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Where is mod_sms?
by Valery Kalinin 29 Nov '11

29 Nov '11

I update Freeswitch from git, but mod_sms not found. modules.conf - not found src/mod/applications/ - not found How to download mod_sms?

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FreeSWITCH Conference Call Tomorrow: Special Guest
by Michael Collins 29 Nov '11

29 Nov '11

Hello all! I just wanted to give everyone a heads up about tomorrow's conference call. We have a special guest scheduled to join us: Jean-Marc Valin from Mozilla! Jean-Marc is a codec guru who is working diligently on the OPUS codec. He will be coming to talk to us about the codec and to answer any questions that you may have. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_30 We look forward to having you on our call tomorrow! -Michael

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Skypopen audio problems
by asdfa asdfasdf 29 Nov '11

29 Nov '11

Hi, i have a little problem i have FS installed, skypeopen_oss client and skypeopen.ko oss sound driver installed. I configure in public a DID number bridge to skypopen/RR/skype22, the call going from DID to SKYPE1 and call SKYPE22, here is the LOG: http://pastebin.freeswitch.org/17877 All work OK, but the sound from DID caller to Skype22 dont reach, no sound, Skype22 to DID caller works ok. Questions: Skype_oss and the skypeopen.ko require a sound card to work? What is the problem? Codecs? Please helpme, thanks and sorry my english.

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Detecting fax on an incoming call leg (direct T.38 case)
by Roland Hänel 29 Nov '11

29 Nov '11

I'm trying to reliably detect whether there is a fax machine on an incoming channel. Right now, it works nicely with the following dial plan: answer spandsp_start_fax_detect sleep 5000 spandsp_stop_fax_detect [... no fax there, so now do stuff that has nothing to do with fax ...] If the remote end is a PSTN fax, spandsp will fire an event upon detecting the CNG tone, and I will react accordingly. However, if the remove end is a T.38 fax, which will immediately re-invite after answer, then the following problem occurs: FreeSWITCH does not answer the re-invite (just says 100 Trying and nothing follows). Even though the remove end fax transmits the CNG tone afterwards, it is not detected by spandsp. I have recorded the call to make sure the CGN tone is there, and it is there. I have even played this tone into another phone call, where it is successfully detected by spandsp. All I can guess is that the non-answered re-invite might block spandsp from detecting the tone? Greetings, Roland

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Anyone got experience with how limit scales with db back end?
by Dave Horton 29 Nov '11

29 Nov '11

I am using limit with db (sqlite) backend to limit all calls into my server. I have 72 inbound channels on gateways sending to my freeswitch, which is running on a single processor dual core 3GHz server, with the sqlite call limits database on the local disk, and at times I am seeing a delay of up to 3 seconds between the arrival of an invite and FS sending a 180 Ringing from the ring ready application. Other delays in signaling I also see intermittently. The only real processing going on before the ring_ready application is executed is the limit application. I see from the code there is a global mutex that looks like will result in all limit checks being single threaded through the code, and I am wondering if I am experience slight intermittent bottlenecks at peak call times. I realize the hash backend is faster and I may switch to that (decision pre-dated my arrival), but I am wondering if anyone out there has any good data about how many cps the limit application with the db backend can scale to. Obviously its dependent on hardware, but I'd still like to get any real world data (or suggestions) that are out there.

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error compiling 1.0.6 on ubuntu 10.10 64 bits
by Valter Nogueira 29 Nov '11

29 Nov '11

after ./configure, make ends with following message gcc -E /home/fastway/freeswitch-1.0.6/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE$x$=x -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD$x$=x 2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h make: *** [src/include/switch_swigable_cpp.h] Error 1 thanks Valter

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tiff file affects transmission over T.38
by Tim Titov 29 Nov '11

29 Nov '11

First I converted a pdf to tiff with magick and no compression. Then I used tiffcp (3.9.2) to compress it as g3. The file could not be transmitted over T.38 Gafachi or flowroute. It was frustrating to figure this out. I hope others don't run into this problem. P.S.: ghostscript works fine.

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Informal survey: calling multiple endpoints sequentially
by Michael Collins 28 Nov '11

28 Nov '11

Hello all, I'm hoping to get some community feedback as I work on the FreeSWITCH Cookbook. The issue at hand is how you handle the dialing of sequential endpoints. For those of you who do this, would you mind replying to the list and letting us know what you do? I'm curious what methods you use, such as: pipe-sep list of endpoints in a bridge/originate continue_on_fail=true and multiple individual bridge apps mod_lcr or similar Jay Binks' transfer_on_fail method etc. I am particularly interested in the problems that you are trying to solve. In other words - how did the method you chose help you solve your problem? Your perspectives will help me very much. Thanks all! -Michael

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FXS doesn't stop ringing
by Komar, Jason 28 Nov '11

