- FreeSWITCH-dev - FreeSWITCH Mailing Lists

FreeSwitch date and time conditions
by Max Machula 03 Nov '11

03 Nov '11

Hello, Hope someone can help me. I have a strange problem with date and time on my FreeSwitch. I have dialplan: <extension name="WW26"> <condition field="destination_number" expression="26"/> <condition field="year" expression="2011"> <action application="bridge" data="USER/26@$${domain}"/> ... skip ... </condition> </extension> Server time: # date Thu Nov 3 21:34:22 GMT 2011 However FreeSwitch can not confirm the year 2011. Log file: Dialplan: sofia/internal/07711567890(a)xx.xx.xx.xx parsing [public->WW26] continue=true Dialplan: sofia/internal/07711567890(a)xx.xx.xx.xxRegex (PASS) [WW26] destination_number(26) =~ /26/ break=on-false Dialplan: sofia/internal/07711567890(a)xx.xx.xx.xxRegex (FAIL) [WW26] year() =~ /2011/ break=on-false I get the same problem if I change field year to wday which set to 1-6 for example now. Any ideas would be much appreciated. Thanks, Max

1 0

Send Event/User Channel Variable from FreeSWITCH to RTMP client
by RaviRaj Mulasa 03 Nov '11

03 Nov '11

Hi Please let us know how to send an Event/ User Channel Variable from FreeSWITCH to a RTMP client. Thanks RaviRaj.

1 0

FreeSWITCH Conference Call Today - New Dialplan Feature, SIP 101
by Michael Collins 02 Nov '11

02 Nov '11

Hello all! In our conference call today we are going to discuss a cool new dialplan feature (better logical OR processing) and then Ken Rice and I will be discussing some basic SIP knowledge to help get everyone up to speed when helping newbies. Talk to you all soon! -Michael

1 0

Newbie Q
by P. Broennimann 02 Nov '11

02 Nov '11

Hi there I am new to FreeSwitch and I have a question... I'd like to record a call/session outputing same length .wav files (all .wav files to be exactly 30 seconds long). I browsed through the FreeSwitch sources from GIT and I found the module 'mod_snapshot'. Would this module be the best start/base for the functionality I am looking for resp. are there other modules/scripts that I should better check first? -> I tried to build the module in MSVC Express 2010 but I get errors -> Anyone was recently successful building it? Thx in advance, P.

3 5

joining LegA and LegB and to have the UUID_LegA and UUID_LegB accessible
by victor.velo＠videotron.ca 01 Nov '11

01 Nov '11

Hello, I tried to join two legs (A,B) using the "api uuid_bridge " + legA_uuid + " " + legB_uuid in ESL program. It seems to work, but I could not have access to legA nor to legB anymore after the bridge. Q: Is there a way to join the two legs that allows me to have access to the two legs after the bridge? Thanks in advance, Victor

2 1

mod_dingaling timing out
by Federico Beffa 31 Oct '11

31 Oct '11

Dear Developers, I'm using a fairly recent version of FS (pulled 20 Oct.) and I'm experiencing a timeout problem with mod_dingaling: With FS running on a fairly fast server (Intel Xeon CPU E31220) with Ubuntu 10.04 server, at an interval of approximately 3h (varying between 2h and 5h) I see mod_dingaling disconnecting (for some minutes) and reconnecting to google talk (client mode). To find out the root cause I've added some debugging lines in the code and have found that the reported reason is that the function "SSL_read" in the function "iks_recv" of the FS customized iksemel library returns SSL_ERROR_SYSCALL. Looking at errno and related message I get: errno = 110 (Connection timed out). I'm behind NAT, but due to the recently introduced "keep alive" feature of mod_dingaling that should not be the cause of the problem. The problem should also not be due to network problems as: * during a disconnect, from the same machine I could ping talk.google.com and "openssl s_client -connect talk.google.com:5222 -starttls xmpp -debug" replies correctly. * simultaneously running FS on a different machine as a different user shows the same problem, but at different points in time and sometimes (?) longer intervals. Reading the man page for SSL_read I learned that this function can also write to the socket (as SSL_write can read). I'm wondering if the use of poll() (in iks_recv) with only read flags in conjunction with SSL_read could somehow be responsible for this. I would appreciate any suggestion on how to investigate deeper the problem. Thanks, Fede

