- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Error building RPM
by Worker Bee 05 Jul '13

05 Jul '13

Hello Everyone; I am new to FreeSwitch. I am attempting to create an rpm distribtion on CentOS6, Freeswitch version 1.2.9 I keep running into the following error: libtool: link: `/usr/lib64/libogg.la' is not a valid libtool archive I would be very appreciative if anyone could shed some light or point me in the right direction. Thanks! Bee

2 1

Question about placing call needed time
by Georgi Stefanov 05 Jul '13

05 Jul '13

Hello All, I have a small question about time needed to place call initiated with originate command. I hope someone to give me some clues, because I am stucked Here is the test FreeSwitch2 dialplan: public: blind transfer to default <extension name="public_did"> <condition field="destination_number" expression=""> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="user/${destination_number}"/> </condition> </extension> default: just play a pretty music <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="export" data="dialed_extension=$1"/> <action application="answer"/> <action application="playback" data="/usr/local/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav"/> </condition> </extension> FreeSwitch1 Using mod_event_socket I connect to FS1 and pass 20 commands like (20 connections): originating from FS1 to user to FS2 - I am using "originate sofia/internal/1000(a)10.101.3.60 6001" for user 1000 til 1020 As result I have 20 calls in FS2 This is cool for now :-) Here comes the question arised from looking FS2 logs: root@GeorgiStefanov:/usr/local/freeswitch/log# grep "NEW RTP SESSION CREATING" freeswitch.log 2013-07-05 17:46:34.897918 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.897918 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.897918 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.917913 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.917913 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.917913 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.917913 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.937911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.957911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.957911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING 2013-07-05 17:46:34.957911 [CRIT] switch_rtp.c:1907 NEW RTP SESSION CREATING root@GeorgiStefanov:/usr/local/freeswitch/log# grep "NEW RTP SESSION CREATING" freeswitch.log | wc -l 20 root@GeorgiStefanov:/usr/local/freeswitch/log# grep " Correct ip/port confirmed" freeswitch.log 2013-07-05 17:46:36.857908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:36.857908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:38.077915 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:38.077915 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:39.317915 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:39.317915 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:40.557916 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:40.557916 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:41.777917 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:41.777917 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:42.977908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:42.977908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:43.617914 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:43.617914 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:45.477914 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:45.477914 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:46.657910 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:46.677908 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:47.297931 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-05 17:46:47.297931 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. root@GeorgiStefanov:/usr/local/freeswitch/log# grep " Correct ip/port confirmed" freeswitch.log | wc -l 20 I have modified switch_rtp.c I have added log "NEW RTP SESSION CREATING" in very begging of function switch_rtp_create - just want to know when media connection preparation is started Does anybody can help me to understand the debug "Correct ip/port confirmed." info. I am worried about the not so much of these 10 sec latency, but the sequence I thought they should be paralel, not one after another. What is FS saying with "Correct ip/port confirmed."? Does FS respond in paralel to all calls?

1 0

Developing a new module
by Lists 02 Jul '13

02 Jul '13

Hi, I posted on the main list asking how to implement a modem using FreeSwitch. I had some helpful replies and I have since found that various existing modules include the bulk of what I need, in particular mod_fsk. I need to program remote analog phone devices that are programmed using data sent by V.23 and dtmf, similar to that used in the FSK caller id protocol. I decided to write a new module and took the mod_skel as recommended in the Authoring modules section of the documentation. I have that compiled and running as a skin for my app. My problem is that the documentation is somewhat fragmented and incomplete for a newbie like me, I'm really struggling to understand all of the complex issues around writing apps that use switch core functions and deal with rtp packets in real time. I'm part of a small startup team and would appreciate any help in getting up to speed quickly with this as we want to launch our new product soon. Can anyone recommend the best way to learn, other than scouring through other peoples code, I would be happy to pay for some support from another developer who could guide me along. I feel capable I will get there eventually, but I'm not sure I have the time to work it all out the hard way! All the best, paul(a)collisters.com lists(a)collisters.com Liverpool, England

