- FreeSWITCH-users - FreeSWITCH Mailing Lists

process-rxfax.py and announcing that a fax was detected/connection grabbed
by Trever L. Adams 29 Mar '12

29 Mar '12

Hello everyone, I am new to FreeSWITCH (several months, but a few bugs kept me really from digging in until now). I am setting up auto-rx for faxes for a home PBX. It works. However, the person who answers the phone hears a beep, then a click. I am hoping there is a way to cause FreeSwitch to play an announcement from inside the process-rxfax.py or the dialplan calling it so that the person who answers the phone knows a fax was detected and that they can hangup. This should only play to the internal user, and not affect rxfax at all. Is this possible? How would I go about splitting the legs of the connection or what not to do that? Thank you, Trever -- "Arbitrary power is most easily established on the ruins of liberty abused to licentiousness." -- George Washington

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multiple bridge_pre_execute_bleg_app
by Trever L. Adams 29 Mar '12

29 Mar '12

Hello Everyone, I am trying hard to do my various setups in a moduluar way. Unfortunately, I have hit a problem where I am binding internal transfers on incoming calls to *01 and *00 to crap on those violating the do not call list. Both are being done in two different xml files in the Incoming dialplan with bridge_pre_execute_bleg_app. Both work if only one of the two is in the dialplan. Put both in and only the second file processed is picked up. Is it possible to do both in different files and have one somehow readd the original entry? Thank you, Trever

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Registration VIA TCP
by Rob Moore 29 Mar '12

29 Mar '12

Hi Guys, Bit of an odd one here. We've been using UDP as the Transport Protocol for a Good while with a number of different SIP Phones but main Yealink SIP-T22P's. Due to some recently changes around the network needing to make things a little more robust we've had to look at shifting registration to TCP (and eventually to TLS but I have this issue to master first). We've had our SIPPhones working on multiple sites using UDP on a setup similar to this: [sipPhone]----------[SonicwallTMG]--------------(internet)---------------[SonicwallTMG] --------[Freeswitch] When changing to TCP phones can register fine and call to the outside world via freeswitch and our gateway providers however when attempting to make a call from the freeswitch box to the sip phone the call times out with NORMAL_TEMPORARY_FAILURE. This would suggest that the problem is related to NATing but we have exactly the same NAT rules for UDP as TCP in both firewalls and calls from the Freeswitch box with UDP can reach the sipPhone fine. Also doing a local test with TCP/TLS in our office works fine with STUN disabled and no other firewalls / routers in the way. I've tried a combination of various phones and configurations all with STUN enabled but as yet to no avail. I'm going to dig deeper on this tomorrow however I thought I'd send something out here to see if there is something simple that I've missed. The problem stinks of an issue with basic configuration of either Freeswitch or the firewalls we have in place but I can't see anything immediately that could be causing an issue as fundamental as this. If Anyone has any suggestions or has experienced this before I'd appreciate any help or direction on offer! Thanks in Advance Rob.

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Forward calls from opensips to freeswitch
by Sherif Omran 29 Mar '12

29 Mar '12

Hello guys, any body knows how to forward calls from opensips dispatcher to freeswitch? Opensips + freeswitch are running on the same server. 1- Opensips dispatcher set to send calls to 127.0.0.1:5070 2- Freeswitch is set to use internal profile at 5070 (please help to check) and other settings? thanks Sherif Omran

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mod_callcenter and moh-sound
by Vik Killa 28 Mar '12

28 Mar '12

The wiki says you can use the channel variable ${hold_music} (http://wiki.freeswitch.org/wiki/Mod_callcenter#moh-sound) Yet when you use <param name="moh-sound" value="${hold_music}"/> You will get: mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/${hold_music}.PCMU If you use <param name="moh-sound" value="$${hold_music}"/> It will play the hold music set in the global variable. Is there any way for mod_callcenter to take a channel variable as the music on hold?

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Compiling Latest GIT
by Peter Fraser 28 Mar '12

28 Mar '12

Hi, I am having all sorts of trouble with compiling freeswitch on Ubuntu 11.10. I have downloaded the source from the latest GIT and and followed the wiki. I have used freeswitch many times in the past however I am not the best on compiling with make etc. I am getting an error after i run the initial make. ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1890:56: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1893:57: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:2945:2: error: #endif without #if nua_subnotref.c: In function 'nua_refer_client_response': nua_subnotref.c:915:7: error: 'sip_refer_sub_t' undeclared (first use in this function) nua_subnotref.c:915:23: error: expected ';' before 'const' nua_subnotref.c:917:11: error: 'rs' undeclared (first use in this function) cc1: all warnings being treated as errors make[9]: *** [nua_subnotref.lo] Error 1 make[8]: *** [all] Error 2 make[8]: *** No rule to make target `nua/libnua.la', needed by ` libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/usr/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/ libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 schepisi@gfi-fraser:/usr/src/freeswitch$ There are more errors but this is where it bombs out. Is anyone able to provide any pointers as to what I might be doing wrong. Thanks in advanced. Peter Fraser

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Re: [Freeswitch-users] Trouble with transfer and voicemail password
by Trever L. Adams 28 Mar '12

