September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Registering SIP Trunks to Freeswitch
by Matt Putnam 08 Sep '12

08 Sep '12

This may have been asked before but my research hasn't turned up any results. I am currently testing freeswitch as a replacement to our current asterisk gateways. I think I have the basics of ip based trunks down but I am running into an issue with trunks registering to the freeswitch box. In asterisk it was a fairly simple process of giving the trunk a username and password and setting it to dynamic to get it to register. What would be the equivalent in freeswitch that would allow me to have customer trunks register to the system? Thanks, Matt Putnam

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mod_perl installation error
by Hardik Patel 07 Sep '12

07 Sep '12

Hi, I am doing installation of freeswitch with mod_perl module but during compilation i got below error. ********ERROR************** making all mod_perl Creating mod_perl.la... /usr/bin/ld: /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a(op.o): relocation R_X86_64_32S against `PL_sv_yes' can not be used when making a shared object; recompile with -fPIC /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a: could not read symbols: Bad value collect2: ld returned 1 exit status cat: .libs/mod_perl.log: No such file or directory make[5]: *** [mod_perl.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_perl-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 *********************************************** making all mod_perl Creating mod_perl.la... /usr/bin/ld: /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a(op.o): relocation R_X86_64_32S against `PL_sv_yes' can not be used when making a shared object; recompile with -fPIC /usr/local/lib/perl5/5.15.4/x86_64-linux/CORE/libperl.a: could not read symbols: Bad value collect2: ld returned 1 exit status cat: .libs/mod_perl.log: No such file or directory make[5]: *** [mod_perl.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_perl-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 can anybody please help me to figure out the issue? Thanks in advance. Thanks, Hardik Patel

2 1

freeswitch - sangoma CAS T1 - a101 configuration
by copycall 07 Sep '12

07 Sep '12

hi, i'm having trouble getting instructions to configure freeswitch with a sangoma a101 T1 card for CAS, B8ZS, ESF. the first application i seek to solve is using this setup as a gateway (replacing a dialogic dmg 2000) for sip from my itsp to the tdm endpoint: a legacy centigram - mitel voicemail server. any help will be greatly appreciated. thank you, dave

4 5

Numerous compile errors
by Carl Johan Jensen 07 Sep '12

07 Sep '12

Hello out there, I have installed Visual C++ 2008 Express edition on my Windows 7 machine. I downloads the freeswitch-1-0-6-tar-gz file from http://files.freeswitch.org. I extract the source code and opens the project in Visual C++. Then I press F7 to start compilation. Numerous errors are returned: At the end of compilation: 120>LINK : fatal error LNK1181: cannot open input file '..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib' 120>Build log was saved at "file://c:\R&D\FreeSwitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\Win32 \Debug\BuildLog.htm" 120>mod_sofia - 1 error(s), 12 warning(s) ========== Rebuild All: 102 succeeded, 17 failed, 11 skipped ========== I investigate the errors and find among several the below errors: 13>c1 : fatal error C1083: Cannot open source file: '..\..\pthreads-w32-2-7-0-release\pthread.c': No such file or directory 13>Build log was saved at "file://c:\R&D\FreeSwitch\freeswitch-1.0.6\libs\win32\pthread\Debug DLL\BuildLog.htm" 13>pthread - 1 error(s), 0 warning(s) 14>Deleting intermediate and output files for project 'make_modem_filter', configuration 'All|Win32' 14>make_modem_filter : error PRJ0008 : Could not delete file 'c:\R&D\FreeSwitch\freeswitch-1.0.6\libs\spandsp\src\msvc\All\RSP0001D534166 324.rsp'. 14>Make sure that the file is not open by another process and is not write-protected. 14>make_modem_filter - 1 error(s), 0 warning(s) 19>agc.c 19>c1 : fatal error C1083: Cannot open source file: '..\..\sphinxbase-0.4.99\src\libsphinxbase\feat\agc.c': No such file or directory 19>bio.c Anyone knows what I miss to do. (I am following the instructions of the book "FreeSWITCH-106-eBook06092012_1187154") Best Regards Carl Johan

2 1

Freeswitch compatible GSM Gateway for 30+ channels
by Vishal Kakkar 07 Sep '12

07 Sep '12

Hi All, I am looking for *Freeswitch compatible GSM Gateway for 30+ channels(SIMS)*. Please suggest in case you have used any. Thanks.

