November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Grandstream GXP 2000
by Jonas Gauffin 18 Dec '11

18 Dec '11

Hello I have problems with TCP and GXP2000 phones. They work fine using UDP, but no calls arrive if I switch to TCP. I've upgraded to the latest GS firmware today to see it that helped, but it didn't. Have anyone else had problems with TCP? Regards, Jonas

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Hardware phones that do 48Khz Celt: Was Re: Is there any license G729?
by Craig Askings 29 Sep '11

29 Sep '11

Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Craig. 2009/6/30 Jason White <jason(a)jasonjgw.net>: > Brian West <brian(a)freeswitch.org> wrote: >> Everyone has this need for lower bandwidth calls... I tend to march the >> other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a >> single ulaw call) > > 48khz Celt (celt@48000 in your codec preferences) sounds wonderful with > FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. > This also works well in 48khz conferences. > > I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, > G.722, G.722.1 and (my current favourite) Celt all the way. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig(a)overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

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T.38 via UPDATE request
by Peter P GMX 19 May '11

19 May '11

Hello, we have the following problem. 2 Fax machines are communicating via Freeswitch. One is externally attached via a Telco who is able to handle T.38. The other one is attached locally. When 2 Fax machines start syncing each other, the Telco sends a SIP UPDATE message with T.38 SDP, as it detects fax during the fax negociations. Freeswitch answers with an SIP OK message back to the telco, and I can see the T.38 SDP on the debug console of freeswitch. Then nothing happens any more until one of fax machines detects timeout. We have set proxy-media to true. However is was done during call setup when both machines communicated with G711 SDP. The UPDATE message was commited by FS to the telco, but was not sent to the other fax, so I think in this case Freeswitch is supposed to transcode between T.38 and G711 which it cannot do, as we know. How can I overcome this scenario? Is this a defect, should freeswitch send the UPDATE message to the other fax? Or is there a workaround? Best regards Peter Here's the UPDATE message: UPDATE sip:mod_sofia@82.115.xx.xxx:5080 SIP/2.0. Call-ID: 5283fe4e-334f-122d-d1b9-001517956764. Contact: <sip:212.91.xxx.xxx:5060>. Content-Type: application/sdp. CSeq: 16340063 UPDATE. From: <sip:06912345678@sip.telco.de>;tag=00-08135-017e041a-4d21f6037. Max-Forwards: 31. Route: <sip:82.115.96.165;lr;ftag=pH663F4S02erm>. To: "030987654321" <sip:030987654321@82.115.xx.xxx>;tag=pH663F4S02erm. User-Agent: Cirpack/v4.41e (gw_sip). Via: SIP/2.0/UDP 212.91.xxx.xxx:5060;branch=z9hG4bK-178C-1923B31. Content-Length: 300. . v=0. o=cp10 125551618103 125551618105 IN IP4 212.91.xxx.xx. s=SIP Call. c=IN IP4 212.91.xxx.xx. t=0 0. m=image 6860 udptl t38. a=sendrecv. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:1000. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy.

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xml_curl and gateways
by Peter P GMX 21 Feb '11

21 Feb '11

According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to set up gateways dynamically, but I do net get it to work: I always get : "Invalid profile" My assumptions for a right xml answer back to FS are as follows, but I think at least one of it is false: 1. ) I start with <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> 2.) I do not want to change settings, so I do NOT add a <settings> tag 3.) I want to enhance an existing profile, so I add a profile tag (<profile name='external'>) 4.) I assume that I can have an existing profile "external" in the xml conf files and can mix additional parts dynamically into the same profile 5.) Within the profile I want to add gateways, so I add a gateway tag (<gateways>) 6.) Then I add the gateways 7.) I add the closing tags Which one of the assumptions if wrong? The request I receive is as follows: {"key_name"=>"name", "action"=>"directory", "tag_name"=>"configuration", "key_value"=>"sofia.conf", "controller"=>"xml_curls", "hostname"=>"freeswitch", "section"=>"configuration"} I only answer once at startup with the follwing: <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> <profile name='external'> <gateways> <gateway name="freeswitch2"> <param name="username" value="1007"/> <param name="realm" value="192.168.0.56:5061"/> <param name="password" value="xxxx"/> <param name="proxy" value="192.168.0.56"/> <param name="expire-seconds" value="30"/> <param name="register" value="true"/> <param name="register-transport" value="tls"/> <param name="retry-seconds" value="30"/> </gateway> </gateways> </profile> </configuration> </section> </document> Does anybody have a correct XML answer for a gateway and can post it? Best regards Peter

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FreeSWITCH 64bit compilation error (Solaris 10)
by email lists 08 Dec '10

08 Dec '10

Forwarding the issue below to see if anyone is familiar with this issue, and/or what our next steps should be. Thanks, Vladimir Looks like a problem with a Makefile not honoring CFLAGS,etc. Perhaps you can report this to the dev team. Other components built fine but this damn spidermonkey is buggering. # file /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch*.o | head /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-g711.o : ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-getgat eway.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-igd_de sc_parse.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libtel etone_detect.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV FPU] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libtel etone_generate.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniso ap.o: ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniss dpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniup npc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE AMD_3DNow] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniwg et.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minixm l.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] # file /opt/freeradius-client/sbin/radacct /opt/freeradius-client/sbin/radacct: ELF 64-bit LSB executable AMD64 Version 1 [SSE2 SSE FXSR AMD_3DNow CMOV FPU], dynamically linked, not stripped Build output: Making all in nua LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la LINK libsofia-sip-ua.la libtool: link: warning: `-version-info/-version-number' is ignored for convenience libraries Making all in packages Creating mod_sofia_la-mod_sofia.lo mkdir .libs Compiling mod_sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia.lo Compiling sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) "sofia.c", line 3522: warning: enum type mismatch: arg #2 (E_ENUM_TYPE_MISMATCH_ARG) Creating mod_sofia_la-sofia_glue.lo Compiling sofia_glue.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_presence.lo Compiling sofia_presence.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_reg.lo Compiling sofia_reg.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_sla.lo Compiling sofia_sla.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia.la making all mod_speex Compiling mod_speex.c... mkdir .libs Compiling mod_speex.c ... Creating mod_speex.so... making all mod_spidermonkey cd config; /usr/sfw/bin/gmake -j1 export ld: fatal: file now.o: wrong ELF class: ELFCLASS64 ld: fatal: File processing errors. No output written to now gmake[7]: *** [now] Error 1 gmake[6]: *** [export] Error 2 gmake[5]: *** [/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/sfw/bin/gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2

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Freeswitch Video Capture and Playback
by Esben Stien 18 Oct '10

18 Oct '10

I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n

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Siptapi and Freeswitch
by Peter P GMX 10 Sep '10

10 Sep '10

Anybody tried siptapi with freeswitch? http://sourceforge.net/projects/siptapi/ This may enable Click2Dial e.g. from Outlook to Freeswitch. So anybody has experience with that solution? Best regards Peter

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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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Re: [Freeswitch-users] Grandstream gateways
by Samuel Mukoti 23 Apr '10

23 Apr '10

Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews <cory(a)voipsupply.com> wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews(a)sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request(a)lists.freeswitch.org > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris <mike(a)jerris.com> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users(a)lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B(a)jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >>

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