June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to enable ESL for ruby?
by dujinfang 29 Jul '09

29 Jul '09

Hi, Following the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library On MacOSX 10.5, I can't get ESL for ruby work. make throws error: sevens-mac-pro:~/workspace/test/freeswitch/trunk/libs/esl$ make rubymod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/Users/ seven/workspace/test/freeswitch/trunk/libs/esl/src/include - DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- prototypes -Wmissing-prototypes" CXXFLAGS="-I/Users/seven/workspace/ test/freeswitch/trunk/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb - I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C ruby g++ -I/Users/seven/workspace/test/freeswitch/trunk/libs/esl/src/ include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686-darwin9 -c esl_wrap.cpp -o esl_wrap.o g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L. Undefined symbols: "_main", referenced from: start in crt1.10.5.o ld: symbol(s) not found collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 Thanks for any help.

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setInputCallback not working with Javascript?
by mszlazak＠aol.com 24 Jul '09

24 Jul '09

I'm getting in build 12653M: [ERR] notify.js:130 mod_spidermonkey()? TypeError: session.setInputCallback is not a function The wiki says this function should work in Javascript. http://wiki.freeswitch.org/wiki/CoreSession_Constructor#session:setInputCal… Also, has there been changes to session.collectInput with type="event"? I get dtmf type events with my callback function but can't seem to get type="event" with speech events. Mark.

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Inbound call routing help
by Lars Zeb 23 Jul '09

23 Jul '09

I want to setup a dialplan for a single DID. I would like it to go to a specific extension, and if not picked up in 15 seconds, go to voicemail. I have set this scenario up and it works. But I would also like this person to be able to call this DID from outside FS via a phone and be able to retrieve their voicemail. I've seen the example of how to pick up an extension's voicemail while inside FS by checking to see if the destination_number is the same as the caller_id_number, and if so, listen to voicemail, otherwise leave the message with voicemail. But I don't have a clue how to accomplish this from outside, other than dedicating another DID to solely retrieving voicemail from outside. Any ideas? Thanks, Lars

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mod_dingaling no audio
by Jingwei Yang 14 Jul '09

14 Jul '09

Hi Guys, Here's my situation: The freeswitch server and my machine are behind the same LAN. If I commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by *originate dingaling/gmail.com/userAAA@gmail.com &echo*). However, external calls have no sound at all no matter whether this param is commented out or not. Please kindly let me know what other params to set to resolve this issue. Thanks, -Jingwei

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PCMU fallback for T.38
by Gabriel Kuri 01 Jul '09

01 Jul '09

hey folks, I'm trying to configure PCMU fallback for T.38. The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with G729 and PCMU in the sdp. the INVITE to the provider includes G729 and PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... m=audio 16458 RTP/AVP 18 0 100 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 once the FAX tone is detected on the PSTN side, FS receives a T.38 re-INVITE from the provider and FS sends back a 488/Not Acceptable (proxy_media=false). at that point the provider than attempts fallback to PCMU with another reINVITE ... m=audio 16816 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 immediately after the PCMU reINVITE, FS closes the channel and the text below is in the FS logs. given the SPA-2102 included PCMU in the original INVITE, even though it was the second preferred codec, shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by the provider? It seems like it's not passing the PCMU Re-INVITE back to the endpoint (SPA-2102), since it originally negotiated G729 with the SPA2102 as that was the 1st codec in the sdp, but trying to transcode between the two (G729 and PCMU)? 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 PCMU/8000 20 ms 160 samples 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1. 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing Reinvite 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 entering state [completed] 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! Thanks, Gabe

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Any advances on T.38 support for FS?
by François Delawarde 01 Jul '09

01 Jul '09

Many issues on Asterisk's T.38 (or probably just on T.38?)... Could it convince those relying on this "modern" version of a 50yo technology to switch to and with FreeSwitch? François.

