May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Facebook...
by Klaus Hochlehnert 27 May '12

27 May '12

Hi Guys, just saw that you already have a Facebook entry. Wouldn't it be possible to connect the news/twitter feeds somehow with Facebook? I'd really like to see all news on Facebook... Cheers, Klaus

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Re: [Freeswitch-users] RTP media issue
by Mr Nathan Downes 27 May '12

27 May '12

Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using <param name="auto-jitterbuffer-msec" value="60"/>, is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added <action application="set" data="sip_jitter_buffer_during_bridge=true" /> because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of <extension name="02-xxxx-9365" > <condition field="context" expression="public"/> <condition field="destination_number" expression="^02xxxx9365;.*$"> <action application="sleep" data="3000"/> <action application="set" data="domain=mgst.xxxxxxxx.com.au"/> <action application="set" data="domain_name=${domain}"/> <action application="set" data="call_direction=inbound"/> <action application="transfer" data="117 XML mgst.xxxxxxxxx.com.au"/> </condition> </extension> It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. <action application="cng_plc"/> <action application="set" data="jitterbuffer_msec=60"/> in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes <nathandownes(a)hotmail.com> wrote: > Hi, > > enable-soa > > <param name="enable-soa" value="true"/> > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message.. Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter(a)gmail.com > To: freeswitch-users(a)lists.freeswitch.org > CC: nathan(a)nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > <param name="enable-soa" value="false"/> > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes <nathandownes(a)hotmail.com> wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound. The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it. In the corresponding outbound trace, between > packet 8144 and 8152, it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152. This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible. This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only. The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > 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-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > 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-use > rs > 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_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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

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Re: [Freeswitch-users] RTP media issue
by Mr Nathan Downes 26 May '12

26 May '12

Hi Peter, Is it possible using pcapsipdump is causing it to look like that? I will be onsite again tomorrow and I will do a few tshark captures on both sides, I just have to start and stop it after each call, as they get huge if I leave it running. I can also get the capture from the provider, and ones from the near end and far end of the fibre network. I did find where to put cng_plc in public.xml as an absolute condition so it runs on each inbound call, the same with that jitterbuffer statement. Should I take them out just for this tracing? Thank you for your help. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Saturday, 26 May 2012 10:42 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue I think the first step is to check the actual signalling. Just as Anthony mentions, the call setup looks weird. Between FS and the ATA (if I understood the different IP's/peers correctly) it seems FS sends early media (183), and it takes the ATA 10 seconds to ACK this. During the time the call is answered on the other end, which dosn't seem to be passed to the ATA (probably because it's so late with the ACK - I'm not sure about that though). So, first check this - it seems to me that FS might still be in "early media" state during the entire call. About the actual RTP, I see one problem here, it's packet 8146 in outbound.pcap. This is a packet that is probably generated by FS (since by that exact time, you're missing one packet from the provider), the problem here is that the timestamp if way off (I'm not sure where FS gets this ts from). It's because of this faulty timestamp that a few packets after this is dropped, since FS won't send packets with a lower timestamp than before. This is also why packet 8152 seems to have a strange timestamp, but it's actually just passed from the other leg, and is the first timestamp that is greater then packet 8146. One possible solution would be to enable rewrite of timestamps (rtp-rewrite-timestamps in the sofia config), also mentioned here: http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio However, I think you also need to check the strangness in the SIP signalling. /Peter ________________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] för Mr Nathan Downes [nathandownes(a)hotmail.com] Skickat: den 26 maj 2012 00:03 Till: 'FreeSWITCH Users Help' Ämne: Re: [Freeswitch-users] RTP media issue Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using <param name="auto-jitterbuffer-msec" value="60"/>, is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added <action application="set" data="sip_jitter_buffer_during_bridge=true" /> because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of <extension name="02-xxxx-9365" > <condition field="context" expression="public"/> <condition field="destination_number" expression="^02xxxx9365;.*$"> <action application="sleep" data="3000"/> <action application="set" data="domain=mgst.xxxxxxxx.com.au"/> <action application="set" data="domain_name=${domain}"/> <action application="set" data="call_direction=inbound"/> <action application="transfer" data="117 XML mgst.xxxxxxxxx.com.au"/> </condition> </extension> It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. <action application="cng_plc"/> <action application="set" data="jitterbuffer_msec=60"/> in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes <nathandownes(a)hotmail.com> wrote: > Hi, > > enable-soa > > <param name="enable-soa" value="true"/> > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message.. Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter(a)gmail.com > To: freeswitch-users(a)lists.freeswitch.org > CC: nathan(a)nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > <param name="enable-soa" value="false"/> > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes <nathandownes(a)hotmail.com> wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound. The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it. In the corresponding outbound trace, between > packet 8144 and 8152, it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152. This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible. This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only. The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > 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-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > 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-use > rs > 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_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 !DSPAM:4fbfffef32764666710318! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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

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Common/Reasonable Assumption on DID/Channel over-subscription
by A E [Gmail] 26 May '12

26 May '12

Hello All, just throwing this out there. What are people generally using these days when designing their services, esp. those that require a user to call a DID to access their system, similar to calling card services. There was a time when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of channels bought in SMB with IP-PBX. I believe this would have changed today and assuming a service is pretty popular, the ALOCs are longer due to cheaper rates and convenience of calling. Does anyone have any real world numbers they can share? Is 10 to 1 a good ratio to ensure a user practically never gets a "circuits are busy"? Thanks in advance

