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IP infomation in SDP
by Jimmy Chang 24 Apr '13

24 Apr '13

Hi, In the default configuration, the internal profile listen port 5060 and external profile listen port 5080. And follow the setting in http://wiki.freeswitch.org/wiki/NAT_Traversal vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=59.120.xx.xx"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> internal.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> external.xml <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> When user make a SIP call from public network using external profile(5080), he encountered no audio. Here is the SDP of 200 OK from FS to user. /v=0// //o=FreeSWITCH 1366781127 1366781128 IN IP4 10.0.110.33// //s=FreeSWITCH// //c=IN IP4 10.0.110.33// //t=0 0// //m=audio 10568 RTP/AVP 18 101// //a=rtpmap:18 G729/8000// //a=fmtp:18 annexb=no// //a=rtpmap:101 telephone-event/8000// //a=fmtp:101 0-16// //a=silenceSupp:off - - - -// //a=ptime:20// //m=video 10530 RTP/AVP 116// //a=rtpmap:116 H264/90000// //a=fmtp:116 profile-level-id=428014/ When user make a SIP call from public network using internal profile(5060), everything OK. Here is the SDP of 200 OK from FS to user. /v=0// //o=FreeSWITCH 1366780920 1366780921 IN IP4 //59.120.xx.xx// //s=FreeSWITCH// //c=IN IP4 //59.120.xx.xx// //t=0 0// //m=audio 10892 RTP/AVP 18 101// //a=rtpmap:18 G729/8000// //a=fmtp:18 annexb=no// //a=rtpmap:101 telephone-event/8000// //a=fmtp:101 0-16// //a=silenceSupp:off - - - -// //a=ptime:20// //m=video 10178 RTP/AVP 116// //a=rtpmap:116 H264/90000// //a=fmtp:116 profile-level-id=428014// / I can't figure out which parameters in sofia profile or in dialplan make port 5060 set the public IP in SDP and work fine. Any advice? Thanks in advance. Jimmy

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Re: [Freeswitch-users] Multi-tenat PBX using xml_curl
by Chris B. Ware 24 Apr '13

24 Apr '13

Anthony, any domain has the same dial-string: <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> and when an extension calls another extension I use: <action application="bridge" data="{ignore_early_media=true}sofia/$${domain}/${Called_Exten}"/> Date: Mon, 22 Apr 2013 16:58:13 -0500 From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] Multi-tenat PBX using xml_curl To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Message-ID: <CAKbxfG9YOB9pPPyBok-srfYQ8a1KRxbE2S7TaKESYwxiG_uFng(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" What is the dial string you are using to call them? On Mon, Apr 22, 2013 at 4:51 PM, Chris B. Ware <chrisbware(a)yahoo.it> wrote: > Hi guys, > > I've looked for a help on this issue on irc and mailing-list archive > without success. > I need to configure a multitenant PBX using xml_curl. I've followed wiki > to do it. > > I've two domains: > > DomainA.voip.com > DomainB.voip.com. > > I can register extensions to those domains but when I call one extensions > from another I receive > an "INVALID PROFILE" error. > > The only way I've found to fix this issue is to add those domains as > aliases under internal profile. > > I thought they were automatically added: what should I set to generate > their aliases automatically? > > Please, help. Thanks. > > _________________________________________________________________________ > 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 > > -- 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

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sip to h323 video call via freeswitch
by Navnath Sonavne 24 Apr '13

24 Apr '13

Anthony, any domain has the same dial-string: <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> and when an extension calls another extension I use: <action application="bridge" data="{ignore_early_media=true}sofia/$${domain}/${Called_Exten}"/> Hi Anthony, You said any domain has same dial-string so i searched it in various files but i did not find it. Can you tell me in which file i can find it? In reply you said when an extension calls another extension I use: <action application="bridge" data="{ignore_early_media=true}sofia/$${domain}/${Called_Exten}"/> I used this action in my default.xml but no success. FYI I am able to convert sip protocol to h323 protocol for Audio call. Here is my extension from default.xml to convert sip to h323 : <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{4})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/> <action application="set" data="PEER_IP=192.168.8.103"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="h323/$1@${PEER_IP}"/> </condition> </extension> I am using this for video call also but no success. I am attaching vars.xml , default.xml , internal.xml , h323.conf.xml , opal.conf.xml and sofia.conf.xml. Please look in these files and tell me if am I going wrong and what should be done to make sip to h323 video call? -Navnath

