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Error compiling latest git version on Fedora 14
by Juan Castro 13 Feb '12

13 Feb '12

Note: I was able to compile it in the same conditions with last Friday's (Feb 10 2012) git. With today's (Feb 13) I can't. After ./bootstrap.sh and ./configure, the modules I enabled were: xml_int/mod_xml_curl endpoints/mod_rtmp applications/mod_callcenter applications/mod_curl applications/mod_http_cache applications/mod_voicemail_ivr The craziest thing: when I go to /usr/src/cloudivr/freeswitch.test/src/mod/xml_int/mod_xml_rpc it seems that last gcc line ran OK! Re-running it confirms it. So, where does the error come from? (snip...) Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_parse.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_serialize.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_abyss.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_cgi.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_string.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_struct.c... Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c... Compiling /usr/src/cloudivr/freeswitch.test/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c... quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/cloudivr/freeswitch.test/src/include -I/usr/src/cloudivr/freeswitch.test/src/include -I/usr/src/cloudivr/freeswitch.test/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/cloudivr/freeswitch.test/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC -o .libs/mod_xml_rpc.o quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/cloudivr/freeswitch.test/src/include -I/usr/src/cloudivr/freeswitch.test/src/include -I/usr/src/cloudivr/freeswitch.test/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/cloudivr/freeswitch.test/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o mod_xml_rpc.o >/dev/null 2>&1 Creating mod_xml_rpc.la... make[1]: ** [all-recursive] Erro 1 make: ** [all] Erro 2 #

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Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem
by David Brazier 13 Feb '12

13 Feb '12

Hi Michael/Chris Did you find out what the problem was? We're seeing something similar. Thanks in advance David On Mon Mar 14, Michel Habib <michelhabib at gmail.com>wrote: > > Sorry for the delay and thanks Chris, > Please find attached logs: http://www.mediafire.com/?8thg9ey7kd26882 > sip client ip 192.168.1.202 > freeswitch ip 192.168.1.35 > mrcp connector/speech server ip 192.168.1.32 > i have attached 3 logs - freeswitch log file, wireshark on freeswitch > server, wireshark on mrcp connector/speech server. > my script simply plays back audio from wave [basic freeswitch function], > then plays back audio from TTS [which cannot be heard] > > Thank you, > Michel. > > ---------- Forwarded message ---------- > > From: Christopher Rienzo <cmrienzo at gmail.com> > > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> > > Date: Wed, 9 Mar 2011 22:25:34 -0500 > > Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server > > [using MRCP Connector] - Audio Problem > > I still would like