28 Mar '12

> go ahead and get a debug log of a call coming in, being answered, and the > recipient dialing *0#xxx# to send the call elsewhere. Drop it into > pastebin.freeswitch.org and use "FreeSWITCH Log" as the syntax highlighting. > > -MC http://pastebin.freeswitch.org/18753 I am not sure how much the log will help. It doesn't show me pushing anything after the transfer code (*01 as I was experimenting and forgot to change it back to *0# as I had it set before). If I dial any thing 1000, 1000#, 1001, 1001# it all fails and plays the invalid_extension.wav. I have changed the code a little bit. It now reads: <extension name="setup bind_digit_action_transfer"> <condition field="destination_number" expression="^SETUP_TRANSFER$"> <action application="log" data="INFO Configuring bind_digit_action to do recording on this session..."/> <action application="bind_digit_action" data="do_transfer,*01,exec:execute_extension,DO_TRANSFER XML Incoming-FXO"/> <action application="digit_action_set_realm" data="do_transfer"/> </condition> </extension> <extension name="setup transfer" continue="true"> <condition field="chan_name" expression="^FreeTDM/3:1/"> <action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/> <action application="set" data="bridge_pre_execute_bleg_data=SETUP_TRANSFER XML Incoming-FXO"/> </condition> </extension> <extension name="bind_digit_action Do Transfer"> <condition field="destination_number" expression="^DO_TRANSFER$"> <action application="log" data="INFO Transferring external call to another internal location..."/> <action application="play_and_get_digits" data="1 10 1 1000 # misc/silence.wav misc/invalid_extension.wav target_extension ^(\d+)$"/> <action application="log" data="INFO Transferring external call to internal ${target_extension} ${1}..."/> <action application="transfer" data="-bleg ${target_extension} XML Internal-FXS"/> </condition> </extension> It seems that I had a problem with play and get digits not working when I was working on my call screener (I stopped work do to bug openzap-137). I seemed to have gotten around it by rewriting the entire thing in lua. I may be remembering incorrectly as this was months ago. Thank you for any help, Trever -- "Selfishness is really self-destruction in slow motion." -— Elder Neal A. Maxwell - Ensign, May 1999, 23

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Re: [Freeswitch-users] Binary download with curl
by Peter Olsson 28 Mar '12

28 Mar '12

Thanks for the input. But I think this is only true if you put the sources under a protected directory, like "C:\Program Files"? If the sources are put somewhere else it should work even with UAC enabled (at least it does for me :)). /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Fu Jiantao Skickat: den 27 mars 2012 11:45 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Binary download with curl Hi , I've got some experience to share with you guys, if you're using WIN7, please turn off the "UAC", or else, the download vb script will failed to run. Hope this can help! 2012/3/27 Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> If you follow the instructions on the wiki the build should work out of the box. The most common problem is that you've enabled autocrlf in git, which causes the build to break. If you still have problems after following the instructions, please report to Jira, since this is something that should be working. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Olli Aro Skickat: den 26 mars 2012 10:05 Till: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: [Freeswitch-users] Binary download with curl Hi all, Been trying to compile Freeswitch on Windows in order to get in the XML curl module, however it is not looking easy (76 more fatal errors to figure out...) Before I start that task I thought it is always worth asking :) Would anyone happen to have a download link to already compiled Windows 32-bit binary for close to recent version of Freeswitch with the XML curl module? Regards, Olli _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f718ac732761986176360!

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Congrats to Gabe Gunderson - New Baby Girl
by Michael Collins 28 Mar '12

28 Mar '12

Hey all, Just a hat tip to Gabe Gunderson (IRC: gundy) whose wife delivered a baby girl yesterday! Now that that business is done he's back to the office to work on the ClueCon website. :P Congrats Gabe! We'll expect to see some pics when we're all having a beer at ClueCon. -MC

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Best practice to use t38_gateway for transcoding from T.38 to ALAW
by Anton Kvashenkin 28 Mar '12

28 Mar '12

Hello, list. My ITSP supports T.38 for faxing pretty well and i _can_ send fax using ATA D-Link 5008s with t38_passthru <action application="export" data="t38_passthru=true"/> Now i want to use FreeSWITCH-box as the fax gateway to send all faxes to asterisk box using transcoding from T.38 to just audio. My setup for outbound call: FAX --> ATA --> T.38 --> FS --> T.38 --> ITSP --> customer PSTN FAX This setup, using t38_passthru, works pretty reliable. Now i wanna: FAX --> ATA (fax disabled for purpose to not RE-INVITE) --> ALAW --> FS --> T.38 --> ITSP --> customer PSTN FAX or FAX --> ATA (fax disabled for purpose to not RE-INVITE) --> ALAW --> Asterisk --> ALAW --> FS --> T.38 --> ITSP --> customer PSTN FAX asterisk as a registrar for all endpoints @ office and FS (virtual machine) act as simple *sic* fax gateway. So, I swear, i tried all the configuratings from wiki ( http://i1110.photobucket.com/albums/h447/jugatsu85/280312-1.jpg) but it doesn't work at all. According to your requestd i'll post all pcap/flat dumps (actually, when i resolve a issue with my ITSP sending 403 Forbidden ^_^)

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