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How to change Music-on-hold while digit_bind_action is used ?
by julien terrasson 24 Apr '13

24 Apr '13

Hi, First of all, i've been discovering freeswitch recently and i found that this is an awesome piece of software. I really enjoy discovering the possibilities of the software and creating scenarios. Thanks for all the people who made this possible and available. The reason i'm posting is that i need help me to guide me in troubleshooting.. This is the scenario i would like to acheive :* * *************************************************** * A & B are registered on FS * A call B. * When A press the star key => B is play a prompt that ask him if he would like to receive an email. => B can then choose to accept (by pressing "1") or reject (by pressing "2") Meanwhile, A is played a specific moh that tell him that he have to wait for B answer. This the way i tried to implement it : ****************************************** DIALPLAN : i'm using bind_digit_action with "*" to start a lua script on the *b_leg context*. => "bind_digit_action","my_context,*,exec:lua,send_email.lua,a,*peer*" EMAIL.LUA : *=> Change the moh on the A leg *api:executeString("uuid_setvar ".. a_leg_uuid .. " hold_music /usr/local/freeswitch/sounds/fr/ca/june/voxService/fr_wait_email.wav"); *=> Play and request digit (B leg)* digit = session:playAndGetDigits(1, 1, 1, 5000, '#', '/usr/local/freeswitch/sounds/fr/ca/june/voxService/fr_ask_email.wav', '', '\\d+'); 3/ Result : ************* When "*" is pressed for the 1st time, FS act as if moh wasn't change. It does only take into account the change the second time "*" is pressed. i assume thats moh change can only be taken into account when bind_digit_action is over. So i would like to know if there is any existing way to change this behaviour ? If not, what would be the best way of implementing this scenario ? Thanks for your answers. Julien

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Jira and the PasteBin... [Please READ ME]
by Ken Rice 24 Apr '13

24 Apr '13

Hey Guys, Please keep the following things in mind. Bugs should be reported via Jira. Yes its ok to ask on the mailing list to see if it you are just doing something wrong, but please open a bug on Jira if its a bug. A segfault is always a bug. (Even if you are doing it wrong, FreeSWITCH should not segfault) The Pastebin is a useful tool for sharing information between people, but please do not reference pastebin URLs in Jira Tickets. Pastebin entries are not guaranteed to survive for any period of time, and they can and will disappear at some point. This means of you open a ticket and you put a backtrace in the pastebin, if the pastebin is cleared so is your backtrace. (if you use the fscore_pb script to help collect the data, there is a download link right on the pastebin that will let you download a .txt file for that entry which you can then attach to the ticket ) Please Attach to jira ticket all backtraces, sip traces and logs as plain text files ( .txt files ) Also, please do not paste the results into the notes or the description of the issue. This just means we have to scroll down 97 screens of stuff to get the last notes on a ticket. These simple things will help out the developers and help make the bugs go away faster! Thanks! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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URGENT HELP! Customers cant make any calls -after MAKE CURRENT
by Sean Devoy 23 Apr '13

23 Apr '13

Hi All, I do truly hope someone responds before morning when my customers get back to work. I did a make current and it went swell! I rebooted, everything registered, looked great - BUT . Everything everyone dials gets codec mismatches and then [INCOMPATIBLE_DESTINATION] The codecs look like this: 2013-04-22 23:42:02.138488 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] But I don't know how to set that parameter in in FS or on the devices! Any help greatly appreciated. Sean

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FreeSWITCH Weekly News and Notes
by Michael Collins 23 Apr '13

