January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

NAT traversal - the final say..!
by Cal Leeming [Simplicity Media Ltd] 05 Feb '13

05 Feb '13

*Any and all feedback on this thread would be much welcomed.* Hello, * * There seems to be a large number of discussions surrounding NAT traversal, as well as lots of documentation, but with no concrete answers. The NAT related wiki documentation is tedious, and depending on the outcome of this thread, I'd like to spend some time cleaning it up. The most common problem (the same as ours) was having a router with broken ALG and a softphone that does not seem to work with STUN. The following REGISTER is sent from a phone. REGISTER sip:1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:57787 ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:2000@192.168.1.102:57787;rinstance=0c7190b115a36513> To: "foxx"<sip:2000@1.2.3.4:5060> From: "foxx"<sip:2000@1.2.3.4:5060>;tag=83311448 Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI. CSeq: 7 REGISTER Expires: 120 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Supported: replaces User-Agent: 3CXPhone 6.0.25732.0 Content-Length: 0 As you can see, the client's public IP is not specified anywhere. FreeSWITCH offers several ways around this, the main ones being; * NDLB-connectile-dysfunction * NDLB-force-rport * apply-nat-acl * sip-force-contact The one that has worked in our case was "NDLB-connectile-dysfunction" (otherwise known as NAT HACK), however there seems to be a lot of negative comments about using this. >From what I can tell, the general argument is that NAT HACK is considered a non RFC compliant hack, and the SIP phones should be doing a better job of keeping to the RFCs. In principle, this is a fair argument - but in practise, it's not a reasonable assumption that all phones are RFC compliant, and (imho) not a reasonable argument to have this functionality disabled by default. So, I'd like to present the following arguments; * Are there any other negative aspects about using NDLB-connectile-dysfunction, other than it is a non compliant RFC hack? * Why is NDLB-connectile-dysfunction not enabled by default when certain conditions are met? In the event that FreeSWITCH receives a REGISTER from a phone specifying a Contact/Via as 192.168.0.0/16, but received on a public IP, then it should be obvious that NAT is broken and automatically try to circumvent it. * People seem to get confused between server side and client side NAT problems, and that they both need to be resolved in a different way. The documentation doesn't seem to reflect this clearly.

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building mod_opal and v1.2stable
by Gilbert Amar 05 Feb '13

05 Feb '13

Hi, I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal 3.10.7 on a debian Ptlib and opal get build with no error, but when trying mod_opal I get Compiling /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -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 -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DP_64BIT -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, switch_frame_t**, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: error: 'class OpalCall' has no member named 'IsSwitchingT38' /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, const switch_frame_t*, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: error: 'class OpalCall' has no member named 'IsSwitchingT38' make[3]: *** [mod_opal.lo] Error 1 make[2]: *** [install] Error 1 make[1]: *** [mod_opal-install] Error 1 make: *** [mod_opal-install] Error 2 Has anyone succeded in building mod_opal ? Should I go with mod_h323 instead ? Thanks for your help

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Callgroup keeps ringing after answer - changed from shared extensions
by Mario G 04 Feb '13

04 Feb '13

Been at this for 3 weeks now, hope someone can help.... This started after replacing shared extensions with unique users and updating from an early 2011 git. Ring group via "pointer" works fine as it always has. However, answer via callgroup variable intermittently (30-40%) leaves one or mores phone ringing even though the the call was answered at another extension. The darn thing is that I can't reproduce it myself to get a trace. Any help/suggestions greatly appreciated. Phones are SPA962 (UDP) and Bria on iPad (TCP, same problems) and they all stopped ringing when they shared IDs. Mario G default.xml ring-group has (abbreviated to save space)------------------: <group name="home"> <users> <user id="100" type="pointer"/> <user id="101" type="pointer"/> ..... more users </users> </group> <group name="work"> <users> <user id="200" type="pointer"/> <user id="201" type="pointer"/> </users> </group> </groups> Extension----------------------: <user id="101"> User: <params> <param name="password" value="$${default_password}"/> </params> <variables> <variable name="voicemail_domain" value="xxxxxx.com"/> <variable name="toll_allow" value="domestic,domestic10,international,local"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Front Desk"/> <variable name="effective_caller_id_number" value="101"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>  <variable name="callgroup" value="home"/> </variables> </user> Bridge command-----------------: <extension name="xxphone"> <condition field="destination_number" expression="^2920$"> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>  <action application="execute_extension" data="2925 XML default"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ringback=file_string://${xxsounds}hi-long.wav!${sayname}!tone_stream://${us-ring};loops=-1"/>  <action application="bridge" data="{originate_timeout=17}${group_call(home@${domain_name}+A)}"/>

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Dequeue DTMF from FreeTDM Channel
by Subhash 03 Feb '13

03 Feb '13

Hi all, I have a freeswitch configured with freetdm and wanpipe.My problem is with dtmf. When calling from an external pstn number to freeswitch, dtmf seems to be detected properly and enqueued it in the channel consistently. But when call came to same channel again, dtmf which is stored in the buffer for the previous call is used in the current call.So when i am done with previous call i want to dequeue the dtmf in the channel .So the problem seems to be something I've misconfigured with the Sangoma card. Can anyone point me in the right direction? Thanks, Subhash.