1 0

FreeSWITCH Weekly News and Notes
by Michael Collins 02 Jul '13

02 Jul '13

Welcome to July! We've had lot's of great news<http://online.wsj.com/article/PR-CO-20130624-907435.html> lately with FreeSWITCH and now there's even more. Our friends at Yealink have reported on a very large VoIP installation in Spain that utilizes Yealink phones and FreeSWITCH servers. The complete writeup can be found here <http://www.yealink.com/News_info.aspx?NewsId=330&CateId=306>. We look forward to reporting on more success stories in the future. With the recent announcement <http://freeswitch.org/node/455> about FreeSWITCH 1.4 beta and WebRTC support there has been a lot of interest in just how to make this all work. On this week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_03> we will be taking a closer look at the code and showing you how to setup both sides of a WebRTC connection with FreeSWITCH. Please join usthis Wednesday at 1PM Eastern, 10AM Pacific for this interesting discussion. We also had an interesting discussion on last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_26>. Lorenzo Magani and Alexandr Dubovikov from the HOMER<http://www.sipcapture.org/> project gave us a tour of their new version 3.5 release. The HOMER SIP capture system is an excellent - and free - tool that can be put into production with any FreeSWITCH server. HOMER is an invaluable tool used for SIP diagnostics and troubleshooting and we highly recommend it for anyone using FreeSWITCH and SIP in a production environment. Have a great week and we'll talk to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

2 1

Github and FreeSWITCH
by Brian West 28 Jun '13

28 Jun '13

Please use http://jira.freeswitch.org to publish patches. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest

3 3

help to correct sip stack to send to kamailio in mod_sofia
by ThanhTruong 28 Jun '13

28 Jun '13

hi all, I am very new in sofia sip, and I am develop one register module for FS using sofia sip stack. in FS, when i receive REGISTER message, i am trying to send it to another registrar server (kamailio). I have implement my code in incoming REGISTER message like: nua_set_params(profile->nua, SIPTAG_FROM_STR("sip:101@192.168.1.112:5062"), NUTAG_ENABLEMESSAGE(1), NUTAG_ENABLEINVITE(1), NUTAG_AUTOALERT(1), NUTAG_SESSION_TIMER(0), NUTAG_AUTOANSWER(0), TAG_NULL()); nua_register(nh, NUTAG_PROXY("sip:192.168.1.112:5062"), NUTAG_REGISTRAR("sip:192.168.1.112:5062"), //registra uri SIPTAG_FROM_STR("sip:192.168.1.112:5062"), NUTAG_INSTANCE (0), NUTAG_M_USERNAME("101"), TAG_NULL()); I have one account 101/xxxx in kamailio, i want to register from softphone to FS, FS send register and process authentication with kamailio, then return 200OK back to softphone. With this code bellow, i send REGISTER message and get response in kamailio as: send 613 bytes to udp/[192.168.1.112]:5062 at 03:10:42.833439: ------------------------------------------------------------------------ REGISTER sip:101@192.168.3.122:42806;rinstance=7742cc70900998e8 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.112;rport;branch=z9hG4bKBB44UFQ9ySjQK Max-Forwards: 70 From: <sip:192.168.1.112:5062>;tag=DXeN4j2DFt27j To: "101" <sip:101@192.168.1.112>;tag=cd424d7d Call-ID: ac980e34-4e79-1231-71ba-3e508d3991e2 CSeq: 45210457 REGISTER Contact: <sip:101@192.168.1.112> User-Agent: FreeSWITCH-mod_sofia/1.5.2b+git~20130527T075833Z~f3541e00f4 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 recv 325 bytes from udp/[192.168.1.112]:5062 at 03:10:42.834220: ------------------------------------------------------------------------ SIP/2.0 404 Not here Via: SIP/2.0/UDP 192.168.1.112;rport=5060;branch=z9hG4bKBB44UFQ9ySjQK From: <sip:192.168.1.112:5062>;tag=DXeN4j2DFt27j To: "101" <sip:101@192.168.1.112>;tag=cd424d7d Call-ID: ac980e34-4e79-1231-71ba-3e508d3991e2 CSeq: 45210457 REGISTER Server: kamailio (4.0.1 (x86_64/linux)) Content-Length: 0 Then, I see that structure of SIP REGISTER message is not correct. Can some one help me to construct this message to send to kamailio ? May we dont care about nua_create ? just using nua_set_params or nua_register or nua_options ? Thank all in advance. ThanhTruong