28 Nov '11

Hello Everyone, My current setup uses a Sangoma A400 card with 2 FXO and 2 FXS ports. I am working toward having my analog lines ported over to SIP trunks, but in the meantime I am working with what I have. When a call comes in over the POTS, my public dialplan transfers the call to extension 2000 which is setup in my default dialplan. This extension is setup like the Local_Extension example, but instead of bridging with this line: <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> I bridge to my FXS port: <action application="bridge" data="freetdm/1/1"/> It seems to work well, but the issue I have is that the FXS port keeps ringing the analog phone even after the call bridges to voicemail. I also tested doing a call in and hangup after 1 ring and the FXS keeps ringing and does not detect the hangup. I appreciate any help that someone can offer. Thanks, Jason Komar

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Help Making a Call through External Sip Provider
by daniel mican 28 Nov '11

28 Nov '11

Greetings, I am having difficulty making external phone calls through freeswitch. I successfully receive calls but cannot sned them. I believe the problem is somewhere in here, I am not resolving contact with my sip gateway. It just keeps trying to invite until it times out. I have double checked all config files, and spent time reading the fs wiki's in hope of find what I am doing wrong. Below is the log and a packet. I have been fumbling with this for days and any help at all would be greatly appreciated. Ty 2011-11-24 09:35:13.274794 [DEBUG] sofia.c:5366 Channel sofia/external/+14102944410 entering state [calling][0] nua: nua_handle_magic: entering nta: timer A fired, retransmit INVITE (20736592) tport_release(0x7f7b2c052c00): 0x7f7b2c10b940 by 0x7f7b2c074b10 with (nil) tport_tsend(0x7f7b2c052c00) tpn = UDP/75.101.162.138:5060 tport_resolve addrinfo = 75.101.162.138:5060 tport_by_addrinfo(0x7f7b2c052c00): not found by name UDP/75.101.162.138:5060 tport_vsend(0x7f7b2c052c00): 1089 bytes of 1089 to udp/75.101.162.138:5060 tport_vsend returned 1089 nta: resent INVITE (20736592) to UDP/75.101.162.138:5060 tport_pend(0x7f7b2c052c00): pending 0x7f7b2c10b940 for udp/10.0.0.4:5080(already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/1 term, 0/2 free nta: timer set next to 2000 ms nta: timer A fired, retransmit INVITE (20736592) tport_release(0x7f7b2c052c00): 0x7f7b2c10b940 by 0x7f7b2c074b10 with (nil) tport_tsend(0x7f7b2c052c00) tpn = UDP/75.101.162.138:5060 tport_resolve addrinfo = 75.101.162.138:5060 tport_by_addrinfo(0x7f7b2c052c00): not found by name UDP/75.101.162.138:5060 tport_vsend(0x7f7b2c052c00): 1089 bytes of 1089 to udp/75.101.162.138:5060 tport_vsend returned 1089 nta: resent INVITE (20736592) to UDP/75.101.162.138:5060 tport_pend(0x7f7b2c052c00): pending 0x7f7b2c10b940 for udp/10.0.0.4:5080(already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/1 term, 0/2 free nta: timer set next to 4000 ms nta: timer A fired, retransmit INVITE (20736592) tport_release(0x7f7b2c052c00): 0x7f7b2c10b940 by 0x7f7b2c074b10 with (nil) tport_tsend(0x7f7b2c052c00) tpn = UDP/75.101.162.138:5060 tport_resolve addrinfo = 75.101.162.138:5060 tport_by_addrinfo(0x7f7b2c052c00): not found by name UDP/75.101.162.138:5060 tport_vsend(0x7f7b2c052c00): 1089 bytes of 1089 to udp/75.101.162.138:5060 tport_vsend returned 1089 nta: resent INVITE (20736592) to UDP/75.101.162.138:5060 tport_pend(0x7f7b2c052c00): pending 0x7f7b2c10b940 for udp/10.0.0.4:5080(already 0) 09:35:28.281385 IP (tos 0x0, ttl 64, id 12974, offset 0, flags [none], proto UDP (17), length 1117) 10.0.0.4.5080 > 75.101.162.138.5060: [udp sum ok] UDP, length 1089 E..]2...@.K. ...Ke.......I..INVITE sip:+14102944410@sip1.teltechsys.com SIP/2.0 Via: SIP/2.0/UDP 68.48.6.208:5080;rport;branch=z9hG4bKZ2Q3Km0KNKUcK Max-Forwards: 70 From: "" <sip:+14102944410@68.48.6.208>;tag=3pmg8NZt4SHNQ To: <sip:+14102944410@sip1.teltechsys.com> Call-ID: 5c4496eb-914c-122f-9d94-0025646e9c00 CSeq: 20736592 INVITE Contact: <sip:gw+sip1@68.48.6.208:5080;transport=udp;gw=sip1> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 13-52-28 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-Carrier: sip.flowroute.com X-Carrier-Prefix: 74278017 X-FS-Support: update_display Remote-Party-ID: <sip:+14102944410@68.48.6.208 >;party=calling;screen=yes;privacy=off

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