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sofia current calls for a user
by babak yakhchali 26 May '12

26 May '12

Hi Is there any way to use sofia apis to find out if currently there is a call from or to a user? reading sofia code shows that its using sip_dialogs table to track user presence and call data but there is no api call such as (sofia ...) which shows that data. thanx

3 5

libtool on Mac OSX Lion
by Neil Patel 26 May '12

26 May '12

Hi All, another OSX Lion build issue: On bootstrap I'm getting this output: anitya:freeswitch neil$ ./bootstrap.sh bootstrap: checking installation... bootstrap: autoconf version 2.59 (ok) bootstrap: automake version 1.7.1 (ok) bootstrap: aclocal version 1.7.1 (ok) bootstrap: libtool version 1.5.24 (ok) Bootstrapping using: autoconf : /usr/local/bin/autoconf automake : /usr/local/bin/automake aclocal : /usr/local/bin/aclocal libtool : /usr/bin/glibtool (1.5.24.) libtoolize: /usr/bin/glibtoolize make : /usr/bin/make (GNU Make 3.81) awk : () Entering directory /usr/local/src/freeswitch/libs/apr Copying libtool helper files ... Using `AC_PROG_RANLIB' is rendered obsolete by `AC_PROG_LIBTOOL' ... Remember to add `AC_PROG_LIBTOOL' to `configure.ac'. You should add the contents of `/usr/share/aclocal/libtool.m4' to `aclocal.m4'. Creating configure bootstrap: autoconf configure:6475: error: possibly undefined macro: AC_PROG_LIBTOOL If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. ... I believe this is behind a build error I'm currently up against on mod_enum: making all mod_enum checking for i386-apple-darwin11.3.0-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 ANSI C... none needed checking how to run the C preprocessor... gcc -E checking for egrep... grep -E checking for AIX... no rm: conftest.dSYM: is a directory checking for i386-apple-darwin11.3.0-gcc... (cached) gcc checking whether we are using the GNU C compiler... (cached) yes checking whether gcc accepts -g... (cached) yes checking for gcc option to accept ANSI C... (cached) none needed checking whether make sets $(MAKE)... yes checking whether gcc supports -std=c99... yes checking whether gcc supports -xc99... no checking for an ANSI C-conforming const... yes checking whether gcc supports -g... yes checking whether gcc supports -O2... yes checking whether gcc supports -Wall... yes checking whether gcc supports -W... yes checking whether gcc supports -Wwrite-strings... yes checking for ANSI C header files... rm: conftest.dSYM: is a directory rm: conftest.dSYM: is a directory yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for getopt.h... yes checking for time.h... yes checking for winsock2.h... no checking for ws2tcpip.h... no checking whether gcc supports -Werror... yes checking whether gcc supports -Wall... (cached) yes checking whether gcc supports -std=c99... (cached) yes checking whether gcc supports -xc99... (cached) no checking for getopt.h... (cached) yes checking for time.h... (cached) yes checking whether we need -std=c99 -D__EXTENSIONS__ -D_BSD_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -D_XOPEN_SOURCE_EXTENDED=1 -D_ALL_SOURCE as a flag for gcc... no checking whether we need -std=c99 -D__EXTENSIONS__ -D_BSD_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -D_ALL_SOURCE as a flag for gcc... no checking whether we need -std=c99 as a flag for gcc... no checking whether we need -D_BSD_SOURCE as a flag for gcc... no checking whether we need -D_GNU_SOURCE as a flag for gcc... no checking whether we need -D_GNU_SOURCE -D_FRSRESGID as a flag for gcc... failed checking whether we need -D_POSIX_C_SOURCE=200112 as a flag for gcc... no checking whether we need -D__EXTENSIONS__ as a flag for gcc... no checking for inline... inline checking for int8_t... yes checking for int16_t... yes checking for int32_t... yes checking for int64_t... yes checking for uint8_t... yes checking for uint16_t... yes checking for uint32_t... yes checking for uint64_t... yes checking for doxygen... no checking for library containing socket... none required checking for library containing inet_pton... none required checking build system type... i386-apple-darwin11.3.0 checking host system type... i386-apple-darwin11.3.0 checking for i386-apple-darwin11.3.0-ar... no checking for ar... /usr/bin/ar /usr/local/src/freeswitch/libs/ldns/configure: line 6475: AC_PROG_LIBTOOL: command not found checking for openssl/ssl.h... yes checking for openssl/err.h... yes checking for openssl/rand.h... yes checking for EVP_sha256... no checking whether byte ordering is bigendian... no checking for ANSI C header files... (cached) yes checking for getopt.h... (cached) yes checking for stdarg.h... yes checking for stdbool.h... yes checking for openssl/ssl.h... (cached) yes checking for netinet/in.h... yes checking for time.h... (cached) yes checking for arpa/inet.h... yes checking for netdb.h... yes checking for sys/param.h... yes checking for sys/mount.h... yes checking for sys/socket.h... yes checking for inttypes.h... (cached) yes checking for sys/types.h... (cached) yes checking for unistd.h... (cached) yes checking for socklen_t... yes checking for ssize_t... yes checking for in_addr_t... yes checking for in_port_t... yes checking for struct sockaddr_storage.ss_family... yes checking for stdlib.h... (cached) yes checking for GNU libc compatible malloc... yes checking for stdlib.h... (cached) yes checking for GNU libc compatible realloc... yes checking for b64_pton... no checking for b64_ntop... no checking for b32_pton... no checking for b32_ntop... no checking for timegm... yes checking for gmtime_r... yes checking for ctime_r... yes checking for isblank... yes checking for isascii... yes checking for inet_aton... yes checking for inet_pton... yes checking for inet_ntop... yes checking for snprintf... yes checking for strlcpy... yes checking for memmove... yes checking for endprotoent... yes checking for endservent... yes checking for sleep... yes checking for random... yes checking for fcntl... yes checking for strtoul... yes checking for getaddrinfo... yes checking for ioctlsocket... no checking whether the C compiler (gcc) accepts the "format" attribute... yes checking whether the C compiler (gcc) accepts the "unused" attribute... yes configure: creating ./config.status config.status: creating Makefile config.status: creating ldns/net.h config.status: creating ldns/util.h config.status: creating packaging/libldns.pc config.status: creating packaging/ldns-config config.status: creating ldns/config.h copying header files make[6]: ./libtool: No such file or directory make[6]: *** [rdata.o] Error 1 make[5]: *** [/usr/local/src/freeswitch/libs/ldns/libldns.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_enum-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 I don't remember exactly how I installed libtool, but it was a script I ran from within FS that automatically wget'd, compiled, and installed it (suggested by wiki or mailing list). It definitely wasn't through macports/fink/brew. Any suggestions on what the problem may be? Thanks!

