July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_spandsp dosn't compile
by mbo 16 Jul '12

16 Jul '12

Hello, I just updated my freeswitch test installation (make sure). mod_spandsp doesn't compile anymore, I get the following error: making all mod_spandsp make[5]: Entering directory `/usr/src/freeswitch/src/mod/applications/mod_spandsp' Creating mod_spandsp_la-mod_spandsp.lo quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109: error: expected specifier-qualifier-list before ‘t81_t82_arith_encode_state_t’ /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202: error: expected specifier-qualifier-list before ‘t81_t82_arith_decode_state_t’ In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:118: error: expected specifier-qualifier-list before ‘t85_decode_state_t’ In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:132: error: expected specifier-qualifier-list before ‘t85_encode_state_t’ In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33: error: expected specifier-qualifier-list before ‘ademco_contactid_report_func_t’ cc1: warnings being treated as errors /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48: error: struct has no members Thanks Markus

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One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true)
by Adelia C. 16 Jul '12

16 Jul '12

We're testing a feature that requires an extra Application Server (B2BUA) on our call path, on the egress side of the call from FreeSwitch. With the extra AS, we have only one-way RTP (Caller -> Callee). Without, RTP is fine. We narrowed it down to FS enforcing symmetric RTP while AS only works with asymmetric RTP. We cannot change the behavior of the AS at this time. Call: Caller - >NTW -> SBC -> FS -> AS -> SBC -> NTW -> Callee. RTP streams, as negotiated (SDP): Caller - > SBCOut:50460 ->FSIn:29526 -> FSOut:17818 -> ASIn:8080 -> ASOutB:19040 -> SBCOut:49162 -> Callee (can't hear a thing) Callee -> SBCOut:49162 ->ASIn:8082 -> ASOut:19000 -> FSIn:17818 -> FSOut:29526 -> SBCIn:50460 -> Caller (good RTP) RTP streams, as sent/receiced: FS negotiates 8080 as the RTP destination to AS in the "caller - > callee" direction. For about .5s, that's how the RTP flows : FSOut:17818 -> ASIn:8080 FS negotiates 19000 as the RTP source from AS in the "callee -> caller" direction. This is how the RTP flows for the duration of the call.: ASOut:19000 -> FSIn:17818 FS switches its RTP stream in the "caller - > callee" direction ~.5s into the call : FSOut:17818 -> ASIn:19000. [ASIn listens on 8080, the caller - > called RTP is dead.] All this from Wireshark trace. I tried the following, with no success: 1. Added to MyFSApp.xml and dialplan\public.xml: <action application="set" data="disable_rtp_auto_adjust=true"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/> Restarted, FS, same result. 2. Commented out <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="use-rtp-timer" value="true"/> Restarted, FS, same result. 3. Started FS with -nonat, then nonatmap Same result. Other info : - Our App on FS runs in B2BUA mode, doesn't proxy the call. We answers the incoming leg before initiating the outgoing leg. I can attach the Wireshark is I didn't provide enough info - I have both FS and AS pcaps. - Our FS code about 2 weeks old. What else can I try? Any other settings I can enable/disable? Can I enable extra RTP logs on FreeSwitch for debug? Thanks, A.C.

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Outgoing Registration Contact
by Andrew Cassidy 14 Jul '12

14 Jul '12

Hi all, is it possible to force FreeSWITCH to use an FQDN in the contact field of an outgoing registration? Say for example you wanted to haev a trunk that you have to register terminate at multiple public IP addresses. Is that something that can be done with FreeSWITCH at the moment? It's not a massive deal, I'll probably end up putting in SBC(s) at some point. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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Using the freeswitch.spec to build RPM
by Wood Peter 14 Jul '12

14 Jul '12

Hi, I'm using CentOS5 and I can successfully compile and install freeswitch using the latest Git clone. But, I'm not very successful building an RPM with the freeswitch.spec file. I couldn't find any documentation or instructions how to build the RPM. This will be a great start for me instead of sending long error messages to the list. Any pointers are appreciated. Thank you Peter

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Failed to compile on CentOS5
by Wood Peter 14 Jul '12