23 Apr '13

A belated welcome to the week of April 22! News and notes is a day late because I spent all day yesterday working on the new FreeSWITCH book <http://www.packtpub.com/freeswitch-1-2/book>. Good news - we have submitted all the content! It's now just a matter of getting all the edits and reviews done. We're a little more than halfway through that process. We'll update you as more news becomes available. Last week we had a conference call that<http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_17>lasted quite a while. Dave Kompel showed us some tricks with using MS Visual Studio 2012 to build FreeSWITCH applications. That discussion dovetailed into a more general discussion about some of the other things Dave has been doing with FreeSWITCH under Windows. In the weeks and months ahead, Dave will be doing some more Windows-only discussions outside of the main weekly conference call. Keep an eye on the mailing list<http://lists.freeswitch.org>for more announcements. Tomorrow we have scheduled an engineer from JeraSoft<http://www.jerasoft.net/products/vcs/index.html>who will be talking about their routing and billing solution that has been tightly integrated with FreeSWITCH 1.2. If you have FreeSWITCH in a production environment where billing and routing are core elements then this discussion <http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_24> will definitely be of interest. Have a great rest of the week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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http_cache with mod_httapi
by Srigo 23 Apr '13

23 Apr '13

Hi, I want to use http_cache to play a file in httapi dialplan but its not working. Here is what im getting. Here is my httapi xml: <document xmlns:xs="http://www.w3.org/2001/XMLSchema" type="xml/freeswitch-httapi"> <params/> <variables> <dtmf>null</dtmf> </variables> <work> <execute application="export" data="app_id=${app_id}"/> <execute application="log" data="DEBUG: Matching calendar: opening_hours, status:open"/> <execute application="flush_dtmf"/> <execute application="log" data="INFO: application=ivr;value=http://xxxxxxx.com:8080/tel/media_srv/ib_fr_wel.wav;user_entry=null;max_failure=0;loop=0;app_id=593924623;story_id=1;status=start"/> <execute application="set" data="${http_prefetch(http://xxxxxxx.com:8080/tel/media_srv/data/iBrowse/ib_fr_wel.wav)}"/> <execute application="set" data="file_cached=${http_get(http://xxxxxxx.com:8080/tel/media_srv/data/iBrowse/ib_fr_wel.wav)}"/> <execute application="set" data="app_id=593924623"/> <execute application="set" data="story_id=1"/> <playback name="user_entry" file="${http_get(http://xxxxxxx.com:8080/tel/media_srv/data/iBrowse/ib_fr_wel.wav)}" error_file="" digit-timeout="1500" input-timeout="2000"> <bind strip="#">~\d+</bind> <bind strip="#">~\d+</bind> <bind strip="#">~\d+</bind> </playback> </work> </document> Here is what im getting in logs: 2013-04-23 14:40:59.234148 [INFO] switch_ivr_async.c:194 Digit parser HTTAPI: Setting realm to 'default' 2013-04-23 14:40:59.234148 [DEBUG] switch_ivr_async.c:285 Digit parser HTTAPI: binding \d+/default/0 callback: 0x7fd528bf0840 data: 0x7fd524046640 2013-04-23 14:40:59.234148 [DEBUG] switch_ivr_async.c:285 Digit parser HTTAPI: binding \d+/default/0 callback: 0x7fd528bf0840 data: 0x7fd524046680 2013-04-23 14:40:59.234148 [DEBUG] switch_ivr_async.c:285 Digit parser HTTAPI: binding \d+/default/0 callback: 0x7fd528bf0840 data: 0x7fd5240466c0 2013-04-23 14:40:59.234148 [INFO] switch_ivr_async.c:194 Digit parser HTTAPI: Setting realm to 'default' 2013-04-23 14:40:59.234148 [ERR] switch_core_file.c:112 Unknown file Format [$] Thank you -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/http-cache-with-mod-httapi-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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PTIME 0.020 vs 0.030 and call quality
by Sean Devoy 23 Apr '13

23 Apr '13

Hi all, Does call quality improve when ptime is shortened? I have seen posts saying 40ms resulted in choppy audio. Clearly FS likes 0.020. I am trying to decide if I should go through and update my Cisco and Linksys devices to use 0.020. Thanks, Sean

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a way to disable hangup_after_bridge when an attended transfer happens?
by Michel Brabants 23 Apr '13

23 Apr '13

Hello, is there an elegant way to disable hangup_after_bridge when an attended transfer happens? Thanks, Michel

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