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Testing Freeswitch performance led to strange behavior
by Apostolos Pantsiopoulos 01 Jul '09

01 Jul '09

In the process of trying to use Freeswitch in a production environment I conducted a number of performance tests using various servers. It was then that I noticed some strange behavior from FS. When I stripped down the scenario I was using to a simple bridge scenario, I stumbled upon a strange behavior. The scenario as I stated is quite simple. |---------| |---------| | |------- Call from sipp------> | | | sipp | | FS | | | <------ Call back to sipp----| | |---------| |---------| I did not use an RTP stream for my calls just to test the signaling alone. The sipp scenario is the standard uac.xml scenario that can be found integrated to sipp with the following options : Test FS 1: sipp <FS_IP>:5060 -s 55555555 -i <SIPP_IP> -mi <SIPP_IP> -ci <SIPP_IP> -r 10 -d 5000 -l 100 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~(35-60) % (35 during the generation of new calls, 60 during the -l limit imposed by the test) Note : 985 of them had a duration (billsec) of 10 and 15 of them had a duration of 11. I tried raising the call rate and limit... Test FS 1: sipp <FS_IP>:5060 -s 55555555 -i <SIPP_IP> -mi <SIPP_IP> -ci <SIPP_IP> -r 20 -d 5000 -l 200 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~0-30 % (0 during the generation of new calls, 30 during the -l limit imposed by the test) THIS IS WHAT MAKES ME WONDER : The distribution of the durations (billsec - not complete durations) : 183 calls with 10 secs billed duration 110 calls with 11 secs billed duration 238 calls with 12 secs billed duration 447 calls with 13 secs billed duration 22 calls with 14 secs billed duration The sipp scenario is simple "hangup the phone after 10 secs". So, why am I seeing these? Of course that has something to do with the stress the machine has been put through during the second test. But I can see it happening to less stressful conditions (i.e. 15 calls per second) to a smaller extend. I captured one of these calls and verified that when the sipp client hangs up exactly 10 secs after the call start, FS receives the BYE and replies with 200 OK. BUT it does not hang the second leg in a timely manner i.e. it sends a BYE to the sipp server side 1-4 seconds AFTER that. That explains the 11, 12, 13, 14 secs durations seen on the second test. What is more interesting is that I would expect to see in the CDRs the first and second leg to have different durations (since the a leg BYE was received and aknowledged by FS in the correct time) i.e. 10 and 14 secs accordingly. But what I get is the same duration for both legs (14 secs). This in my opinion is very dangerous on production environments (you get charged by your provider more seconds that you charge your clients - or - you falsely charge your clients with bigger durations although they hunged up corectly (and you acknowledged it)). NOTE No 1 : All the performance recommendations found in the wiki has been applied. In fact only the essential modules that could make this scenario work were loaded. NOTE No 2 : I tried using asterisk (as a point of reference - don't get me wrong - I am not trying to start a flame war here). And it succeeded doing on the same hardware 60 calls/sec with a channel limit of 400 sim. calls using only 50% of the cpu (maximum). No under any circumstances I have seen the behavior above (this inability to hang call legs in a timely manner). Even when I pushed asterisk to the limits (80 calls per second 600 max call limit) and it started failing on some calls it never failed to hangup the calls for both legs on exactly 10 secs. NOTE No 3 : As you can tell I was using a very small machine for my tests. When I moved the same tests to larger installations (Quad Core Opterons and Xeons) I got proportional results to the above. NOTE No 4 : The tests were performed in a LAN environment and since there was no RTP involved I think there were no bandwidth issues there. NOTE No 5 : The tests were performed using numerous SVN versions (latest : 13610), the stable version and the 1.0.4pre8 version. NOTE No 6 : Using the -hp switch made no noticeable change in behavior. I am not trying to complain for FS's performance (far from it). I am just somewhat disappointed seeing it performing in such a strange manner when under stress. I would prefer a design that drops the calls after a certain threshold than a design that incorrectly handles them all (I am aware of the max sessions per second in switch.conf.cml - but I am starting to see this behavior even with the cpu idling at about 80%). I don't know if anyone else had the same experience when testing Freeswitch. I can happily supply with all the test details (config files, captures etc) to all interested parties. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Cant register a pointer. What wrong?
by Alexey Lubimov 01 Jul '09