to see the logs for your call. > > > > > > On Tue, Mar 8, 2011 at 7:18 AM, Michel Habib <michelhabib at gmail.com > >wrote: > > > >> Yes, I get the Audio from FS in regular calls - I already disabled all > >> possible firewalls - all 3 machines [softphone, freeswitch, Speech > Server > >> (and mrcp connector) ] are on a switch. > >> 192.168.5.107 is the freeswitch server > >> 192.168.5.110 is the MRCP-Connector/Speech Server 2007 server > >> I made too many iterations on the configuration below: > >> > >> <include> > >> <profile name="mrcp-connector" version="2"> > >> <param name="client-ip" value="192.168.5.107"/> > >> <param name="client-port" value="5090"/> > >> <param name="server-ip" value="192.168.5.110"/> > >> <param name="server-port" value="5070"/> > >>  > >> <param name="sip-transport" value="udp"/> > >>  > >>  > >> <param name="rtp-ext-ip" value="192.168.5.107"/> > >> <param name="rtp-ip" value="192.168.5.107"/> > >> <param name="rtp-port-min" value="4000"/> > >> <param name="rtp-port-max" value="5000"/> > >>  > >> <param name="rtcp" value="1"/> > >>  > >> <param name="rtcp-bye" value="2"/> > >>  > >> <param name="rtcp-tx-interval" value="5000"/> > >>  > >> <param name="rtcp-rx-resolution" value="1000"/> > >> > >>  > >>  > >>  > >> <param name="codecs" value="PCMU PCMA L16/96/8000"/> > >> > >>  > >> <synthparams> > >> </synthparams> > >> > >>  > >> <recogparams> > >>  > >> </recogparams> > >> </profile> > >> </include> > >> > >> > >> > >> ---------- Forwarded message ---------- > >> > >>> From: Christopher Rienzo <cmrienzo at gmail.com> > >>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> > >>> Date: Mon, 7 Mar 2011 09:32:51 -0500 > >>> Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server > >>> [using MRCP Connector] - Audio Problem > >>> Do you get audio between FS and your SIP client when not using ASR/TTS? > >>> > >>> Show me the MRCP profile configuration and your FreeSWITCH logs during > >>> the call. > >>> > >>> > >>> > >>> On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib at gmail.com > >wrote: > >>> > >>>> Hello All, > >>>> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP > >>>> calls and use its ASR and TTS Services successfully] > >>>> I am also using MRCP Connector from AumTech - which allows me to use > ASR > >>>> and TTS Services through an MRCP Client . > >>>> Now, i am using Freeswitch mod unimrcp to use ASR and TTS. > >>>> > >>>> for TTS, I can successfully make the call, the Audio RTP of the TTS > >>>> voice is transferred succesfully from Speech Server [through MRCP > Connector] > >>>> back to the Freeswitch Server. > >>>> However, Freeswitch is not sending back the Audio RTP to the SIP > client. > >>>> > >>>> for ASR, I can successfully define the grammar and start recognition, > >>>> but the audio RTP sent to speech server [through MRCP Connector] is > silent > >>>> [empty]. > >>>> > >>>> I am suspecting something is wrong with the RTP Configuration - can > you > >>>> help me? > >>>> > >>>> Let me now if you need any specific logs/scripts/configuration? > >>>> > >>>> Thank you, > >>>> Michel. > >>>> > >>> >