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Unable to build Git tree on Mac OSX Lion
by Neil Patel 26 May '12

26 May '12

Hi All, I am getting this error when trying to build latest git on Mac OSX Lion, after fresh checkout and running ./boostraph.sh and ./configure: making all mod_amr Creating mod_amr.so... i686-apple-darwin11-llvm-gcc-4.2: -bundle not allowed with -dynamiclib gcc -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DAMR_PASSTHROUGH -shared -o .libs/mod_amr.so -dynamic -bundle -force-flat-namespace .libs/mod_amr.o /usr/local/src/freeswitch/.libs/libfreeswitch.dylib -lodbc -lresolv -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -lpq /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lpthread -lm -L/usr/local/src/freeswitch/libs/srtp -ldl -lssl -lcrypto -lz -lncurses /usr/local/lib/libjpeg.dylib /usr/local/lib/libodbc.dylib -liconv make[5]: *** [mod_amr.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_amr-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Heeding Mario's advice, I tried making with open-ssl flag, but that didn't help. If I go about commenting out offending modules, the same "bundle not allowed with -dynamiclib" appears for any number of other mods. I have installed all the prerequisites for Mac OSX (to my knowledge), including Apple and Unix dev tools. I am 100% macports/fink/brew free, AFAIK. Jira filed, but no response for a while: http://jira.freeswitch.org/browse/FS-4240 My sense this is a missing tool or config issue with my setup (pretty much fresh Lion install). Thanks in advance for your help! Neil

3 5

Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up
by Sherif Omran 25 May '12