01 Jul '09

I sofia_reg.c:1765 have two user records - good #110 and bad #111. bad.xml: <include> <user id="111"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="$${default_vm_password}"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="111"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Bad"/> <variable name="effective_caller_id_number" value="111"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> good.xml: <include> <user id="110"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="$${default_vm_password}"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="110"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Good"/> <variable name="effective_caller_id_number" value="110"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> Good user 110 work without any problem. But user "Bad" user 111 can't register to freeswitch. In log I can see only one message - 2009-06-30 16:31:36.590970 [WARNING] sofia_reg.c:1765 Cant register a pointer. good user exists: freeswitch@internal> user_exists id 110 neolant.ru true and bad user exists! freeswitch@internal> user_exists id 111 neolant.ru true good user don't have attr type: freeswitch@internal> user_data 110(a)neolant.ru attr type -ERR no reply but bad user have attr type! : freeswitch@internal> user_data 111(a)neolant.ru attr type pointer Good user have password: freeswitch@internal> user_data 110(a)neolant.ru param password 123456 But bad user no have param pasword! freeswitch@internal> user_data 111(a)neolant.ru param password -ERR no reply What's wrong in these configuration? How I can debug and resolve these problems?

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Error in OpenZap
by Baskar 01 Jul '09

01 Jul '09

*Hi, **I have installed the latest trunk with A102D Sangoma card when i load the openzap i get this error ** freeswitch(a)localhost.localdomain> load mod_openzap 2009-06-30 14:47:33 [NOTICE] zap_io.c:2626 zap_global_init() Modules configured: 1 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '1' API CALL [load(mod_openzap)] output: -ERR [module load file routine returned an error] 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:1-15' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '1:16' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '1:17-31' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'name' / 'OpenZAP' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'number' / '2' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'trunk_type' / 'E1' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:1-15' freeswitch(a)localhost.localdomain> 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'd-channel' / '2:16' 2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config() unknown param [] 'b-channel' / '2:17-31' 2009-06-30 14:47:33 [INFO] zap_io.c:2370 load_config() Configured 0 channel(s) 2009-06-30 14:47:33 [ERR] zap_io.c:2633 zap_global_init() No modules configured! 2009-06-30 14:47:33 [ERR] mod_openzap.c:2401 mod_openzap_load() Error loading OpenZAP 2009-06-30 14:47:33 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so **Module load routine returned an error** How can i resolve the above problem. how can i load the openzap My Configuration files default.xml <extension name="Long Distance - wanpipe"> <condition field="destination_number" expression="^9(\d{5,15})$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="OpenZAP/1/a/${dialed_ext}"/> </condition> </extension> /etc/wanpipe/wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 0 PCIBUS = 2 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF = NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <pri_spans> <span id="1"> <param name="mode" value="user"/> <param name="dialect" value="euro"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> /usr/local/freeswitch/libs/openzap/conf/openzap.conf [span wanpipe1] number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 **How can i resolve the above problem. how can i load the openzap. can some one assist me to resolve this process.* * -- Thanks with Regards, N.Baskar *

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ClueCon 2009 - Great News: Early Bird Extended
by Michael Collins 30 Jun '09

30 Jun '09

Great news for all of those who've not been able to sign up yet for ClueCon: we've decided to extend the early bird special through Tuesday July 21! Note: this is also the last day you can book a room with the Wyndham, but please don't wait until the last minute. More good news and reasons to sign up right away: fun giveaways! Our sponsors and partners are supplying goodies to give away to conference attendees. Here are some highlights: >From Sangoma - B600 cards (qty 3) and A101D (qty 1) >From Snom - 20 phones, mix of models 320, 360, 270 Stay tuned for more updates and surprises about ClueCon 2009! -Michael http://www.cluecon.com 877.742.CLUE

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