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Access the Remote-Party-ID header from a SIP invite.
by Gael Martin 13 Feb '12

13 Feb '12

Hi, I need to read the content of the Remote-Party-ID header in a SIP invite from a provide but I can't seem to find the right channel variable that has the content I tried: <action application="log" data="CONSOLE RPID ${sip_h_Remote-Party-ID}"/> <action application="log" data="CONSOLE RPID ${Remote-Party-ID}"/> <action application="log" data="CONSOLE RPID ${sip_h_remote-party-id}"/> <action application="log" data="CONSOLE RPID ${sip_rpid}"/> But it's always blank, even so I can see the Header in the SIP invite. Any ideas? Gael

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High CPU usage with only 30 channels?
by Tim St. Pierre 13 Feb '12

13 Feb '12

Hello, I'm having some scaling problems with our Freeswitch system. The machine running Freeswitch seems to be working very hard during the day, and we don't have that big a load. I need to be able to double our subscriber base, and I don't really want to go to a cluster just yet. Here's what top -CP displays during busy hour: tim@stefan%top -PC last pid: 75968; load averages: 4.64, 2.92, 1.74 up 10+08:54:14 11:17:25 129 processes: 1 running, 128 sleeping CPU 0: 4.5% user, 0.0% nice, 1.9% system, 0.0% interrupt, 93.6% idle CPU 1: 3.8% user, 0.0% nice, 2.6% system, 0.0% interrupt, 93.6% idle Mem: 632M Active, 2093M Inact, 322M Wired, 83M Cache, 384M Buf, 785M Free Swap: 4096M Total, 36K Used, 4096M Free PID USERNAME THR PRI NICE SIZE RES STATE C TIME CPU COMMAND 58521 freeswitch 98 74 r30 749M 648M ucond 0 528:54 0.10% freeswitch 983 root 1 44 0 10824K 2272K select 1 0:40 0.00% ntpd As you can see, the load average is pretty high, at least for something not that big. FS often has up to about 110 threads going. At night, this will settle right down to a 0.05 load average. CPU load never gets that high, but the machine still seems busy. Here's some other details: 40 channels, nearly all of them in CS_EXECUTE or CS_EXCHANGE_MEDIA We see only a few CPS, maybe 10 if it's really busy, but most calls are longer. Nearly all our customers are regular office users. There are 340 phones registered. About 300 of those are behind NAT getting an options-ping every 30s. They re-register every 5 minutes, so we see around 1 registration attempt per second, although it can be a little bursty. Each phone has at least 5 BLF subscriptions, There are 1156 rows in the sip_subscriptions table. All our trunks are SIP. Trunk side is g.711u and endpoints can do either g.711u or g.722 We use ODBC in the core - as well as for sofia, voicemail, and anything else that can use it. Directory and dialplan are done using CURL and handled by another server. The database is also on another server. Our MySQL system sits at about 100 Queries per second during the day , down to about 80 qps at night. This system started to choke until I changed a lot of the tables to the MEMORY engine and installed SSDs as the primary storage devices. Now the database is pretty happy - about 0.5 load average at the moment. It was up to 2 with 60% CPU going to mysqld before I went to memory tables. I keep hearing about FS systems handling thousands of users on one box. I'm nowhere near that and it seems to be maxed out. This system has a dual core Xeon 5100 at 2.6 GHz and 4GB of ram. Memory doesn't seem to be the issue, as there is still some unused. Operating system is FreeBSD 8.0-RELEASE (amd64) Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) I'm trying to update, but newer version break functionality, so I have to find a release that work before I can upgrade. What should I look for to remedy this, or is this considered normal and I just need more power? I could easily upgrade this box and install a pair of quad core chips. Any other information that would be helpful?

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IVR greet-long repeat
by George Sapak 13 Feb '12

13 Feb '12

any way to repeat the greet-long message a few time without user needing to press any buttons? Thank You, George

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Rotate cdr-records as root user (when FS is running from freeswitch user).
by Henrik Aagaard Sørensen 13 Feb '12

13 Feb '12

I'm having FS running from the user freeswitch on CentOS 6. When calling (as the user root): /usr/local/freeswitch/bin/fs_cli -x 'cdr_csv rotate' >/dev/null nothing happens. The cdr-records does not get rotated. Can anyone help me on why this is happening and how I can rotate the files as root?

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xmpp outbound caller ID
by Robert 12 Feb '12

12 Feb '12

I just setup FS to be able to make phone calls through Gtalk and I was wondering if there is a way to set the outbound caller ID? I've tried various forms of the following the dialplan effective_caller_id_number=${###} On the receiving phone the Google number still shows up.

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setting higher ptime with sangoma transcoding card
by Arafath-uz-zaman khan 12 Feb '12

12 Feb '12

Hello All I am using FS with Sangoma transcoding card. Can anyone suggest me how can i user 60 or more ptime for G729 codec? -- Arafath-uz-zaman khan

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How to inspect the jitterbuffer?
by Holger Freyther 11 Feb '12

11 Feb '12

Hi all, I have a setup with a FreeTDM/wanpipe trunk and most of the time the audio skips on the path to this trunk (e.g. after a period of silence, RTP marker bit set by the sender). It is not an issue with FreeTDM/wanpipe but more the way the RTP input is handled by FreeSWITCH. My theory is the audio skips because the RTP 'queue' has an under-run (while it shouldn't, there is no packet loss). So how can I see if the jitterbuffer is enabled for a given call (or globally)? Can I somehow see under-runs? The recovered clock/ clock skew? I couldn't find any of this in the wiki[1]. thanks holger [1] http://wiki.freeswitch.org/wiki/Jitterbuffer

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Skype module killing
by Balamurugan Mahendran 11 Feb '12

