May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch stable on embedded devices?
by GillesToo 22 May '12

22 May '12

Hello I'd' like to hear from people who run Freeswitch on embedded devices such as the Seagate DockStar, the SheevaPlug, etc. I intend to connect the device to the PSTN through an Obi110 VoIP gateway along with an ITSP over the Net, and support up to 30 internal extensions. Is Freeswitch on such small hardware rock-solid, is it easy to recompile for such non-x86 platforms, or should I get a regular x86 host like the Asus EeeBoxPC (www.asus.com/Eee/)? Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-stable-on-embedded… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 5

default dialplan: signal busy when busy and voicemail disabled
by John Jacob 22 May '12

22 May '12

Hi, I guess you are all aware of the default dialplan entry for local extensions: <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> [...] <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> So, if a caller calls a callee, which disabled his voicemail and is currently busy/unavailable FreeSWITCH plays a good bye message and hangs up. This is unexpected for most of our clients and they are interpreting it as a malfunctioning PBX. I'd like to change this default behavior and respect the extension's setting for "vm-enabled". If it's enabled, the default behavior is fine, if it's disabled, I'd like FreeSWITCH to signal *busy* to the caller. It looks like I can't test for user parameters with conditions, so I don't have a clue how to implement this? For now I just removed answer,sleep and bridge to voicemail and busy is signalled just fine, but I'm aware that I killed voicemail completely by doing this ;) Any hints or tipps? I can't implement dynamic dialplans with mod_curl though, because those don't fit into the current setup. Thanks, Johannes

3 2

Re: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports
by Mario G 22 May '12