3 3

ReINVITE failure with ACK+SDP scenario
by BNML 28 Jun '13

28 Jun '13

Hi all, I am having trouble with a transfert failure in a special scenario which i wasn't able to find documentation or configuration option solving it. I have a PBX creating two outbound call, it want to bridge them on the FreeSWITCH. It is sending a first INVITE without SDP and it will send the SDP in the ACK. (This is not breaking RFC3261 but I admit rarely seen scenario. http://tools.ietf.org/html/rfc3261#section-13.2.1 : "The UAC MAY choose to add a message body to the INVITE.". Looking at this scenario there is, indeed, at least one interesting scenario where it can helps avoiding creating another useless Re-INVITE transaction, see below) For the second call it is sending a re-invite with SDP. FreeSWITCH will understand and forward (because of "bypass_media") the second Re-INVITE but does never forward the first Re-INVITE (The one without SDP in the INVITE but in the ACK) to carrier, so this will end up in one-way-audio. A little schema to help : PBX FREESWITCH COMMENT 1) INVITE (a-leg) [THERE IS NO SDP] 2) 200 OK (fs-a-leg)[FreeSWITCH answer with the SDP information before any media update occurs, but my PBX is now aware of media informations] 3) INVITE+SDP (b-leg) [This is b-leg updated media information with previous 200 OK (fs-a-leg)] 4) 200 OK (fs-b-leg)[My PBX updated media and has b-leg freeswitch media information that it will send in the ACK+SDP to finalize the both-legs Re-INVITE] 5) ACK (b-leg) [Acknoledgement of b-leg Re-INVITE transaction] 6) ACK+SDP (a-leg) [ *** problem *** it assume FreeSWITCH will update media information and forward to the carrier which it won't] The solution i imagine is FreeSWITCH SHOULD trigger a Re-INVITE either for the a-leg at INVITE (without SDP in "1") receiption (and update media information when receiving the ACK+SDP) _OR_ create a Re-INVITE transaction at "6" when receiving ACK+SDP. One way or another, I am supposing FreeSWITCH refuse to forward the INVITE without SDP at first maybe because of carrier compatibility ? If not, this seems the way to go. Has anyone faced such a scenario ? Am I rigth assuming a patch is needed ? If yes, any suggestions are welcome. Thanks you BNML

1 0

Multiple ptime in SDP
by Wolfgang Pichler 28 Jun '13

28 Jun '13

Hi all, i did have a problem while doing a interconnection to a customer, with multiple ptime in sdp. His equipment did not accepted the sdp freeswitch offered *m=audio 52628 RTP/AVP 0 8 3 9 18 101 13* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-16* *a=ptime:20* *m=audio 52628 RTP/AVP 4 101 13* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-16* *a=ptime:30* * * *to avoid this - i have added sdp_m_per_ptime=false before bridge. * * * *I have done some readings about this - and it is not clear to me if freeswitch accts according to this draft http://tools.ietf.org/html/draft-garcia-mmusic-multiple-ptimes-problem-01* *It seems to be a mix between method 2 and method 6 - but i am not sure.* * * *Can anyone clearify which method freeswitch is using here ?* * * *br,* *Wolfgang Pichler*

3 2

free sweech problemm
by David Rot 27 Jun '13

27 Jun '13

Dear all We came across the following problem with a Free Switch Avaya telephone exchange: When calls are accepted continuously the software is responding correctly, however, when calls are accepted non-continuously (pauses between calls) then the software stops answering as if the Free Switch doesn't any calls. Only after restarting the Free Switch then calls are being responded as expected again. Does anyone know about this problem and can help us get over this? I attach the conf directory Thank you and regards

1 0

FreeSWITCH Weekly Conference Call Today
by Michael Collins 26 Jun '13

26 Jun '13

Hello all! We have the guys from the HOMER project joining us today. Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_26 Hope you can join us! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0