25 May '12

Hello guys, here is the trace (see the attachment) kind regards Sherif --- On Fri, 5/25/12, SamyGo <govoiper(a)gmail.com> wrote: From: SamyGo <govoiper(a)gmail.com> Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Friday, May 25, 2012, 10:46 AM Hi again,If you want kamailio register to the provider then use UAC module. Kamailio will use the username/password and register with the provider. Regards,Sammy On Fri, May 25, 2012 at 12:20 PM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: Dear Sammy, Thank you for your question ... Yes, the GW is registered through FS because I did not know how to register it to kamailio. But it seems better to register it to kamailio. One more information, calls from 1001 to 1002 go to kamailio then to FS then return back to kamailio smoothly. Thus I would suggest that I change registering the gateway from FS to kamailio. but How to? --- On Fri, 5/25/12, SamyGo <govoiper(a)gmail.com> wrote: From: SamyGo <govoiper(a)gmail.com> Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Friday, May 25, 2012, 8:08 AM Hi,These are SIP traces on FreeSWITCH console, whereas you are saying and it seems that kamailio is not detecting the answering (200 OK)of the call from extension 1002. Please, can you take a sip trace..!! I see you've both kamailio and FS on same server! Please take a pcap from the linux console using the following command. #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv Please be quick on turning the sip trace on and off as quickly as possible to avoid extra packets. Once done open the file in wireshark ; apply filter "sip || rtp" and then save the resulting capture in separate file. Send us the new file to analyse. One more silly question probably, I see REGISTERs coming to your FS as well and the calls to gateways are made from FS too !!, umm...just thinking what are you using kamailio for!!? ThanksBR,Sammy On Fri, May 25, 2012 at 4:40 AM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: Hi all, here is the siptrace: To figure it out 1- gateway called bluesip.net. It send invide using caller number(a)bluesip.net 2- This call should go to extension kb-1002. kb means go from freeswitch port 6090 to kamailio port 5060 3- It should go to call extension 1002 in Kamailio 4- Extension 1002 rings but when I reply, it does not notice I replied ./fs_cli _____ ____ ____ _ ___ | ___/ ___| / ___| | |_ _| | |_ \___ \ | | | | | | | _| ___) | | |___| |___ | | |_| |____/ \____|_____|___| ******************************************************* * Anthony Minessale II, Ken Rice, * * Michael Jerris, Travis Cross * * FreeSWITCH (http://www.freeswitch.org) * * Paypal Donations Appreciated: paypal(a)freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/ * ******************************************************* Type /help <enter> to see a list of commands +OK log level [7] freeswitch@internal> tracelevel -ERR tracelevel Command not found! freeswitch@internal> sofia global siptrace on +OK Global siptrace on recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: ------------------------------------------------------------------------ INVITE sip:gw+trunk_1000@78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 Record-Route: <sip:217.74.179.29;ftag=as00589402;lr=on> Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 From: "+41793940965" <sip:+41793940965@bluesip.net>;tag=as00589402 To: <sip:+498972101087914@bluesip.net> Contact: <sip:+41793940965@217.74.179.28> Call-ID: 4253e4bd2d624caf7f18faf85390b68f(a)bluesip.net CSeq: 102 INVITE User-Agent: blueSIP PSTN GW Max-Forwards: 69 Date: Thu, 24 May 2012 23:08:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 367 P-hint: USRLOC v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 Record-Route: <sip:217.74.179.29;ftag=as00589402;lr=on> From: "+41793940965" <sip:+41793940965@bluesip.net>;tag=as00589402 To: <sip:+498972101087914@bluesip.net> Call-ID: 4253e4bd2d624caf7f18faf85390b68f(a)bluesip.net CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Content-Length: 0 ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/+41793940965(a)bluesip.net [69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965(a)bluesip.net entering state [received][100] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec sofia/internal/+41793940965(a)bluesip.net PCMU/8000 20 ms 160 samples 64000 bits 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/+41793940965(a)bluesip.net Original read codec set to PCMU:0 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/+41793940965(a)bluesip.net) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965(a)bluesip.net) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965(a)bluesip.net) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/+41793940965(a)bluesip.net SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/+41793940965(a)bluesip.net) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965(a)bluesip.net) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965(a)bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/+41793940965(a)bluesip.net) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965(a)bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965(a)bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965(a)bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context public Dialplan: sofia/internal/+41793940965(a)bluesip.net parsing [public->from_kamailio] continue=false Dialplan: sofia/internal/+41793940965(a)bluesip.net Regex (PASS) [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965(a)bluesip.net Action transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965(a)bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965(a)bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965(a)bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965(a)bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965(a)bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965(a)bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965(a)bluesip.net transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/+41793940965(a)bluesip.net) State Change CS_EXECUTE -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/+41793940965(a)bluesip.net to XML[kb-1002@default] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965(a)bluesip.net) State EXECUTE going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965(a)bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965(a)bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965(a)bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965(a)bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context default Dialplan: sofia/internal/+41793940965(a)bluesip.net parsing [default->vbox] continue=false Dialplan: sofia/internal/+41793940965(a)bluesip.net Regex (FAIL) [vbox] destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965(a)bluesip.net parsing [default->vmenu] continue=false Dialplan: sofia/internal/+41793940965(a)bluesip.net Regex (FAIL) [vmenu] destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965(a)bluesip.net parsing [default->kbridge] continue=false Dialplan: sofia/internal/+41793940965(a)bluesip.net Regex (PASS) [kbridge] destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965(a)bluesip.net Action set(proxy_media=true) Dialplan: sofia/internal/+41793940965(a)bluesip.net Action set(call_timeout=50) Dialplan: sofia/internal/+41793940965(a)bluesip.net Action set(continue_on_fail=true) Dialplan: sofia/internal/+41793940965(a)bluesip.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/+41793940965(a)bluesip.net Action set(sip_invite_domain=78.138.90.58) Dialplan: sofia/internal/+41793940965(a)bluesip.net Action export(sip_contact_user=ufs) Dialplan: sofia/internal/+41793940965(a)bluesip.net Action bridge(sofia/78.138.90.58/1002@78.138.90.58:5060) Dialplan: sofia/internal/+41793940965(a)bluesip.net Action answer() Dialplan: sofia/internal/+41793940965(a)bluesip.net Action voicemail(default ${domain_name} 1002) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965(a)bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965(a)bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965(a)bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965(a)bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965(a)bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965(a)bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965(a)bluesip.net set(proxy_media=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965(a)bluesip.net SET [proxy_media]=[true] EXECUTE sofia/internal/+41793940965(a)bluesip.net set(call_timeout=50) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965(a)bluesip.net SET [call_timeout]=[50] EXECUTE sofia/internal/+41793940965(a)bluesip.net set(continue_on_fail=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965(a)bluesip.net SET [continue_on_fail]=[true] EXECUTE sofia/internal/+41793940965(a)bluesip.net set(hangup_after_bridge=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965(a)bluesip.