22 May '12

Wish I could help more... this is what I tried today and all worked fine, I am on Snow Leopard. You may want to try to redo the git, may have been an xfr issue? 1, git pull followed by make current - worked 2. clean src fs dir, new git reload, bootstrap.sh, configure with ssl, make install - worked fine On May 21, 2012, at 10:29 AM, Neil Patel wrote: > Hi Mario, > > Thanks for your valuable posts on getting FS up and running on OS X. > > I'm doing so myself and ran into a snag compiling latest Git: > http://jira.freeswitch.org/browse/FS-4230 > > Given your expertise on FS and OSX I thought you may be able to help me get through this issue. Any suggestions? > > Thanks, > Neil > > On Fri, May 11, 2012 at 9:41 AM, Mario G <mario_fs(a)mgtech.com> wrote: > Some answers for you: > > On May 10, 2012, at 7:23 PM, Chris Mylonas wrote: > >> That's great Mario. I will update my blog accordingly where I posted some stuff (http://mrvoip.com.au/blog/install-freeswitch-osx-mysql-must-remove-macports) >> >> I just winged it because it used to work, so most of the prerequisites were installed. >> I did not put the --with-openssl flag when doing configure. What functionality do I lose? > > From what I remember, it complained about gnutis or something like that, you have to check the messages but it is really important, used to not be required. It took several days to figure this one out. I think the message was: > configure: error: --with-ssl was given, but GNUTLS is not available. > >> >> >> pkgconfig was already on my system from earlier efforts - so I guess I got lucky with that one. >> To be honest, once I resolved my macports issues (by removing it) it was just like doing it in linux. > > The funny thing is I still see a few pkgconfig missing messages during the make but FS works fine for me, I think it's for things I am not using. I will be looking into this more before I update the wiki. > >> >> I'd like to give it another shot with a fresh OSX install - but I really don't want to go about setting up my machine again! I have an old imac that is not doing much, but I don't think it will run 10.6 :( I will try and give it a shot this afternoon seeing as my girlfriend is out taking photos :) > > FYI, when I have not updated for a long time or make major changes like updating xCode: I backup the Mac Mini running FS as a bootable partition (I use SuperDuper), then I connect it to an iMac and boot using firewire target mode to boot from the backup. Now I can play round all I want. If all goes well, I just restore to the Mini, if not I can start over. For frequent updates I just backup the FS bin and source in the Mini with and do a git pull, etc.... > Mario G >> >> Kind Regards, >> Chris >> >> >> >> >> On 11/05/2012, at 11:48 AM, Mario G wrote: >> >>> Chris, I have been running FS on 10.6,x since 2010 fine and have been updating a lot recently to help testing, I have had no problems. I don't use macport stuff. I wrote the original FS osX wiki but there are some differences since 2010 (I am working on fixing the wiki in the next few weeks). The wiki install instructions work if you add the following. Hope this helps: >>> >>> 1. Add this info: >>> FreeSWITCH™ has many functions that invoke PKG-CONFIG, so it must be downloaded and installed separately. >>> # Got to [http://pkgconfig.freedesktop.org/releases/ here] and download the latest pkg-config, by default it is placed into your Downloads folder. MUST USE .25 for OSXdue to glib2 dependencies! >>> # Open Downloads and click the file to uncompress it. >>> # Launch the Terminal application if not already running and issue these commands to move the source to the src directory, build and install: >>> cd ~/Downloads >>> mv pkg-config-0.25 /usr/local/src >>> cd /usr/local/src/pkg-config-0.25 >>> ./configure >>> make >>> sudo make install <– sudo is required because binaries are loaded into /usr/bin >>> >>> >>> 2. You MUST do this: >>> ./bootstrap.sh >>> ./configure --with-openssl >>> >>> 3. The FLITE fix is no longer needed. >>> >>> On May 10, 2012, at 5:01 PM, Chris Mylonas wrote: >>> >>>> Hi FS Users, >>>> >>>> tl;dr; - removed macports, removed tree, pulled fresh tree = installed FS on OSX. >>>> >>>> >>>> Here is the longer version that had a bit of a pre-emptive whinge about libtool version mismatch. But we never got there - it worked! >>>> >>>> I'm in the process of re-installing FS on OSX (10.6.8). I have removed Macports to try and get this up and running. >>>> I re-bootstrapped and got this error: >>>> >>>> quiet_libtool: Version mismatch error. This is libtool 2.4, but the >>>> quiet_libtool: definition of this LT_INIT comes from libtool 2.2.4. >>>> quiet_libtool: You should recreate aclocal.m4 with macros from libtool 2.4 >>>> quiet_libtool: and run autoconf again. >>>> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 63 >>>> make: *** [all] Error 2 >>>> >>>> Removed the whole git tree just in case there was some left over junk. Pulled a fresh tree and noticed this remark during a fresh bootstrap >>>> >>>> arakis:freeswitch-git chrismylonas$ rm -Rf freeswitch/ >>>> arakis:freeswitch-git chrismylonas$ git clone git://git.freeswitch.org/freeswitch.git >>>> Cloning into freeswitch... >>>> remote: Counting objects: 185609, done. >>>> remote: Compressing objects: 100% (39208/39208), done. >>>> remote: Total 185609 (delta 143241), reused 181443 (delta 140035) >>>> Receiving objects: 100% (185609/185609), 77.85 MiB | 267 KiB/s, done. >>>> Resolving deltas: 100% (143241/143241), done. >>>> arakis:freeswitch-git chrismylonas$ cd freeswitch/ >>>> arakis:freeswitch chrismylonas$ ./bootstrap.sh >>>> bootstrap: checking installation... >>>> bootstrap: autoconf version 2.61 (ok) >>>> bootstrap: automake version 1.10 (ok) >>>> bootstrap: aclocal version 1.10 (ok) >>>> bootstrap: libtool version 2.2.4 (ok) >>>> Bootstrapping using: >>>> autoconf : /usr/bin/autoconf >>>> automake : /usr/bin/automake >>>> aclocal : /usr/bin/aclocal >>>> libtool : /usr/bin/glibtool (2.2.4.) >>>> libtoolize: /usr/bin/glibtoolize >>>> make : /usr/bin/make (GNU Make 3.81) >>>> awk : () >>>> >>>> It is still reporting a 2.2.4 version of libtool. >>>> ... >>>> ... >>>> ... >>>> In the end, it has been compiled and installed though >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> 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 >>> >>> 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 >> >> 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 > > 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 > >