11 Feb '12

Skype module crashed the FS freeswitch@internal> load mod_skypopen +OK Reloading XML -ERR [module load file routine returned an error] 2012-02-11 14:18:55.142045 [INFO] mod_pocketsphinx.c:482 PocketSphinx Reloaded freeswitch@internal> 2012-02-11 14:18:55.142045 [INFO] mod_enum.c:812 ENUM Reloaded 2012-02-11 14:18:55.142045 [INFO] switch_time.c:1035 Timezone reloaded 530 definitions 2012-02-11 14:18:55.162036 [WARNING] mod_skypopen.c:1738 [406c21c|1b4c78b] [WARNINGA 1738 ][skype101 ][IDLE,IDLE] STARTING interface_id=1 2012-02-11 14:18:55.162036 [ERR] skypopen_protocol.c:1549 [406c21c|1b4c78b] [ERRORA 1549 ][none ][N/A,N/A] Received error code 5 from X Server 2012-02-11 14:18:55.162036 [ERR] skypopen_protocol.c:1550 [406c21c|1b4c78b] [ERRORA 1550 ][none ][N/A,N/A] Display error for 1, :101 2012-02-11 14:18:55.162036 [ERR] skypopen_protocol.c:1747 [406c21c|1b4c78b] [ERRORA 1747 ][skype101 ][DEAD,IDLE] Fatal display error for :101 - successed to jump 2012-02-11 14:18:55.162036 [WARNING] skypopen_protocol.c:1772 [406c21c|1b4c78b] [WARNINGA 1772 ][skype101 ][DEAD,IDLE] Removing skype interface #skype101 2012-02-11 14:18:55.362032 [NOTICE] mod_skypopen.c:1768 [406c21c|1b4c78b] [NOTICA 1768 ][skype101 ][DEAD,IDLE] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2012-02-11 14:19:05.362034 [ERR] mod_skypopen.c:1781 [406c21c|1b4c78b] [ERRORA 1781 ][skype101 ][DEAD,IDLE] Failed to connect to a SKYPE API for interface_id=1, no SKYPE client running, please (re)start Skype client. Skypopen exiting 2012-02-11 14:19:05.362034 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_skypopen.so **Module load routine returned an error** freeswitch@internal> load mod_skypopen Socket interrupted, bye! 5 XSELINUXs still allocated at reset SCREEN: 0 objects of 168 bytes = 0 total bytes 0 private allocs DEVICE: 4 objects of 48 bytes = 192 total bytes 0 private allocs CLIENT: 0 objects of 160 bytes = 0 total bytes 0 private allocs WINDOW: 0 objects of 32 bytes = 0 total bytes 0 private allocs PIXMAP: 1 objects of 16 bytes = 16 total bytes 0 private allocs GC: 0 objects of 56 bytes = 0 total bytes 0 private allocs CURSOR: 0 objects of 8 bytes = 0 total bytes 0 private allocs CURSOR_BITS: 0 objects of 8 bytes = 0 total bytes 0 private allocs DBE_WINDOW: 0 objects of 24 bytes = 0 total bytes 0 private allocs TOTAL: 5 objects, 208 bytes, 0 allocs 4 DEVICEs still allocated at reset DEVICE: 4 objects of 48 bytes = 192 total bytes 0 private allocs CLIENT: 0 objects of 160 bytes = 0 total bytes 0 private allocs WINDOW: 0 objects of 32 bytes = 0 total bytes 0 private allocs PIXMAP: 1 objects of 16 bytes = 16 total bytes 0 private allocs GC: 0 objects of 56 bytes = 0 total bytes 0 private allocs CURSOR: 0 objects of 8 bytes = 0 total bytes 0 private allocs CURSOR_BITS: 0 objects of 8 bytes = 0 total bytes 0 private allocs DBE_WINDOW: 0 objects of 24 bytes = 0 total bytes 0 private allocs TOTAL: 5 objects, 208 bytes, 0 allocs 1 PIXMAPs still allocated at reset PIXMAP: 1 objects of 16 bytes = 16 total bytes 0 private allocs GC: 0 objects of 56 bytes = 0 total bytes 0 private allocs CURSOR: 0 objects of 8 bytes = 0 total bytes 0 private allocs CURSOR_BITS: 0 objects of 8 bytes = 0 total bytes 0 private allocs DBE_WINDOW: 0 objects of 24 bytes = 0 total bytes 0 private allocs TOTAL: 1 objects, 16 bytes, 0 allocs Thanks, Bala

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