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/+41793940965(a)bluesip.net set(sip_invite_domain=78.138.90.58) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965(a)bluesip.net SET [sip_invite_domain]=[78.138.90.58] EXECUTE sofia/internal/+41793940965(a)bluesip.net export(sip_contact_user=ufs) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [sip_contact_user]=[ufs] EXECUTE sofia/internal/+41793940965(a)bluesip.net bridge(sofia/78.138.90.58/1002@78.138.90.58:5060) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/+41793940965(a)bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] to event 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002@78.138.90.58:5060 [69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/1002@78.138.90.58:5060) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002@78.138.90.58:5060) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002@78.138.90.58:5060) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/1002@78.138.90.58:5060 SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/1002@78.138.90.58:5060 Patched SDP --- v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 +++ v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: ------------------------------------------------------------------------ INVITE sip:1002@78.138.90.58:5060 SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK Max-Forwards: 67 From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58:5060> Call-ID: 412c8312-2098-1230-8985-00163e6bb553 CSeq: 28614398 INVITE Contact: <sip:ufs@78.138.90.58:6090> User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 372 P-hint: USRLOC X-FS-Support: update_display,send_info Remote-Party-ID: "+41793940965" <sip:+41793940965@78.138.90.58>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/1002@78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002@78.138.90.58:5060) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002@78.138.90.58:5060) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/1002@78.138.90.58:5060) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002@78.138.90.58:5060) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/1002@78.138.90.58:5060 SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002@78.138.90.58:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002@78.138.90.58:5060) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002@78.138.90.58:5060) Running State Change CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002@78.138.90.58:5060) State CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002@78.138.90.58:5060) State CONSUME_MEDIA going to sleep 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/1002@78.138.90.58:5060 entering state [calling][0] recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: ------------------------------------------------------------------------ SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58:5060> Call-ID: 412c8312-2098-1230-8985-00163e6bb553 CSeq: 28614398 INVITE Server: kamailio (3.1.5 (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK Record-Route: <sip:78.138.90.58;lr;nat=yes> Call-ID: 412c8312-2098-1230-8985-00163e6bb553 From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58>;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs CSeq: 28614398 INVITE Contact: "Mama" <sip:1002@192.168.178.1:6060;transport=UDP> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER Content-Length: 0 ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002@78.138.90.58:5060 entering state [proceeding][180] 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/1002@78.138.90.58:5060! send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 Record-Route: <sip:217.74.179.29;ftag=as00589402;lr=on> From: "+41793940965" <sip:+41793940965@bluesip.net>;tag=as00589402 To: <sip:+498972101087914@bluesip.net>;tag=S7UZQygFt62Nm Call-ID: 4253e4bd2d624caf7f18faf85390b68f(a)bluesip.net CSeq: 102 INVITE Contact: <sip:+498972101087914@78.138.90.58:6090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Remote-Party-ID: "Outbound Call" <sip:1002@bluesip.net>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/+41793940965(a)bluesip.net! 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965(a)bluesip.net entering state [early][180] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/+41793940965(a)bluesip.net! recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK Record-Route: <sip:78.138.90.58;lr=on;nat=yes> From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58:5060>;tag=549D2DD03BBA7C67 Call-ID: 412c8312-2098-1230-8985-00163e6bb553 CSeq: 28614398 INVITE Contact: <sip:1002@41.34.123.243;uniq=6AC0DF4D2E498C8ACA82CB22226B9> User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) Content-Length: 0 ------------------------------------------------------------------------ 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/1002@78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK Record-Route: <sip:78.138.90.58;lr;nat=yes> Call-ID: 412c8312-2098-1230-8985-00163e6bb553 From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58>;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs CSeq: 28614398 INVITE Contact: "Mama" <sip:1002@192.168.178.1:6060;transport=UDP> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER Content-Length: 0 ------------------------------------------------------------------------ 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002@78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK Record-Route: <sip:78.138.90.58;lr=on;nat=yes> From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58:5060>;tag=549D2DD03BBA7C67 Call-ID: 412c8312-2098-1230-8985-00163e6bb553 CSeq: 28614398 INVITE Contact: <sip:1002@41.34.123.243;uniq=6AC0DF4D2E498C8ACA82CB22226B9> User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) Content-Length: 0 ------------------------------------------------------------------------ 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/1002@78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK Record-Route: <sip:78.138.90.58;lr=on;nat=yes> From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58:5060>;tag=549D2DD03BBA7C67 Call-ID: 412c8312-2098-1230-8985-00163e6bb553 CSeq: 28614398 INVITE Contact: <sip:1002@41.34.123.243;uniq=6AC0DF4D2E498C8ACA82CB22226B9> User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) Content-Length: 0 ------------------------------------------------------------------------ 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/1002@78.138.90.58:5060 entering state [proceeding][180] recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: ------------------------------------------------------------------------ CANCEL sip:gw+trunk_1000@78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 From: "+41793940965" <sip:+41793940965@bluesip.net>;tag=as00589402 To: <sip:+498972101087914@bluesip.net> Call-ID: 4253e4bd2d624caf7f18faf85390b68f(a)bluesip.net CSeq: 102 CANCEL User-Agent: blueSIP PSTN GW Max-Forwards: 69 Content-Length: 0 P-hint: USRLOC ------------------------------------------------------------------------ send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 From: "+41793940965" <sip:+41793940965@bluesip.net>;tag=as00589402 To: <sip:+498972101087914@bluesip.net>;tag=S7UZQygFt62Nm Call-ID: 4253e4bd2d624caf7f18faf85390b68f(a)bluesip.net CSeq: 102 CANCEL Content-Length: 0 ------------------------------------------------------------------------ send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 From: "+41793940965" <sip:+41793940965@bluesip.net>;tag=as00589402 To: <sip:+498972101087914@bluesip.net>;tag=S7UZQygFt62Nm Call-ID: 4253e4bd2d624caf7f18faf85390b68f(a)bluesip.net CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965(a)bluesip.net entering state [terminated][487] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/+41793940965(a)bluesip.net) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/+41793940965(a)bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/+41793940965(a)bluesip.net [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/1002@78.138.90.58:5060) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1002@78.138.90.58:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1002@78.138.90.58:5060 [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002@78.138.90.58:5060) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002@78.138.90.58:5060) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002@78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/1002@78.138.90.58:5060 