March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ESL Wrapper
by Rex Alex 08 Aug '09

08 Aug '09

Hi All, I am new to freeswitch but installed(freeswitch version 1.0.3), configured and tested successfully. Now I want to do the dialling funtions through a php script. Read about Event Socket Library(ESL). How to implement the same in freeswitch. Please assist. Thanks, Rex

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setInputCallback not working with Javascript?
by mszlazak＠aol.com 24 Jul '09

24 Jul '09

I'm getting in build 12653M: [ERR] notify.js:130 mod_spidermonkey()? TypeError: session.setInputCallback is not a function The wiki says this function should work in Javascript. http://wiki.freeswitch.org/wiki/CoreSession_Constructor#session:setInputCal… Also, has there been changes to session.collectInput with type="event"? I get dtmf type events with my callback function but can't seem to get type="event" with speech events. Mark.

3 2

PCMU fallback for T.38
by Gabriel Kuri 01 Jul '09

01 Jul '09

hey folks, I'm trying to configure PCMU fallback for T.38. The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with G729 and PCMU in the sdp. the INVITE to the provider includes G729 and PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... m=audio 16458 RTP/AVP 18 0 100 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 once the FAX tone is detected on the PSTN side, FS receives a T.38 re-INVITE from the provider and FS sends back a 488/Not Acceptable (proxy_media=false). at that point the provider than attempts fallback to PCMU with another reINVITE ... m=audio 16816 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 immediately after the PCMU reINVITE, FS closes the channel and the text below is in the FS logs. given the SPA-2102 included PCMU in the original INVITE, even though it was the second preferred codec, shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by the provider? It seems like it's not passing the PCMU Re-INVITE back to the endpoint (SPA-2102), since it originally negotiated G729 with the SPA2102 as that was the 1st codec in the sdp, but trying to transcode between the two (G729 and PCMU)? 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 PCMU/8000 20 ms 160 samples 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1. 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing Reinvite 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 entering state [completed] 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! Thanks, Gabe

5 8

SIP dump to DB
by kokoska rokoska 18 May '09

18 May '09

Hi all, I'm facing the problem I need all SIP messages "going thru" FreeSWITCH (I know FS i B2BUA - so, better to say just "all SIP messages") logged somewhere and this log have to be "searchable" (by call-id etc) and I should be able to simply delete "old" messages... And more over - it should be done on not trivial SIP messages amount - say hundreds messages per second. My questions are: 1. Do you have any suggestion how to do it with FreeSWITCH? 2. Or it is not possible now, and "bounty" is necessary? :-) 3. How hard it will be to implement? ---------- FYI: I think something like SER/Kamailio/OpenSIPS siptrace is what I'am (probably) looking for: http://www.kamailio.org/docs/modules/1.4.x/siptrace.html It could go (based on my tests) up to 7-10.000 messages per second till MySQL "dies"... ---------- Thanks for your time :-) Best regards, kokoska.rokoska

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pocketsphinx and event socket
by Peter P GMX 16 May '09

16 May '09

Hello, I have tried the pizza demo and did it get to work so far. However I would like to use pocketsphinx through event socket. I saw in the wiki that there is a chapter for Speech Synthesis Commands: http://wiki.freeswitch.org/wiki/Mod_commands#Speech_Synthesis_Commands However this is empty. Also http://wiki.freeswitch.org/wiki/ASR didn't give me a hint. As I am not a Java programmer, it's hard for me to determine how the pizza demo actually works. Anybody has a sample how he did it e.g. in Php/Perl or so? (I am working with Ruby) Or back to the basics: Is it possible to use pocketsphinx through event socket? Best regards Peter

5 15

upper registration in FS?
by xbipin 20 Apr '09

20 Apr '09

i have been trying to do this but seem to have been lost, actually i want to use freeswitch as a session border controller so basically all the clients that try to register to FS will actually be authenticated by voipswitch but i want FS to be in between and proxy everything and with any of its advanced options turned off so as to simply work as topology hiding, proxy, SBC only. -- View this message in context: http://www.nabble.com/upper-registration-in-FS--tp22764757p22764757.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

8 14

[Remote SIP client] Couple of questions
by Gilles 17 Apr '09

17 Apr '09

Hello, I have a couple of questions related to having SIP users connecting from the Net to a Freeswitch server through NAT routers on both ends: 1. How must I configure routers on both ends? I understand that I need to route incoming TCP/UDP 5080 into the Freeswitch server, but what about the other router? I guess I also need to route this port to let the SIP phone ring, but what about data (RTP/RTCP)? 2. The Freeswitch server is connected to the POTS with either an OpenVox PCI card or a Linksys 3102 box: When a call is made between the POTS and a remote SIP phone (ie. out there on the Net, not on the same LAN as the Freeswitch server), is there a way for data to flow directly from the POTS to the remote SIP client instead of through the Freeswitch server? Thank you.