2 2

SAVPF and a=crypto
by peter＠uringme.com 22 May '12

22 May '12

I'm trying out sipML5 ( http://code.google.com/p/sipml5/ ) with FS and I had a problem with their demo. Their code uses a=crypto lines on RTP/SAVPF and FS is bouncing it with: 2012-05-21 13:04:47.754843 [ERR] sofia_glue.c:4478 a=crypto in RTP/AVP, refer to rfc3711 The crypto line is in RTP/SAVPF, though, not RTP/AVP. Is this line coming up because it's not RTP/SAVP? Am I missing a module for RTP/SAVPF, or is this even supported? Following is a trace -- I'm calling x7779 on my server which just plays a voice prompt. INVITE sip:7779@x.x.x.x SIP/2.0 Via: SIP/2.0/UDP 87.106.69.240:4060;branch=z9hG4bK-524287-1---a56eca302b8d6412;rport Via: SIP/2.0/TCP y.y.y.y:62676;branch=z9hG4bK1P36uFnLdbzHiEzWdVAd0G6nk6kdOdEt;rport=62676;received=y.y.y.y Max-Forwards: 69 Contact: "Peter Krause"<sip:7777@87.106.69.240:4060;transport=udp;ws-src-ip=y.y.y.y;ws-src-port=62676>;+sip.ice To: <sip:7779@x.x.x.x> From: <sip:7777@uringme>;tag=gz6bNBeAkHM7eAR2eoJG Call-ID: 434d70e1-5cc1-d096-442c-499c8cd600ad CSeq: 6437 INVITE Content-Type: application/sdp Organization: Doubango Telecom User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0 Content-Length: 3483 v=0 o=- 275367574 1 IN IP4 127.0.0.1 s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000) t=0 0 m=audio 54450 RTP/SAVPF 103 104 0 8 106 105 13 126 c=IN IP4 y.y.y.y a=rtcp:54451 IN IP4 y.y.y.y a=candidate:0 1 udp 2130706432 192.168.1.8 50256 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 udp 2130706432 192.168.1.8 50257 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=candidate:0 1 udp 1912602624 y.y.y.y 54450 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 udp 1912602624 y.y.y.y 54451 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=candidate:0 1 tcp 1694498816 192.168.1.8 62677 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 tcp 1694498816 192.168.1.8 62678 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:33ayFPGAoi+nni8kbYZfBO7tH3qI1SaH7ilFU7GT a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Hd6rAXtyms1rJgC4OTJk/K4AGpnwLUn9Aqdx1sbd a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 a=sendrecv m=video 54452 RTP/SAVPF 100 101 102 c=IN IP4 y.y.y.y a=rtcp:54453 IN IP4 y.y.y.y a=candidate:0 1 udp 2130706432 192.168.1.8 50258 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 udp 2130706432 192.168.1.8 50259 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=candidate:0 1 udp 1912602624 y.y.y.y 54452 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 udp 1912602624 y.y.y.y 54453 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=candidate:0 1 tcp 1694498816 192.168.1.8 62679 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 tcp 1694498816 192.168.1.8 62680 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=mid:video a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:RIxXHNE/5GtDfkXnAV4CLXBQGxPzlzBg3Xyg3Lfs a=rtpmap:100 VP8/90000 a=rtpmap:101 red/90000 a=rtpmap:102 ulpfec/90000 a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 a=sendrecv ------------------------------------------------------------------------ send 448 bytes to udp/[87.106.69.240]:4060 at 17:04:47.774032: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 87.106.69.240:4060;branch=z9hG4bK-524287-1---a56eca302b8d6412;rport=4060 Via: SIP/2.0/TCP y.y.y.y:62676;branch=z9hG4bK1P36uFnLdbzHiEzWdVAd0G6nk6kdOdEt;rport=62676;received=y.y.y.y From: <sip:7777@uringme>;tag=gz6bNBeAkHM7eAR2eoJG To: <sip:7779@x.x.x.x> Call-ID: 434d70e1-5cc1-d096-442c-499c8cd600ad CSeq: 6437 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2012-05-21 13:04:47.754843 [NOTICE] switch_channel.c:816 New Channel sofia/external/7777@uringme [120f19be-a367-11e1-b22e-57011ca30f08] 2012-05-21 13:04:47.754843 [DEBUG] sofia.c:4770 Channel sofia/external/7777@uringme entering state [received][100] 2012-05-21 13:04:47.754843 [DEBUG] sofia.c:4781 Remote SDP: v=0 o=- 275367574 1 IN IP4 127.0.0.1 s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000) t=0 0 m=audio 54450 RTP/SAVPF 103 104 0 8 106 105 13 126 c=IN IP4 y.y.y.y a=rtcp:54451 IN IP4 y.y.y.y a=candidate:0 1 udp 2130706432 192.168.1.8 50256 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 udp 2130706432 192.168.1.8 50257 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=candidate:0 1 udp 1912602624 y.y.y.y 54450 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 udp 1912602624 y.y.y.y 54451 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=candidate:0 1 tcp 1694498816 192.168.1.8 62677 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 tcp 1694498816 192.168.1.8 62678 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:33ayFPGAoi+nni8kbYZfBO7tH3qI1SaH7ilFU7GT a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Hd6rAXtyms1rJgC4OTJk/K4AGpnwLUn9Aqdx1sbd a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 m=video 54452 RTP/SAVPF 100 101 102 c=IN IP4 y.y.y.y a=rtcp:54453 IN IP4 y.y.y.y a=candidate:0 1 udp 2130706432 192.168.1.8 50258 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 udp 2130706432 192.168.1.8 50259 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=candidate:0 1 udp 1912602624 y.y.y.y 54452 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 udp 1912602624 y.y.y.y 54453 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=candidate:0 1 tcp 1694498816 192.168.1.8 62679 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 tcp 1694498816 192.168.1.8 62680 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=mid:video a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:RIxXHNE/5GtDfkXnAV4CLXBQGxPzlzBg3Xyg3Lfs a=rtpmap:100 VP8/90000 a=rtpmap:101 red/90000 a=rtpmap:102 ulpfec/90000 a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 2012-05-21 13:04:47.754843 [ERR] sofia_glue.c:4478 a=crypto in RTP/AVP, refer to rfc3711 2012-05-21 13:04:47.754843 [DEBUG] switch_channel.c:2592 (sofia/external/7777@uringme) Callstate Change DOWN -> HANGUP 2012-05-21 13:04:47.754843 [NOTICE] sofia.c:4998 Hangup sofia/external/7777@uringme [CS_NEW] [INCOMPATIBLE_DESTINATION]