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002@78.138.90.58:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002@78.138.90.58:5060) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1002@78.138.90.58:5060) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002@78.138.90.58:5060) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002@78.138.90.58:5060) State REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002@78.138.90.58:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002@78.138.90.58:5060) State REPORTING going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1002@78.138.90.58:5060) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 (sofia/internal/1002@78.138.90.58:5060) Locked, Waiting on external entities 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002@78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 (sofia/internal/1002@78.138.90.58:5060) Ended 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1002@78.138.90.58:5060 [CS_DESTROY] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1002@78.138.90.58:5060) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002@78.138.90.58:5060) Running State Change CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002@78.138.90.58:5060) State DESTROY 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/1002@78.138.90.58:5060 SOFIA DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002@78.138.90.58:5060 Standard DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002@78.138.90.58:5060) State DESTROY going to sleep 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed. Cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 sofia/internal/+41793940965(a)bluesip.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965(a)bluesip.net) State EXECUTE going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965(a)bluesip.net) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965(a)bluesip.net) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/+41793940965(a)bluesip.net hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/+41793940965(a)bluesip.net Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965(a)bluesip.net) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/+41793940965(a)bluesip.net) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965(a)bluesip.net) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965(a)bluesip.net) State REPORTING send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: ------------------------------------------------------------------------ CANCEL sip:1002@78.138.90.58:5060 SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK Max-Forwards: 67 From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58:5060> Call-ID: 412c8312-2098-1230-8985-00163e6bb553 CSeq: 28614398 CANCEL Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 ------------------------------------------------------------------------ recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: ------------------------------------------------------------------------ ACK sip:gw+trunk_1000@78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 From: "+41793940965" <sip:+41793940965@bluesip.net>;tag=as00589402 Call-ID: 4253e4bd2d624caf7f18faf85390b68f(a)bluesip.net To: <sip:+498972101087914@bluesip.net>;tag=S7UZQygFt62Nm CSeq: 102 ACK User-Agent: Sip EXpress router(0.9.7 (i386/linux)) Content-Length: 0 ------------------------------------------------------------------------ 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 sofia/internal/+41793940965(a)bluesip.net Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965(a)bluesip.net) State REPORTING going to sleep 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/+41793940965(a)bluesip.net) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965(a)bluesip.net [BREAK] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 (sofia/internal/+41793940965(a)bluesip.net) Locked, Waiting on external entities 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/internal/+41793940965(a)bluesip.net) Ended 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/+41793940965(a)bluesip.net [CS_DESTROY] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/+41793940965(a)bluesip.net) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/+41793940965(a)bluesip.net) Running State Change CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965(a)bluesip.net) State DESTROY 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/+41793940965(a)bluesip.net SOFIA DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 sofia/internal/+41793940965(a)bluesip.net Standard DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965(a)bluesip.net) State DESTROY going to sleep recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: ------------------------------------------------------------------------ SIP/2.0 200 canceling Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b Call-ID: 412c8312-2098-1230-8985-00163e6bb553 CSeq: 28614398 CANCEL Server: kamailio (3.1.5 (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK Record-Route: <sip:78.138.90.58;lr;nat=yes> Call-ID: 412c8312-2098-1230-8985-00163e6bb553 From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58>;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs CSeq: 28614398 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER Content-Length: 0 ------------------------------------------------------------------------ send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: ------------------------------------------------------------------------ ACK sip:1002@78.138.90.58:5060 SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK Max-Forwards: 67 From: "+41793940965" <sip:+41793940965@78.138.90.58>;tag=tgNrSS1jQFS8F To: <sip:1002@78.138.90.58>;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs Call-ID: 412c8312-2098-1230-8985-00163e6bb553 CSeq: 28614398 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60926 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=USeHUmjpmrFUB Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60926 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60927 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=v279vF3SH15DQ Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60927 REGISTER Contact: <sip:1002@192.168.178.26:58881;ob>;expires=300 Date: Thu, 24 May 2012 23:13:13 GMT User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: ------------------------------------------------------------------------ NOTIFY sip:1002@192.168.178.26:58881;ob SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF Route: <sip:1002@41.34.123.243:58881>;ob Max-Forwards: 70 From: <sip:1002@78.138.90.58>;tag=XB12yamXeav0j To: <sip:1002@78.138.90.58> Call-ID: e0efa252-2098-1230-8985-00163e6bb553 CSeq: 28614532 NOTIFY Contact: <sip:mod_sofia@78.138.90.58:6090> User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:1002@78.138.90.58 Voice-Message: 1/0 (0/0) ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF Call-ID: e0efa252-2098-1230-8985-00163e6bb553 From: <sip:1002@78.138.90.58>;tag=XB12yamXeav0j To: <sip:1002@78.138.90.58>;tag=z9hG4bKBZ68g9yKg77FF CSeq: 28614532 NOTIFY Content-Length: 0 ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60928 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=ymtU0540BKjKe Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60928 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60929 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=ZXKm20N48U85S Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60929 REGISTER Contact: <sip:1002@192.168.178.26:58881;ob>;expires=300 Date: Thu, 24 May 2012 23:18:09 GMT User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: ------------------------------------------------------------------------ NOTIFY sip:1002@192.168.178.26:58881;ob SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a Route: <sip:1002@41.34.123.243:58881>;ob Max-Forwards: 70 From: <sip:1002@78.138.90.58>;tag=06cD4U6754yrN To: <sip:1002@78.138.90.58> Call-ID: 91100602-2099-1230-8985-00163e6bb553 CSeq: 28614680 NOTIFY Contact: <sip:mod_sofia@78.138.90.58:6090> User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:1002@78.138.90.58 Voice-Message: 1/0 (0/0) ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a Call-ID: 91100602-2099-1230-8985-00163e6bb553 From: <sip:1002@78.138.90.58>;tag=06cD4U6754yrN To: <sip:1002@78.138.90.58>;tag=z9hG4bKc8Z1j4FQDgy2a CSeq: 28614680 NOTIFY Content-Length: 0 ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60930 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=1F655pQB3DNBH Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60930 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60931 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=2rZy7H8e0pByc Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60931 REGISTER Contact: <sip:1002@192.168.178.26:58881;ob>;expires=300 Date: Thu, 24 May 2012 23:23:04 GMT User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: ------------------------------------------------------------------------ NOTIFY sip:1002@192.168.178.26:58881;ob SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp Route: <sip:1002@41.34.123.243:58881>;ob Max-Forwards: 70 From: <sip:1002@78.138.90.58>;tag=31rQ9cSjXZ1gr To: <sip:1002@78.138.90.58> Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 CSeq: 28614828 NOTIFY Contact: <sip:mod_sofia@78.138.90.58:6090> User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:1002@78.138.90.58 Voice-Message: 1/0 (0/0) ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 From: <sip:1002@78.138.90.58>;tag=31rQ9cSjXZ1gr To: <sip:1002@78.138.90.58>;tag=z9hG4bKDHStmZ0taSmNp CSeq: 28614828 NOTIFY Content-Length: 0 ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: ------------------------------------------------------------------------ REGISTER sip:bluesip.