Re: [Freeswitch-users] What's problem in SVN ?
Dome Charoenyost d...@tel.co.th wrote: What's problem in SVN ? Not thing update after 23/12/2009 (16055) Surely the FreeSWITCH developers are entitled to spend time with their families/friends after a highly productive year of work. Note that there are holidays in many countries at this time of year. I would like to wish everyone involved in the FreeSWITCH project a pleasant and refreshing holiday, and much success in 2010. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] What's problem in SVN ?
Oh...sory i forgot chismas and new year. if someone come to thailand please let's me know :) BG Dome C. 2009/12/28 Jason White ja...@jasonjgw.net: Dome Charoenyost d...@tel.co.th wrote: What's problem in SVN ? Not thing update after 23/12/2009 (16055) Surely the FreeSWITCH developers are entitled to spend time with their families/friends after a highly productive year of work. Note that there are holidays in many countries at this time of year. I would like to wish everyone involved in the FreeSWITCH project a pleasant and refreshing holiday, and much success in 2010. ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] No audio after Remote SDP:
If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. I would have to see what happens after the 401 to see if it really did fail. /b On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: This is all I see and then registration fails. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] SNOM shared lines with TLS problems?
Shared will require some testing with TLS. I need traces, console logs and you to do some foot work to see if you can provide more details. /b On Dec 24, 2009, at 8:35 AM, Yehavi Bourvine wrote: Hello, Is there anyone who is using SNOM with TLS encryption and shared lines and it works? We have 1.0.5pre9 connected to SNOM-820 with shared lines between 2-3 SNOM phones. The TLS is defined by adding transport=tls to the registrar field (proxy is left blank). We noticed the following behaviour: With non-shared line UDP and TLS both work ok. With shared lines UDP works ok. with shared line TLS works as long as only one phone is registered. After the second TLS shared line registers we get busy for this extension. From the SNOM trace there is no incoming call attempt at all from FreeSwitch. Anyone has this setup working and can share some tips? Thanks, __Yehavi: ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events!
most likely cause would be connecting a socket then not regularly reading from it causing the buffer to fill up. any event socket connection must select on the socket and do regular read attempts or all the events will accumulate on the server side until some sanity check is reached and it begins to throw them away, the fist time there is room in this buffer again (when you consume some from the socket leaving space in the queue) it will report how many have been lost since the last read. One way to cause this would be suspend fs_cli with ctl-z and bring it back to the foreground after some time. On Thu, Dec 24, 2009 at 7:05 AM, Nicolas Brenner nico...@medularis.comwrote: I just got into the fs cli and when I ran a 'show calls' I got the following message: 2009-12-24 09:58:20.058365 [CRIT] mod_event_socket.c:337 Lost 8456 events! What does this mean? does it mean the event_socket did not report 8456 events? Why could this happen? The answer to this is pretty critical to me, as I make and monitor calls through the socket. Thanks for your help! Nicolas ___ FreeSWITCH-users mailing list 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:+19193869900 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Adding H263 Video to Existing CallFailsFirst Time
Okay. I uncommented the following lines and the video start works as correctly: param name=media-option value=bypass-media-after-att-xfer/ param name=inbound-bypass-media value=true/ Thanks, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Tuesday, December 22, 2009 8:33 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing CallFailsFirst Time No. The following lines is commented out (internal.xml): !--param name=media-option value=bypass-media-after-att-xfer/-- !--param name=inbound-bypass-media value=true/-- Thanks, Jerry -Original Message- From: Peter P GMX [mailto:prometheus...@gmx.net] Sent: Tuesday, December 22, 2009 3:21 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events!
Anthony, thank you very much for your response. The daemon that was reading the events froze, so apparently that was the source of the problem and your explanation fits perfectly. On Mon, Dec 28, 2009 at 12:47 PM, Anthony Minessale anthony.miness...@gmail.com wrote: most likely cause would be connecting a socket then not regularly reading from it causing the buffer to fill up. any event socket connection must select on the socket and do regular read attempts or all the events will accumulate on the server side until some sanity check is reached and it begins to throw them away, the fist time there is room in this buffer again (when you consume some from the socket leaving space in the queue) it will report how many have been lost since the last read. One way to cause this would be suspend fs_cli with ctl-z and bring it back to the foreground after some time. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] What's problem in SVN ?