1 1

Ring group auto answers and then diconnects
by Cj B 22 May '12

22 May '12

Hi all, I've setup a pretty simple ring group using fusionpbx and multi-tenant. I have ext's 9001-9004 and a ring group of Ext 8003. All 4 exts are part of the ring group. When I dial the ring group ext it gets answered right away but none of the phones ever ring. Then after 30 seconds (the timeout) I get the greeting "Good bye" and then it disconnects. I've posted all the output that happens here: http://pastebin.com/dZjwKVSA Can someone please help me out, this seems like it should be a pretty simple thing to do, so i'm sure it's just a setup issue that I've done wrong! Thanks. Cj B

3 3

Registration Failed with status Service Unavailable [503]
by Daniel Sokolowski 22 May '12

22 May '12

Hi All, I'm running FreeSWITCH Version 1.2.0 (git-b3b2c37 2012-05-18 13-41-16 -0500) and have been running a FreeSWITCH for over a year now more or less with no problems. I believe though I either have found a bug or a problem in my Debian setup. If I do a system reboot FreeSWITCH fails to connect to my 3 gateways: 2012-05-21 21:15:31.115850 [ERR] sofia_reg.c:1986 freephoneline.ca Registration Failed with status Service Unavailable [503]. failure #14 2012-05-21 21:15:31.115850 [ERR] sofia_reg.c:1986 internetcalls.com Registration Failed with status Service Unavailable [503]. failure #14 2012-05-21 21:15:31.215876 [ERR] sofia_reg.c:1986 voip.ms Registration Failed with status Service Unavailable [503]. failure #14 The work around is to manually call the init.d script '/etc/init.d/freeswitch restart'. I have been able to duplicate this behaviou over a dozen times now. I believe during boot FreeSWITCH somehow detects wrong network settings and keeps using them on retries - only a restart clears this issue. My init.d script was taken from the FS wiki and can be found here: http://dpaste.com/750951/ --- I already modified it to require '$network' service with no effect on the issue I'm having. Does anyone have any idea what might be causing this? -- Daniel Sokolowski Web Engineer Danols Web Engineering http://webdesign.danols.com/ Office: 613-817-6833 Fax: 613-817-4553 Toll Free: 1-855-5DANOLS Kingston, ON K7L 1H3, Canada Notice of Confidentiality: The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review re-transmission dissemination or other use of or taking of any action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you received this in error please contact the sender immediately by return electronic transmission and then immediately delete this transmission including all attachments without copying distributing or disclosing same.