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H Max-Forwards: 70 From: <sip:bluesip/sherifomran@bluesip.net;transport=udp>;tag=6r0vBQZS650Fg To: <sip:bluesip/sherifomran@bluesip.net;transport=udp> Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf CSeq: 28601493 REGISTER Contact: <sip:gw+trunk_1000@78.138.90.58:6090;transport=udp;gw=trunk_1000> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="c6cdcafe0418e519bc9ee0d8fa3d4d74" Content-Length: 0 ------------------------------------------------------------------------ recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKetjKptHy71a8H From: <sip:bluesip/sherifomran@bluesip.net;transport=udp>;tag=6r0vBQZS650Fg To: <sip:bluesip/sherifomran@bluesip.net;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf CSeq: 28601493 REGISTER WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: ------------------------------------------------------------------------ REGISTER sip:bluesip.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD Max-Forwards: 70 From: <sip:bluesip/sherifomran@bluesip.net;transport=udp>;tag=6r0vBQZS650Fg To: <sip:bluesip/sherifomran@bluesip.net;transport=udp> Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf CSeq: 28601494 REGISTER Contact: <sip:gw+trunk_1000@78.138.90.58:6090;transport=udp;gw=trunk_1000> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="4c09dbe4b9accb52d4104b40dfe20040" Content-Length: 0 ------------------------------------------------------------------------ recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKF3BcrN214a1tD From: <sip:bluesip/sherifomran@bluesip.net;transport=udp>;tag=6r0vBQZS650Fg To: <sip:bluesip/sherifomran@bluesip.net;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf CSeq: 28601494 REGISTER Contact: <sip:gw+trunk_1000@78.138.90.58:6090;transport=udp;gw=trunk_1000>;q=0.5;expires=3600 Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60932 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=4ajgB89Nt8Q3K Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60932 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60932 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=4ajgB89Nt8Q3K Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60932 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60933 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=5KB9c3tSQHepF Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60933 REGISTER Contact: <sip:1002@192.168.178.26:58881;ob>;expires=300 Date: Thu, 24 May 2012 23:28:00 GMT User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: ------------------------------------------------------------------------ NOTIFY sip:1002@192.168.178.26:58881;ob SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS Route: <sip:1002@41.34.123.243:58881>;ob Max-Forwards: 70 From: <sip:1002@78.138.90.58>;tag=6v41eyBXmt48a To: <sip:1002@78.138.90.58> Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 CSeq: 28614976 NOTIFY Contact: <sip:mod_sofia@78.138.90.58:6090> User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:1002@78.138.90.58 Voice-Message: 1/0 (0/0) ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 From: <sip:1002@78.138.90.58>;tag=6v41eyBXmt48a To: <sip:1002@78.138.90.58>;tag=z9hG4bKgc54SgK51KQDS CSeq: 28614976 NOTIFY Content-Length: 0 ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: ------------------------------------------------------------------------ REGISTER sip:bluesip.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m Max-Forwards: 70 From: <sip:bluesip/salahomran@bluesip.net;transport=udp>;tag=5F739Uep9vaXm To: <sip:bluesip/salahomran@bluesip.net;transport=udp> Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf CSeq: 28601493 REGISTER Contact: <sip:gw+trunk_1002@78.138.90.58:6090;transport=udp;gw=trunk_1002> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" Content-Length: 0 ------------------------------------------------------------------------ recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKHNyXUB48yvD0m From: <sip:bluesip/salahomran@bluesip.net;transport=udp>;tag=5F739Uep9vaXm To: <sip:bluesip/salahomran@bluesip.net;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf CSeq: 28601493 REGISTER WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: ------------------------------------------------------------------------ REGISTER sip:bluesip.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg Max-Forwards: 70 From: <sip:bluesip/salahomran@bluesip.net;transport=udp>;tag=5F739Uep9vaXm To: <sip:bluesip/salahomran@bluesip.net;transport=udp> Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf CSeq: 28601494 REGISTER Contact: <sip:gw+trunk_1002@78.138.90.58:6090;transport=udp;gw=trunk_1002> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" Content-Length: 0 ------------------------------------------------------------------------ recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKjyQpX6mcv53jg From: <sip:bluesip/salahomran@bluesip.net;transport=udp>;tag=5F739Uep9vaXm To: <sip:bluesip/salahomran@bluesip.net;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf CSeq: 28601494 REGISTER Contact: <sip:gw+trunk_1002@78.138.90.58:6090;transport=udp;gw=trunk_1002>;q=0.5;expires=3600 Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60934 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=75XtgSv0H3tUp Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60934 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: ------------------------------------------------------------------------ REGISTER sip:sip.pcfone.com:6090 SIP/2.0 Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD Max-Forwards: 70 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D To: "Sherif 1003" <sip:1002@sip.pcfone.com> Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60935 REGISTER User-Agent: Telephone 1.0.2 Contact: "Sherif 1003" <sip:1002@192.168.178.26:58881;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 From: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D To: "Sherif 1003" <sip:1002@sip.pcfone.com>;tag=8eQKjmD4ecHej Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e CSeq: 60935 REGISTER Contact: <sip:1002@192.168.178.26:58881;ob>;expires=300 Date: Thu, 24 May 2012 23:32:56 GMT User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: ------------------------------------------------------------------------ NOTIFY sip:1002@192.168.178.26:58881;ob SIP/2.0 Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B Route: <sip:1002@41.34.123.243:58881>;ob Max-Forwards: 70 From: <sip:1002@78.138.90.58>;tag=9QgcmFy7BN70D To: <sip:1002@78.138.90.58> Call-ID: a1be7708-209b-1230-8985-00163e6bb553 CSeq: 28615124 NOTIFY Contact: <sip:mod_sofia@78.138.90.58:6090> User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:1002@78.138.90.58 Voice-Message: 1/0 (0/0) ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B Call-ID: a1be7708-209b-1230-8985-00163e6bb553 From: <sip:1002@78.138.90.58>;tag=9QgcmFy7BN70D To: <sip:1002@78.138.90.58>;tag=z9hG4bKK7gFZ15FSet5B CSeq: 28615124 NOTIFY Content-Length: 0 ------------------------------------------------------------------------ freeswitch@internal> --- On Thu, 5/24/12, Kristian Kielhofner <kris(a)kriskinc.com> wrote: From: Kristian Kielhofner <kris(a)kriskinc.com> Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Thursday, May 24, 2012, 7:51 PM Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: > > Hi all, > > My topology is as follows: > > Kamailio -> FS (SBS+Media server) > > I came across an issue with my system as follows. I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > <include> > <gateway name="trunk_1000"> > <param name="realm" value="bluesip.net"/> > <param name="username" value="bluesip/xxxxxx"/> > <param name="register" value="true"/> > <param name="extension" value="kb-1002"/> > </gateway> > </include> > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > 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 > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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