The issues you ran into are probably sorted out now. Give it a try and if its still not working, post the build errors. Mike On Dec 28, 2009, at 8:15 AM, Dome Charoenyost wrote: Oh...sory i forgot chismas and new year. if someone come to thailand please let's me know :) BG Dome C. 2009/12/28 Jason White ja...@jasonjgw.net: Dome Charoenyost d...@tel.co.th wrote: What's problem in SVN ? Not thing update after 23/12/2009 (16055) Surely the FreeSWITCH developers are entitled to spend time with their families/friends after a highly productive year of work. Note that there are holidays in many countries at this time of year. I would like to wish everyone involved in the FreeSWITCH project a pleasant and refreshing holiday, and much success in 2010. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] sound rpms
This is a total work in progress that has not even merged into tree. So it is not known to work or not work anywhere. Patches to correct issues are welcome. Mike, I took another look at this and don't really know enough about rpm building to diagnose this. Frankly, the format of the latest spec is so wildly different from anything I have ever touched I am at a loss:) Is there a simple manual way for me to properly get the sounds for MOH etc installed? is it acceptable to simply run the buildsounds-callie.sh script with the sounds_location pointed to my /opt/freeswitch/sounds directory? Thanks! jlc ___ FreeSWITCH-users mailing list 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
[Freeswitch-users] Presence Change Distribution
Is there a setting to control how fast FS distributes presence changes to subscribers? Currently, it appears to take several minutes before I see presence changes. I would like to see them almost instantaneously, if possible. Thanks and Best Regards, Jerry ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] sound rpms
the build system already has targets for all of this and there are tarballs you can manually download and extract as well that are located in http://files.freeswitch.org/. if you NEED packages, you will have to wait until that work is complete or figure out what the error is. Mike On Dec 28, 2009, at 2:54 PM, Joseph L. Casale wrote: This is a total work in progress that has not even merged into tree. So it is not known to work or not work anywhere. Patches to correct issues are welcome. Mike, I took another look at this and don't really know enough about rpm building to diagnose this. Frankly, the format of the latest spec is so wildly different from anything I have ever touched I am at a loss:) Is there a simple manual way for me to properly get the sounds for MOH etc installed? is it acceptable to simply run the buildsounds-callie.sh script with the sounds_location pointed to my /opt/freeswitch/sounds directory? Thanks! jlc ___ FreeSWITCH-users mailing list 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
[Freeswitch-users] twitter.com/freeswitch (its not ours)
Dear FreeSWITCHers, Someone has registered the freeswitch name and is squatting on twitter with it. They haven't used it in over a year and I would like to have this for our project as its clearly confusing. If you own this account please contact me off list. Thanks, Brian ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4
I'm still not done with this I think we found a bug in the lib... Viktor fixed it today and I'm going to retry after I get done testing G729 more today! ;) /b On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote: Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn trunk, I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone client, however I am not seeing the enrollment option popup in zfone 0.92 build 218 on windows in front of an x-lite client. Any suggestions on what I should look at to troubleshoot this? I am waiting for the Tivi folks to send a 2.0.7 beta for windows mobile, but until then ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] forcing ptime settings
On Sun, Dec 27, 2009 at 5:01 AM, Karl J. Vesterling k...@ken-ton.com wrote: Setting the codec negotiation to scrooge resolved my problems w/ CallCentric. I'd bet that'd do it for him as well. *Lessons Learned by me:* 1.) Listen to Brian. 2.) When in doubt, refer to rule 1. Can I get that framed? :) -MC ___ FreeSWITCH-users mailing list 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
[Freeswitch-users] problems getting openzap compiled for use with freeswitch
I am following the wiki page here
http://wiki.freeswitch.org/wiki/OpenZAP#Zaptel_Installation
setup is on Ubuntu 9.0.4
using the debian method, when I run
module-assistant build zaptel-source
the compilation fails as below, not sure what I am missing, reading further
FS wiki pages and googling haven't enlightened me, any suggestions? I am
trying to get 2x X100P's working
dh_testdir
dh_testroot
rm -f *-stamp
# Delete the generated bristuff symlinks:
rm -f -f cwain.[ch] qozap.[ch] zaphfc.[ch] ztgsm.[ch]
# Add here commands to clean up after the build process.
rm -rf modexamples
rm -f tonezones.txt
rm -f version.h
rm -rf debian/fake
# * Makefile does not exist when running svn-buildpackage
# as the source tree is not there.
# FIXME: This will fail with an ugly warning on the clean of the
# modules build. However only fter the actuual clean.
[ ! -f Makefile ] || /usr/bin/make dist-clean || true
make[1]: Entering directory `/usr/src/modules/zaptel'
make: Entering an unknown directory
make: Leaving an unknown directory
rm -f torisatool
rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest
ztscan
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f libtonezone.so libtonezone.a *.lo
/usr/bin/make -C /usr/src/linux ARCH=i386
SUBDIRS=/usr/src/modules/zaptel/kernel HOTPLUG_FIRMWARE=yes
KBUILD_OBJ_M=pciradio.o tor
make[2]: Entering directory `/usr/src/linux-headers-2.6.28-11-server'
CLEAN /usr/src/modules/zaptel/kernel
CLEAN /usr/src/modules/zaptel/kernel/.tmp_versions
make[2]: Leaving directory `/usr/src/linux-headers-2.6.28-11-server'
make[2]: Entering directory `/usr/src/modules/zaptel/kernel/xpp/utils'
rm -f *.o init_fxo_modes print_modes perlcheck zt_registration.8 xpp_sync.8
lszaptel.8 xpp_blink.8 zapconf.8 zaptel_hardware.8
make[2]: Leaving directory `/usr/src/modules/zaptel/kernel/xpp/utils'
make: Entering an unknown directory
make: Leaving an unknown directory
make[1]: Leaving directory `/usr/src/modules/zaptel'
#rm -f debian/manpage.links debian/manpage.refs debian/*.8
dh_clean
/usr/bin/make -f debian/rules kdist_clean kdist_config binary-modules
make[1]: Entering directory `/usr/src/modules/zaptel'
dh_testdir
dh_testroot
rm -f *-stamp
# Delete the generated bristuff symlinks:
rm -f -f cwain.[ch] qozap.[ch] zaphfc.[ch] ztgsm.[ch]