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BYE message is not relayed to the UAC
by Arif Hossain 21 May '12

21 May '12

Hi, We have the following network architecture : UAC1------------------------->kamailio-------------------->VoipSwitch----->PSTN---------->Phone1 (Sip Client) Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session is terminated cleanly. But if Phone1 hangs up the BYE message which comes to kamailio and goes back to VoipSwitch instead of relayed to UAC1 . So The session becomes a zombie one, And UAC1 unfortunately gets billed for a session which should be terminated. Following is the Call flow as seen from VoipSwitch : | <kamilio IP> | | | | <VoipSwitch IP> | |134.856 | INVITE SDP | |(7890) ------------------> (5060) | |134.858 | 407 Proxy Authentication Required | |(7890) <------------------ (5060) | |134.902 | ACK | | |(7890) ------------------> (5060) | |135.408 | INVITE SDP | |(7890) ------------------> (5060) | |135.414 | 100 Trying| | |(7890) <------------------ (5060) | |140.121 | 183 Session Progress SDP | |(7890) <------------------ (5060) | |140.184 | RTP (g729) | | |(61868) <------------------ (5136) | |141.295 | RTP (g729) | | |(61868) ------------------> (5136) | |153.701 | 200 OK SDP | |(7890) <------------------ (5060) | |153.713 | RTP (g729) | | |(61868) ------------------> (5136) | |154.126 | ACK | | |(7890) ------------------> (5060) | |159.988 | BYE | | |(7890) <------------------ (5060) | |160.031 | BYE | | |(7890) ------------------> (5060) | |160.478 | BYE | | |(7890) ------------------> (5060) | |161.412 | BYE | | |(7890) ------------------> (5060) | |163.280 | BYE | | |(7890) ------------------> (5060) | |167.015 | BYE | | |(7890) ------------------> (5060) | |170.750 | BYE | | |(7890) ------------------> (5060) | |174.481 | BYE | | |(7890) ------------------> (5060) | |178.216 | BYE | | |(7890) ------------------> (5060) | |181.952 | BYE | | |(7890) ------------------> (5060) | |185.687 | BYE | | |(7890) ------------------> (5060) | |188.018 | 408 Request Timeout | |(7890) ------------------> (5060) | |211.849 | BYE | | |(7890) ------------------> (5060) | |212.292 | BYE | Sip Traces : kamailio------>VoipSwitch I'm posting only the offending BYE msg instead of full trace , because of the mail will become difficult to read . If more traces needed, i can post it. The following BYE message is sent by VoipSwitch: BYE sip:ipphone@205.164.40.74 SIP/2.0 Route: <sip:108.166.195.189:7890;lr=on;nat=yes> CSeq: 2 BYE Via: SIP/2.0/UDP 205.164.40.74:5060 From: sip:008801673345531@205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: "123456" <sip:ipphone@205.164.40.74>;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 Kamailio instead of relaying the message, sends a BYE message towards VoipSwitch: BYE sip:ipphone@205.164.40.74 SIP/2.0 Max-Forwards: 10 CSeq: 2 BYE Via: SIP/2.0/UDP 108.166.195.189:7890;branch=z9hG4bK4b2b.5d893e95.0 Via: SIP/2.0/UDP 205.164.40.74:5060;rport=5060 From: sip:008801673345531@205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: "123456" <sip:ipphone@205.164.40.74>;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 When the first BYE message comes from VoipSwitch , kamailio does the following : May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [receive.c:289]: receive_msg: cleaning up May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [parser/sdp/sdp.c:751]: _sdp = 0x831bf10 May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [parser/sdp/sdp.c:753]: sdp = 0x83043dc May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: <core> [parser/sdp/sdp.c:755]: session = 0x8304504 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:632]: method: <BYE> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:634]: uri: <sip:ipphone@205.164.40.74> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <1> <BYE> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bk19052612230719933454843>; state=16 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [receive.c:145]: After parse_msg... May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [receive.c:186]: preparing to run routing scripts... May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=arILprdVR1srJ76HHlt4BEc3XsyaWcZm May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [76]; uri=[sip:ipphone@205.164.40.74] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body ["ipphone" <sip:ipphone@205.164.40.74>] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=1905251223419334290936029 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: siputils [checks.c:76]: totag found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:85]: is_preloaded: No May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==15 && [205.164.40.74] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 7890 matches port 5060 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==15 && [205.164.40.74] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [forward.c:448]: check_self: host != me May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 15==15 && [108.166.195.189] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 7890 matches port 7890 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:792]: Topmost route URI: 'sip:108.166.195.189:7890;lr=on;nat=yes' is me May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:257]: No next Route HF found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:811]: No next URI found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:983]: params are <;lr=on;nat=yes> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: siputils [checks.c:76]: totag found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=2501 , global msg id=2500 , T on entrance=0xffffffff May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=29177, isACK=0 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:564]: DEBUG: proceeding to pre-RFC3261 transaction matching May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb61626a4, callback type 1, id 0 entered May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [msg_translator.c:204]: check_via_address(205.164.40.74, 205.164.40.74, 0) May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) May 20 02:26:04 VOS20-108 last message repeated 5 times May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: <core> [receive.c:289]: receive_msg: cleaning up May 20 02:26:22 VOS20-108 /usr/local/sbin/kamailio[16445]: DEBUG: <core> [udp_server.c:486]: udp_rcv_loop: probing packet received from 180.234.62.230 38722 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_reply.c:1134]: ->>>>>>>>> T_code=0, new_code=408 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_reply.c:1636]: DEBUG: relay_reply: branch=0, save=0, relay=0 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: <core> [msg_translator.c:204]: check_via_address(205.164.40.74, 205.164.40.74, 0) May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: <core> [mem/shm_mem.c:105]: WARNING:vqm_resize: resize(0) called May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb61626a4, callback type 128, id 0 entered May 20 02:26:54 VOS20-108 /usr/local/sbin/kamailio[16451]: WARNING: <core> [timer.c:450]: WARNING: our timer runs faster then real-time (300000 ms / 4800 ticks our time .-> 299923 ms / 4798 ticks real time) OT: How do you guys maintain 80 column mails? i do not use a mail client, use gmail mailbox. -- -aft -- -aft