3 4

Calls being cut off while recording a message
by Andy Ayers 07 Apr '09

07 Apr '09

Hi, I'm using freeswitch as a glorified answering machine. FS registers with a VOIP gateway and all calls into the gateway go through an ivr menu and are allowed to leave a message which gets recorded to a file. The FS box is behind a NAT firewall. Everything works fine except that intermittently, calls keep getting cut off after a number of seconds. I've attached a snapshot of the log at the point that the call gets cut off, can anyone suggest why this is happening or how I can prevent it? Many thanks Andy

2 12

High CPU load but only few sessions
by Helmut Kuper 06 Apr '09

06 Apr '09

Hello, since a few days I observe a high CPU load of my FS server, but I have no idea what it could be. There are only a few sessions running and there is only a few log activity. 2 days ago I restarted FS, but no change. The top command shows this: top - 15:02:33 up 106 days, 30 min, 4 users, load average: 0.24, 0.35, 0.42 Tasks: 190 total, 1 running, 189 sleeping, 0 stopped, 0 zombie Cpu(s): 7.2%us, 12.2%sy, 0.0%ni, 80.0%id, 0.2%wa, 0.2%hi, 0.2%si, 0.0%st Mem: 4151776k total, 4003664k used, 148112k free, 414708k buffers Swap: 15623204k total, 88k used, 15623116k free, 2021412k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 14048 ippbx 20 0 1555m 1.2g 10m S 43 30.5 333:11.03 freeswitch 14049 ippbx 20 0 1555m 1.2g 10m S 0 30.5 5:06.88 freeswitch 14054 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:14.38 freeswitch 14055 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:58.50 freeswitch 14057 ippbx 20 0 1555m 1.2g 10m S 0 30.5 13:05.20 freeswitch 20511 ippbx 20 0 1555m 1.2g 10m S 0 30.5 0:00.14 freeswitch so only one process (PID: 14048) is causing that load. It's not the parent process (the initial FS startup process) as ps -elf shows: ippbx@ippbx-prod-node0:~/ippbx.prod$ ps -eLf | grep frees ippbx 14033 1 14033 0 28 Mar23 ? 00:00:01 bin/freeswitch -nc ippbx 14033 1 14034 0 28 Mar23 ? 00:00:08 bin/freeswitch -nc ippbx 14033 1 14035 0 28 Mar23 ? 00:03:39 bin/freeswitch -nc ippbx 14033 1 14036 0 28 Mar23 ? 00:00:07 bin/freeswitch -nc ippbx 14033 1 14037 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 14038 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 14039 0 28 Mar23 ? 00:00:02 bin/freeswitch -nc ippbx 14033 1 14042 0 28 Mar23 ? 00:03:41 bin/freeswitch -nc ippbx 14033 1 14043 0 28 Mar23 ? 00:00:03 bin/freeswitch -nc ippbx 14033 1 14044 0 28 Mar23 ? 00:00:01 bin/freeswitch -nc ippbx 14033 1 14045 0 28 Mar23 ? 00:00:25 bin/freeswitch -nc ippbx 14033 1 14046 0 28 Mar23 ? 00:01:20 bin/freeswitch -nc ippbx 14033 1 14047 0 28 Mar23 ? 00:05:32 bin/freeswitch -nc ippbx 14033 1 14048 7 28 Mar23 ? 05:33:35 bin/freeswitch -nc ippbx 14033 1 14049 0 28 Mar23 ? 00:05:07 bin/freeswitch -nc ippbx 14033 1 14050 0 28 Mar23 ? 00:01:01 bin/freeswitch -nc ippbx 14033 1 14051 0 28 Mar23 ? 00:25:43 bin/freeswitch -nc ippbx 14033 1 14052 0 28 Mar23 ? 00:00:01 bin/freeswitch -nc ippbx 14033 1 14054 0 28 Mar23 ? 00:04:14 bin/freeswitch -nc ippbx 14033 1 14055 0 28 Mar23 ? 00:04:58 bin/freeswitch -nc ippbx 14033 1 14056 0 28 Mar23 ? 00:06:23 bin/freeswitch -nc ippbx 14033 1 14057 0 28 Mar23 ? 00:13:05 bin/freeswitch -nc ippbx 14033 1 14058 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 14059 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20518 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20519 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20521 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20522 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 20526 19854 20526 0 1 15:03 pts/0 00:00:00 grep frees Doing a strace on PID 14048 prints tons of "epoll_wait(21, {}, 4, 0) = 0" lines on the screen, which eats all of my desktop pc's cpu power :/ So can a developer say what this is, or what and how should I debug to find out the cause of this? Can I shot it down via kill or "kill -9" without crashing FS totally? regards helmut

4 8

Originate and Conference
by Peter P GMX 01 Apr '09

01 Apr '09

Hello, when I originate a call via event socket and transfer it to a conference, it doesn't ask for a PIN. Example: originate sofia/default/222331 &conference(222500) The same happens when I originate a call and transfer it to the 222500 destination, which then transfers the call to the conference. originate sofia/default/222331 &transfer(222500) If I manually dial from the phone (222331) to the conference it correctly asks for the PIN. Anybody has a clue why this happens and how to enable the PIN while originating? Best regards Peter

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