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FreeSWITCH Mirror on Github
by Ken Rice 02 Feb '13

02 Feb '13

Hey Guy, As many of you know there is a Github mirror of the GIT repo at https://github.com/FreeSWITCH/FreeSWITCH . This is intended as a READ ONLY mirror of our master repos. I have noticed that some people are actually doing push request and opening issues there. Please do not do that. The Core dev team rarely (if ever) reviews the other functions that Github provides. If you have an issue or a possible patch, please open a jira and attach the patch. If you have already patched your local copy, using the git diff¹ command to get the patch will work, just pipe its output to a .txt file and attach to your Jira. Thanks Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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[CS_NEW] [WRONG_CALL_STATE] with incoming calls
by Frederick Pruneau 01 Feb '13

01 Feb '13

Hi guys! I have a problem with my freeswitch server and I need some help. All incoming calls are not working since yesterday. All I did is to change the IP address. The only error message I have is this one: [WARNING] switch_core_state_machine.c:514 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb sofia/internal/XXX@XXX.XXX.XXX.XXX:5060 Abandoned [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/XXX@XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] What does Wrong call state mean and what can I do to fix this issue? Everything else is working. I can make outgoing calls and call other extensions. Thanks in advance. Fred

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mod_httapi problem when trying to play an audio hosted in S3 amazon cloud
by Leonardo N. S. Pereira 01 Feb '13

01 Feb '13

Hi. I'm having problem when trying to play an audio hosted in S3 Amazon cloud The MP3 URL is: / / /h//ttp://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2Fblind_willie.mp3?Expires=1349381157&AWSAccessKeyId=AKIAJKQT6QPSI3BXJJ4A&Signature=bnJgjbnxqlgsdYIcKIbYUa9gqao%3D/ The result is: 2012-10-04 20:27:28.891298 [DEBUG] switch_ivr.c:605 sofia/external/+15752222222(a)67.231.8.195 Command Execute playback(http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2… EXECUTE sofia/external/+15752222222(a)67.231.8.195 playback(http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2… 2012-10-04 20:27:28.903180 [ERR] mod_httapi.c:2545 File at url [http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2Fblind_wi…] is unreachable! If I use the Browser with same URL, I'm able to listen to the sample audio. Any idea about what is wrong? Thanks in advance, Leonardo

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simple install question
by Sean Devoy 01 Feb '13

01 Feb '13

Hi, I want to switch from the stable release version to the git head and rebuild in hopes of ridding myself of the delay in connection of the audio. On almost all my calls I must hesitate about 1 second before saying hello or the caller does not hear me. I have seen many discussions saying that this is probably fixed in the git root. So here are my questions: The procedure appears to be: Log in to root cd /usr/local/src/freeswitch git checkout master Which responds: Branch master set up to track remote branch master from origin. Switched to a new branch 'master' Then just: make current Is that correct? Do I need to stop the running FS while running this? If so, how long will I be down? (ball park it) I realize this is a RTFM kind of question, but I can't risk an outage. I need confirmation from someone who understands. Thanks, Sean

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problems with freeswitch + zrtp in proxy-media mode
by Eli Burke 01 Feb '13

01 Feb '13

I'm trying to use Freeswitch with an open source voip client called Linphone, but cannot manage to establish functioning ZRTP sessions. The problem *seems* to be related to the RTP stream's SSRC. I'm trying to conclusively determine if the problem lies with Linphone, Freeswitch, or one of the underlying zrtp implementations (libzrtp and zrtpcpp). Here's what I've determined so far: 1) ZRTP support is compiled and functioning. I can establish ZRTP sessions between two iPhones running Groundwire. 2) linphone + ZRTP works correctly if I make a one legged call to a Freeswitch echo extension 3) linphone + ZRTP works correctly if I turn on bypass-media 4) linphone gets a CRC error when trying to verify the ZRTP packet on any two-legged call 5) disabling the CRC check results in a successful ZRTP session, but the audio is extremely garbled Is Freeswitch modifying the ZRTP packets when running in proxy mode? If so, how is it that some clients are able to handle this situation? Thanks for any suggestions, Eli

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ICTFAX installation in FS
by Archana Venugopan 01 Feb '13

01 Feb '13

Hi, Am trying to install ICTFAX. As per this site ictfax-20-installation-guide<http://ictfax.org/content/ictfax-20-installation-guide> I followed the instructions. But got stuck in the below procedures 1. Create a symbolic link for /usr/ictfax in /var/www/html ln -s /usr/ictfax /var/www/html/ictfax ln: creating symbolic link `/var/www/html/ictfax': No such file or directory. There is no ictfax directory in the folder. So what to do? 2. Now visit http://DOMAIN.COM/ictfax and follow the installation instructions for ICTFax (drupal based) front end installation When I click on the link its giving 404 error. Can anyone please guide me on how to proceed further on this. Many thanks Regards, Archana

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