14 Jul '12

Latest Git clone fails to compile on CentOS5. The compile error out with this: ... Compiling src/switch_regex.c ... Compiling src/switch_rtp.c ... src/switch_rtp.c: In function 'read_rtp_packet': src/switch_rtp.c:2950: error: 'switch_rtp_t' has no member named 'rtp_recv_msg' src/switch_rtp.c: In function 'read_rtcp_packet': src/switch_rtp.c:3069: error: 'switch_rtp_t' has no member named 'rtp_recv_msg' make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 make: *** [all] Error 2 Using the last commit from yesterday compiles successfully: git checkout ba6c404 Any idea what could be wrong? Thank you, Peter

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Polycom IP501
by Todd Bailey 14 Jul '12

14 Jul '12

Hey all, I just purchased this phone ( ebay $35 total) and should be here in a few days. I think I got a boot server (tftp) all set up so the phone will have a place to get the latest boot code. I've read through most of the FS user docs on this phone and it seems like it should be plug and play after I get the configuration files updated and in place. I not sure what to expect, for the initial install. Basic functionality and maybe a few higher features or does the fun begin after the phone is installed? This phone is intended as a beta test, if it works ok, meaning plug in and it works and advanced features don't take days or weeks to sort out, I'll be adding a few more. Any one have experience with this particular phone on FS and any issues or gotcha's to be aware of? thanks

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_sip._udp SRV DNS records on Windows 2008 R2 server
by Sean Devoy 14 Jul '12

14 Jul '12

Hi all, This is a Windows DNS Server question. I have added SRV records for _sip._udp.bizfocused.com, even though _sip is not one of the stock choices in the dropdown list from the add SRV record dialog. I can see them in DNS console listed under my domain: bizfocused.com +_udp (2 _sip records show up here) However, I cannot find any tool that can successfully request/display those values (like nslookup). And of course when I turn on "Use DNS SRV" on my phone it just hangs on "CHECKING DNS". Any ideas for hosting _sip._udp SRV records would be greatly appreciated. Do I have to move my DNS to a UNIX box or DNS Service provider? Thanks, Sean

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FreeSwitch Failover Question
by Sean Devoy 13 Jul '12

13 Jul '12

Hi All, I have a question concerning FS failover for high availability. My SIP provider offers the ability to route to a second server if the primary is down. That is simple enough, but what about the phones? The desktop devices are not registered with the back FS server. The only idea I have so far would be to change the DNS addresses to the backup upon failure, but that can take a long time to propagate to the phones. How do other people handle this issue? Thanks, Sean

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Anyone want my SSD tips on the FS Wiki?
by Mario G 13 Jul '12

13 Jul '12

I recently spent a lot of time learning how to keep an SSD fast and increase its life. It took a ton of research and I combined a lot of info from several sources to install the SSD. Turns out, just installing an SSD and using it as an HDD without doing anything else is not a good idea. Although the topic is not specific to FreeSwitch I am wondering how many out there would like me to put together a FS wiki page fr tuning an SSD. It's a lot of work so I won't do it unless there are enough requests or the developers want it. Mario G

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trunk_1 registration failed (timeout)
by Aaron Berndsen 13 Jul '12

13 Jul '12

My freeswitch configuration was working fine until I upgraded to ubuntu 12.04. Now sofia status indicates that registering trunk_1 to my sip provider times out: 2012-07-12 09:02:51.252018 [WARNING] sofia_reg.c:449 Timeout Registering trunk_1 2012-07-12 09:02:53.252075 [WARNING] sofia_reg.c:468 trunk_1 Failed Registration [908], setting retry to 45 seconds. A major clue is that sofia_dig on the ipv4 of my sip provider returns instantly, while if I use the dns hostname it takes about 3 minutes to return the information. freeswitch@internal> sofia_dig 64.254.145.209 Preference Weight Transport Port Address ================================================================================ 1 1.000 udp 5060 64.254.145.209 2 1.000 tcp 5060 64.254.145.209 Ubuntu 12.04 introduced dns-cacheing, but I have tried turning this off (comment out dns=dnsmasq in /etc/NetworkManager/NetworkManager.conf) and the problem persists. Any tips or suggestions would be helpful. I should point out that my system is behind a router, though I've had the FS configuration working with NAT before. Cheers, -Aaron

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