2 1

Help needed regaridng Contact Center solution
by Nitin Tomer 07 Sep '12

07 Sep '12

Hi, I am working on a Contact Center solution. It will support mail, chat and call queries. The requirements are: 1. An end-customer calls, the call is handled by FreeSWITCH Auto Attendant. 2. Customer is presented with a menu and makes selection. His call is put on hold and an entry is made in my system's database for incoming queries. 3. These queries are shown to agents handling calls. 4. And agent clicks on a query, he is shown an extension where call is parked. He dials that and is connected to the customer. 5. He talks to the customer and resolve his queries. Please guide me on how can do it. The application will be written in Java. I am an experienced programmers in Java/J2EE; but doesn't have much knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on Ubuntu 12.04; and have installed X-Lite Softphones on two Windows machines. I configured these phones to work with FreeSWITCH and they are working fine. I think that I'd have to use Valet Parking to park customer's call on an extension, then pass that extension to agent who will dial it and will be connected to the customer. Please tell me whether this approach will work? And how should I go about it. All help would be much appreciated. Regards Nitin

20 31

continue_on_fail problem
by Evgeniy Movlyan 07 Sep '12

07 Sep '12

Hi to all! For example i have 2 phones: 7604503 and 7604504. If i'm calling to 7604504 and it's busy, then call must be transfered to 7604503. My dialplan is: <context name="default"> <extension name="BESTNET"> <condition field="destination_number" expression="^(7604504)$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=USER_BUSY"/> <action application="set" data="tansfer_on_fail=USER_BUSY BUSY XML error"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> </context> <context name="error"> <extension name="BUSY"> <condition field="${originate_disposition}" expression="^USER_BUSY$" continue="false" break="on-true"> <action application="log" data="IM HERE!!!!!!!!!!!!!!!!"/> <action application="set" data="dialed_extension=$1"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=false"/> <action application="bridge" data="user/7604503@${domain_name}"/> </condition> </extension> </context> But it's doesn't work:( Please help and sorry for my english:) -- Evgeniy Movlyan, BestNet Ltd.

1 0

gtalk rings once and cuts
by Tichafara Sigauke 07 Sep '12

07 Sep '12

I have managed to setup mod_dingaling on my Freeswitch, I would like to route inbound calls from gtalk to my IVR running on Freeswitch but calls cut after one ring. My setup looks like: gtalk client A-------------------------------->gtalk Client@Freeswitch----------------->IVR on the same Freeswitch I have attached the fs-cli debug fs_cli_debug.txt and profile setup client_profile.txt, I would appreciate any help. Tich

1 0

Attended transfer to a conference room
by Emrah 07 Sep '12

07 Sep '12

Hi all, I am experiencing a strange issue with SIP based attended transfers. If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up. There is no much activity on the SIP side of things, it seems to be very much related to FS. Some info: I call out through a provider configured on the external profile, from a phone registered on the internal profile. It is not a codec conflict. Both lines are answered when I actually finalize the transfer. I tried with multiple phones and softphones. I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated. Best, Emrah

4 15

Phone that does IPv6 properly?
by Tim St. Pierre 07 Sep '12

07 Sep '12

Hello, I'm excited about moving to IPv6, and not having to deal with NAT ever again. We have IPv6 address space and transit, and FS does IPv6, so now I need some phones. I'm told that there will be IPv6 firmware coming for my 400+ Aastra phones, but that might be a while. Are some IP phones out there today that have good IPv6 support? Specifically, it has to do the usual network protocols (SLAC, router solicitation, etc.) DHCP6 support would be nice. It has to be able to look up an SRV record, and then get AAAA records for the SIP proxies referenced. Everything is native IPv6, so it isn't really important that it can do tunnels or anything special. I'm not even sure that I need IPv4 on it at all, since it will be deployed in an IPv6 only environment. G.722 audio would be nice too. Thanks for any suggestions! -Tim

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