile, however I need it and had to reenable it. I think that it may also be only happening on user accounts that have multiple devices registered. This erratic behavior of FreeSWITCH over TLS is very disconcerting and none of my messages seem to have created any traction so far. The Wiki page about TLS is outdated and contains wrong information. For instance, it says all the RTP and signaling will be encapsulated over the TLS port if TLS is used and this is absolutely incorrect. RTP will remain negotiated over UDP and will have to be separately encrypted. SSLv23 is deprecated now and so TLS v1.1 or v1.2 are the only viable options. Now. I think it would benefit everyone if those among you that have had a successful experience with FS and TLS could share some of their best practices. So far, I have been successful at making and receiving calls over TLS and SRTP. Lately I've been experiencing call drops with the 408 RECOVERY_ON_TIMER_EXPIRE. In addition to that, the call setup process will randomly fail because my clients experience an SSL error while sending the auth packet after receiving the 407. Does this speak to anyone? Cheers

3 2

"record" vs. "record_session"
by Stanislav Sinyagin 17 Feb '15

17 Feb '15

Confluence describes the "record" application, and also has references to "record_session" which return 404 not found. mod/applications/mod_dptools/mod_dptools.c defines both applications. The old wiki page doesn't seem to be available -- maybe someone can restore the content from backup? thanks

3 3

mysql, different tables based on NAS
by Miha 17 Feb '15

17 Feb '15

Hi, it is possible/how to insert based on NAS, accouting requests in different table, database? tnx miha

2 1

dtmf-duration
by Podrigal, Aron 17 Feb '15

17 Feb '15

Hi, I'm setting the dtmf-duration in my profile * <param name="dtmf-duration" value="800"/>* but freeswitch doesn't honor my setting when sending the dtmf 2015-02-17 05:28:36.210973 [DEBUG] switch_ivr_bridge.c:470 Send signal sofia/internal/17183318463(a)198.24.59.141 [BREAK] 2015-02-17 05:28:36.230973 [DEBUG] switch_rtp.c:4346 Send start packet for [#] ts=93280 dur=160/160/2000 seq=39003 lw=93280 2015-02-17 05:28:36.250973 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=320/320/2000 seq=39004 lw=93440 2015-02-17 05:28:36.270973 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=480/480/2000 seq=39005 lw=93600 2015-02-17 05:28:36.290980 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=640/640/2000 seq=39006 lw=93760 2015-02-17 05:28:36.310975 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=800/800/2000 seq=39007 lw=93920 2015-02-17 05:28:36.330975 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=960/960/2000 seq=39008 lw=94080 2015-02-17 05:28:36.350973 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=1120/1120/2000 seq=39009 lw=94240 2015-02-17 05:28:36.373149 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=1280/1280/2000 seq=39010 lw=94400 2015-02-17 05:28:36.390974 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=1440/1440/2000 seq=39011 lw=94560 2015-02-17 05:28:36.410974 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=1600/1600/2000 seq=39012 lw=94720 2015-02-17 05:28:36.430973 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=1760/1760/2000 seq=39013 lw=94880 2015-02-17 05:28:36.450974 [DEBUG] switch_rtp.c:4245 Send middle packet for [#] ts=93280 dur=1920/1920/2000 seq=39014 lw=95040 2015-02-17 05:28:36.470984 [DEBUG] switch_rtp.c:4245 Send end packet for [#] ts=93280 dur=2080/2080/2000 seq=39015 lw=95040 2015-02-17 05:28:36.470984 [DEBUG] switch_rtp.c:4245 Send end packet for [#] ts=93280 dur=2080/2080/2000 seq=39016 lw=95040 2015-02-17 05:28:36.470984 [DEBUG] switch_rtp.c:4245 Send end packet for [#] ts=93280 dur=2080/2080/2000 seq=39017 lw=95040 any help? Thanks

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