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unable to record to .ogg files
by Vincent Xia 20 Jun '13

20 Jun '13

recording to .wav is ok but to .ogg, FS reports: 2013-06-19 20:24:34.993063 [ERR] switch_core_file.c:150 Invalid file format [ogg] for [D:/recordings/1006_11500_20130619202434.ogg]! 2013-06-19 20:24:34.993063 [ERR] switch_ivr_async.c:2004 Error opening D:/recordings/1006_11500_20130619202434.ogg is there any additional mod required for .ogg, just like mod_shout is needed for .mp3?

3 6

Passing custom SIP header through originate command
by Limit 20 Jun '13

20 Jun '13

Hi! I'am trying to create a fake call to extension passing some additional information in custom SIP header "User-to-User". Command syntax: originate {sip_h_X-User-to-User=48656c6c6f;encoding=hex}sofia/internal/1002(a)10.10.104.125 &playback('test.wav') XML test Call originated but according to sofia siptrace my custom header doesn't transmit to 1002 endpoint. SIP trace looks as follows: send 1298 bytes to udp/[10.10.104.125]:5060 at 08:12:48.524061: ------------------------------------------------------------------------ INVITE sip:1002@10.10.104.125 SIP/2.0 Via: SIP/2.0/UDP 10.10.104.125;rport;branch=z9hG4bKU0j6a2D3Zg02g Max-Forwards: 70 From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:mod_sofia@10.10.104.125:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 373 X-User-to-User: 48656c6c6f;encoding=hex X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@10.10.104.125>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1371708570 1371708571 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=audio 7398 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 7198 RTP/AVP 98 99 34 31 100 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=rtpmap:100 H263-2000/90000 ------------------------------------------------------------------------ recv 1298 bytes from udp/[10.10.104.125]:5060 at 08:12:48.525061: ------------------------------------------------------------------------ INVITE sip:1002@10.10.104.125 SIP/2.0 Via: SIP/2.0/UDP 10.10.104.125;rport;branch=z9hG4bKU0j6a2D3Zg02g Max-Forwards: 70 From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:mod_sofia@10.10.104.125:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 373 X-User-to-User: 48656c6c6f;encoding=hex X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@10.10.104.125>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1371708570 1371708571 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=audio 7398 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 7198 RTP/AVP 98 99 34 31 100 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=rtpmap:100 H263-2000/90000 ------------------------------------------------------------------------ send 319 bytes to udp/[10.10.104.125]:5060 at 08:12:48.525061: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ recv 319 bytes from udp/[10.10.104.125]:5060 at 08:12:48.525061: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ send 1352 bytes to tcp/[10.10.104.125]:30101 at 08:12:48.551064: ------------------------------------------------------------------------ INVITE sip:1002@10.10.104.125:30101;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TCP 10.10.104.125;branch=z9hG4bKv9BZcXy6vSpNc Max-Forwards: 69 From: "" <sip:0000000000@10.10.104.125>;tag=UvKaX51QD0XSp To: <sip:1002@10.10.104.125:30101;transport=TCP;ob> Call-ID: 09412ee0-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:mod_sofia@10.10.104.125:5060;transport=tcp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 373 X-User-to-User: 48656c6c6f;encoding=hex X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@10.10.104.125>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1371708570 1371708571 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=audio 7398 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 7198 RTP/AVP 98 99 34 31 100 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=rtpmap:100 H263-2000/90000 ------------------------------------------------------------------------ recv 284 bytes from tcp/[10.10.104.125]:30101 at 08:12:48.558064: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.10.104.125;received=10.10.104.125;branch=z9hG4bKv9BZcXy6vSpNc Call-ID: 09412ee0-5424-1231-178e-171268c989f7 From: <sip:0000000000@10.10.104.125>;tag=UvKaX51QD0XSp To: <sip:1002@10.10.104.125;ob> CSeq: 45521920 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 479 bytes from tcp/[10.10.104.125]:30101 at 08:12:48.577066: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/TCP 10.10.104.125;received=10.10.104.125;branch=z9hG4bKv9BZcXy6vSpNc Call-ID: 09412ee0-5424-1231-178e-171268c989f7 From: <sip:0000000000@10.10.104.125>;tag=UvKaX51QD0XSp To: <sip:1002@10.10.104.125;ob>;tag=d1fb24be0fba4269bbf3fce36d8bb485 CSeq: 45521920 INVITE Contact: "1002" <sip:1002@10.10.104.125:30101;transport=TCP;ob> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 930 bytes to udp/[10.10.104.125]:5060 at 08:12:48.580067: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125>;tag=tKtHUaHmgQ76a Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:1002@10.10.104.125:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Length: 0 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1002@10.10.104.125 X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:1002@10.10.104.125>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ recv 930 bytes from udp/[10.10.104.125]:5060 at 08:12:48.580067: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125>;tag=tKtHUaHmgQ76a Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:1002@10.10.104.125:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Length: 0 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1002@10.10.104.125 X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:1002@10.10.104.125>;party=calling;privacy=off;screen=no Maybe there is some originate command option to translate SIP headers, or some sort of variable to do that? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Passing-custom-SIP-header-thr… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