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fax_header question
by Tamas Jalsovszky 25 May '12

25 May '12

Hello, I'm playing with fax sending and I need to set the fax header for my own line. I can set fax_header and that works nearly perfectly for me. What I need to figure out is, how to set the whole header line, as it seems, spandsp adds the date and time to the left side and ident and page number to the right side. I tried to fill the whole fax_ident variable, but found from the codes, that it has a 50 character limit. The best would be to have fully configurable the header and maybe define the page number as a pattern (like efax or hylafax has if I recall well). Any idea? Kind regards, jalsot

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Re: [Freeswitch-users] RTP media issue
by Nathan Downes 25 May '12

25 May '12

Hi, enable-soa <param name="enable-soa" value="true"/> Set the value to "false" to diable SIP SOA from sofia to tell sofia not to touch the exchange of SDP I don't think this is related to the exchange of an SDP message.. Can you elaborate more before I try it? I can't make things worse or change things I don't understand. From: djbinter(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org CC: nathan(a)nortec.com.au Subject: Re: [Freeswitch-users] RTP media issue Date: Fri, 25 May 2012 11:19:46 +1000 <param name="enable-soa" value="false"/> Sent from my iPad On May 24, 2012, at 5:01 PM, Nathan Downes <nathandownes(a)hotmail.com> wrote: Hi, I had previous reported an issue with poor voice quality, appearing to stem from occasion wrong timestamps coming from provider, but the end user's experience was much worse than what I could see/hear in the trace. I have finally captured an event inbound and outbound. The thing I don't understand is I thought even though FS proxied the media it didn't touch it or change anything, but it appears it is. The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and http://www.nortec.com.au/outbound.pcap.gz Inbound is from my trunk provider to FS box and outbound is FS box to ATA in FTTH GPON. The event I am talking about, if both traces are open, is in the inbound one inbetween packet 8114 and 8117 the provider drops a packet or I don't receive it. In the corresponding outbound trace, between packet 8144 and 8152, it appears FS misses a whole heap of packets (.1 seconds) between 8146 and 8152 then it increases the timestamp only by 40 rather than 160 on packet 8152. This seems to not affect SIP phones themselves but causes issues with the FTTH GPON ATA. This causes a gap in the audio for the end user, and when they miss a high number of packets even though it sounds good on the inbound trace the end users experience is horrible. This trace is actually a good one, but the wrong timestamp can occur once per second, causing end user to lose 10%+ of incoming audio only. The issue only affects the audio coming from provider to FS to end user. I am chasing it up with the voice provider to try and eliminate the occasional packet loss, but if I could stop/fix FS from doing its adjustment/gap/something the end user wouldn't even notice it. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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

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