# Add here commands to clean up after the build process.
rm -rf modexamples
rm -f tonezones.txt
rm -f version.h
rm -rf debian/fake
# * Makefile does not exist when running svn-buildpackage
# as the source tree is not there.
# FIXME: This will fail with an ugly warning on the clean of the
# modules build. However only fter the actuual clean.
[ ! -f Makefile ] || /usr/bin/make dist-clean || true
make[2]: Entering directory `/usr/src/modules/zaptel'
make: Entering an unknown directory
make: *** menuselect: No such file or directory. Stop.
make: Leaving an unknown directory
make[2]: [clean] Error 2 (ignored)
rm -f torisatool
rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest
ztscan
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f libtonezone.so libtonezone.a *.lo
/usr/bin/make -C /usr/src/linux ARCH=i386
SUBDIRS=/usr/src/modules/zaptel/kernel HOTPLUG_FIRMWARE=yes
KBUILD_OBJ_M=pciradio.o tor
make[3]: Entering directory `/usr/src/linux-headers-2.6.28-11-server'
make[3]: Leaving directory `/usr/src/linux-headers-2.6.28-11-server'
make[3]: Entering directory `/usr/src/modules/zaptel/kernel/xpp/utils'
rm -f *.o init_fxo_modes print_modes perlcheck zt_registration.8 xpp_sync.8
lszaptel.8 xpp_blink.8 zapconf.8 zaptel_hardware.8
make[3]: Leaving directory `/usr/src/modules/zaptel/kernel/xpp/utils'
make: Entering an unknown directory
make: *** ppp: No such file or directory. Stop.
make: Leaving an unknown directory
make[2]: *** [clean] Error 2
make[2]: Leaving directory `/usr/src/modules/zaptel'
#rm -f debian/manpage.links debian/manpage.refs debian/*.8
dh_clean
for templ in ; do \
cp $templ `echo $templ | sed -e 's/_KVERS_/2.6.28-11-server/g'` ; \
done
for templ in `ls debian/*.modules.in` ; do \
test -e ${templ%.modules.in}.backup || cp ${templ%.modules.in}
${templ%.modules.in}.backup 2/dev/null || true; \
sed -e 's/##KVERS##/2.6.28-11-server/g ;s/#KVERS#/2.6.28-11-server/g ;
s/_KVERS_/2.6.28-11-server/g ; s/##KDREV##/2.6.28-11.42
done
dh_testdir
dh_testroot
dh_clean -k
cp -a /usr/src/modules/zaptel/debian/generated/* .
./configure
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... gcc -E
checking for a BSD-compatible install... /usr/bin/install -c
checking whether ln -s works... yes
checking for GNU make... make
[Freeswitch-users] Personal Greeting
Hi I am new to Freeswitch so my question may be a stupid question. I just want to know how to disable the personal greeting to the default one. One user has recorded his personal greeting now how can he make this default. I could not find any option for the same. Plz advice. Thanks regards Sharad garg -- View this message in context: http://n2.nabble.com/Personal-Greeting-tp4226681p4226681.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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
[Freeswitch-users] Message Wait Lamp for an unregistered user
Hi, I am trying to integrate Freeswitch as a Media Server with our own SIP Server. For this, initially, I am using freeswitch for Auto Attendant Voicemail Application. In this case, all the SIP users are registered with my SIP Server whenever Auto Att or Voicemail application is required, my SIP Server just forward the call to Freeswitch. Everything seems ok working fine except some small small points. One of the point is - whenever there is a voice message in the mailbox of a user, Freeswitch is not generating the MWI (Notify) to my SIP Server. So just want to know , is there any way so that freeswitch can light up / light -off the message wait lamp for the users of my SIP Server. Thanks in advance for your answers. Regards Sharad -- View this message in context: http://n2.nabble.com/Message-Wait-Lamp-for-an-unregistered-user-tp4226726p4226726.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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
[Freeswitch-users] Hold is broken in trunk 16055
Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. sip trace for trunk 16055 re-invite request sent to fs when client hold the line INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c6494 To: sip:65960...@10.56.0.189:5060;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ==sip trace for fs 1.0.4 =re-invite request sent to FS when client want to hold the all INVITE sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: sip:65960...@10.56.0.189:5060;transport=udp User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 254 v=0 o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 ACK sip:65960...@10.56.0.189:5060;transport=udp SIP/2.0 Contact: sip:37186004...@10.56.90.223 sip%3a37186004...@10.56.90.223 From: sip:37186004...@10.56.0.189:5060;tag=1c8147 To: sip:65960...@10.56.0.189:5060;tag=tH78Sc30vXKXK