FS stress test issue
by Vincent Xia 20 Jun '13

20 Jun '13

while i was doing stress test over FS, things look pretty good as FS is able to hold 800 concurrent calls with a call loss rate at around 0.1%, when leveling up to 1000 concurrent calls, the call loss soars to 10-20%, i made some modification to switch.conf like <param name="max-sessions" value="2000"/> <param name="sessions-per-second" value="100"/> but things didn't get any better, the host CPU usage rate is around 80-90%, memory usage: 10%, network load 10%, so i think there should be someway we can make FS handle more concurrent calls, any ideas?

8 10

0.0.0.0 reinvite in SDP
by Miha 20 Jun '13

20 Jun '13

HI, I am having one issue with music on hold. A calls B (B is at different voip provider that is using netcentrex softswitch), after B do xfer it puts A on hold and send reinvite with 0.0.0.0 in sdp, when C try to pickup this call, just different music on hold appears but there is no media from C. how to deal with this issue? tnx! miha

2 7

Add IP to ACL from the CLI
by Tom Carlson 19 Jun '13

19 Jun '13

I have two freeswitch servers that I am linking together, so that I need to add the IP of each box to the ACL of the other box. However, one of these servers has a dynamic IP from the internet service provider. I have a cron job running that determines the IP address of the dynamic box, every 5 minutes, and if it changes, writes that IP address to the domain name server, so that registration, etc. to the domain name always works, since the domain name always points to the current IP address.. However, Freeswitch does not accept domain names in the ACL. It only accepts IP addresses. That means I need to change acl.conf.xml, every time the ip address changes. I would rather just execute a command using fscli to add the IP address do the ACL. Is this possible?

3 2

Better approach for multi-tenant SIP testing (part 1)
by Cal Leeming [Simplicity Media Ltd] 19 Jun '13

19 Jun '13

Hello all, After nearly a year of active development on building our own voice platform, we've adopted some cool tricks to speed up the workflow and make testing less tedious. Over the next few months I'll be posting more articles on the tricks we have used and the lessons we have learnt. Here is part 1 - a write up about a neat hack we used to virtualize phone testing; http://blog.simplicitymedialtd.co.uk/533/quick-and-easy-approach-for-multi-… For those that don't have time to read, here is a screenshot; http://i.imgur.com/e9kUdxg.jpg This trick, which we've used in development for approx 6 months, has saved countless hours in development overheads, with one-click start/stop functionality and all the necessary modules right there in one place. Of course, this should never be used as a full replacement as nothing beats a proper hardware test to catch those weird edge cases, but it does allow you to do the majority of your testing (95%) from a single screen rather than a desk full of phones. It also makes attaching wireshark a lot easier, without having to use span ports and the alike. Hope this helps someone else! Cal

8 18

SIP Trunk to Avaya CM
by John Doe 19 Jun '13

19 Jun '13

Does anyone have the configuration settings for creating a sip trunk back to an Avaya CM 5.2?

5 6

Does 1.4 has video transcoding?
by Luis Daniel Lucio Quiroz 19 Jun '13

19 Jun '13

wondering if this is possible in 1.4

2 1

Can Freeswitch do audio conditioning/noise filtering on pass-through calls?
by D D 19 Jun '13

19 Jun '13

Hi, I would like to use Freeswitch to filter out low noises from calls that pass through it. How can I do this? Thanks, David

2 1

fs_ivrd and database connectivity
by Ali Pey 19 Jun '13

19 Jun '13

Hello, I have a perl script that is called for every call through fs_ivrd. This script needs to connect to a Database and perform some queries and writing. My problem here is that the script needs to open and close a connection to the database for every call. I am looking for a way that I can connect to the database once and then use the same connection in my perl script to improve performance. Any suggestions/Advice? Thanks, Ali Pey

1 0