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FreeSWITCH News and Notes
by Michael Collins 21 May '12

21 May '12

Monday, May 21, 2012 We're looking forward this week to a followup on a conversation that we began last Wednesday. After Fred Dixon gave a great demonstration of Big Blue Box <http://www.bigbluebutton.org?cc12-0521> we had our very own Ken Rice give an introduction to writing a simple dialplan module in C. This week Ken Rice will follow up on the discussion and answer any questions you may have. If you missed last week's discussion you can download the audio by clicking the "torrent" link on the past calls<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls?cc12-0521>page. In ClueCon news we are happy to report that Norm Tomlins and Voice Network<http://www.voicenetwork.ca?cc12-0521>are back to sponsor us again in 2012. We look forward to seeing Norm again this August. We have also posted a number of presentations on this year's schedule <http://www.cluecon.com/schedule?cc12-0521>. More presentations are in the works, so stay tuned for further announcements. Visit the ClueCon site to register, or call us at 877.742.CLUE if you have any questions about speakers, sponsoring, or anything else related to the event. Only 77 days until ClueCon! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0521

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Eaves Drop
by Rob Moore 21 May '12

21 May '12

Hi All, Hopefully looking for a quick answer here. We are using the Eavesdrop function to supply a "call barging" service for a call centre we are running through a web UI but we have a few problems around transfers where we park the call and switching the barge from one agent to another. Basically I need some method of removing the media bug from the agents channel. I've searched the Wiki and done a few googles but I can't seem to find anything that we can call to remove a bug. We're doing all of our control of freeswitch through lua so anything that we can call from the CLI would be great. Anyone have any suggestions? Thanks in advance! Rob

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Polycom blind transfer uses unexpected dialplan context
by Vik Killa 21 May '12

21 May '12

I have a registered UA with the directory variable 'user_context' set to 'the.domain.com' When the registered UA dials a number, it first goes through the dialplan context 'the.domain.com' as expected, depending on the destination number dialed, the call will be transferred to another context, in this example 'routing' context. When the UA performs an attended transfer, the dialed number goes through the contexts as expected (the.domain.com->routing) but when performing a blind transfer the dialed number starts in the 'routing' context. Why wouldn't the dialed number follow the UA's 'user-context' variable and start in the 'the.domain.com' context? This is happening on a Polycom 330 with today's FS GIT.

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