1 0

choppy sound with .wav file played in the background
by Han Xin 26 Oct '11

26 Oct '11

Hi all, I have another question to ask again :P When I use: switch_ivr_play_file(session, NULL, file_path, NULL); to play a file such as "misc/8000/sorry.wav" directly, the sound is so choppy. Am I doing the background sound playing the right way? Han --

3 6

mod_portaudio multiple endpoints support
by Moises Silva 25 Oct '11

25 Oct '11

Hello, I've been doing some changes to mod_portaudio sponsored by Comrex Corporation. The changes involve adding support for using multiple audio devices simultaneously, including devices with multiple channels (note that is not exactly the same as stereo). The changes are available at git:// github.com/moises-silva/mod_portaudio-endpoints.git Complete documentation on the new functionality in portaudio can be found in the sample configuration in the form of comments (conf/autoload_configs/portaudio.conf.xml). The portaudio changes are totally backwards compatible, existing configurations should keep working as expected. In addition to the new portaudio functionality there are a few other changes that were necessary to use the new functionality. All the changes described below: 1. mod_portaudio.c and pablio.c modifications to allow usage of multiple streams, channels and devices. 2. uuid_outgoing_answer API added to mod_commands.c, this API allows to answer outgoing channels. This may only be useful for portaudio endpoints since most, if not all, endpoints receive the answer notification using some sort of signaling message (200 OK for example). 3. Changes to switch_generate_sln_silence() in switch_resample.c to allow specifying -1 as a special divisor value meaning to zero-out the buffer completely. This option was required by Comrex too since the equipment they have seems to be amplifying the random low-level noise that switch_generate_sln_silence() currently generates by default and they don't like the "hissing noise" on the background while the call is answered and there is no ringback tone. I'd appreciate anyone who can review and comment. I just merged a few moments ago with latest git therefore reviewing should be easy with git diff adding the github repo as a remote. I'd like to get approval from the core FreeSWITCH team to merge this into mainstream. Needless to say I'll be available in the case any bugs are introduced in the portaudio code, virtually impossible, but just in case :-) Thanks, - Moy

6 8

Building FS on Windows7/64
by marco blaisou 24 Oct '11

24 Oct '11

Ok, after digging once more into the problem I decided to use a *Windows 7/32* machine. The built worked effortless!! A very nice run. Back to a *Windows 7/64*, I'm still having troubles. Some of the issue seems related to gawk while building libsofia_sip And it seems that once the issue occurs, you're stuck with the built.... In any case please note the following remarks: * 1) if you uses Windows sdk 7.1* edit fake-rfc2533.h and comment out : #ifdef nodef // *****add this #ifndef HAVE_STRUCT_ADDRINFO struct addrinfo { int ai_flags; /* AI_PASSIVE, AI_CANONNAME */ int ai_family; /* PF_xxx */ int ai_socktype; /* SOCK_xxx */ int ai_protocol; /* 0 or IPPROTO_xxx for IPv4 and IPv6 */ size_t ai_addrlen; /* length of ai_addr */ char *ai_canonname; /* canonical name for hostname */ struct sockaddr *ai_addr; /* binary address */ struct addrinfo *ai_next; /* next structure in linked list */ }; #endif /* !HAVE_STRUCT_ADDRINFO */ #endif // *****add this *2) in sofia_presence.c* if (mod_sofia_globals.debug_presence > 1) { int i; // to be added for (i = 0; i < argc; i++) {

2 1

Music on Hold gets choppy and cuts off intermittently when VAD is enabled on Freeswitch
by Jyotshna Cherukuri 24 Oct '11

24 Oct '11

Hi, I have 2 user agent SIP profiles set up on my Freeswitch server(one profile uses PCMU as its codec-prefs and other uses SILK@8000Hz) and I enabled VAD on PCMU sip profile. When I make a phone call which involves SILK-PCMU transcoding and if I put PCMU user ON HOLD, I am seeing an issue with MOH being cut off . It seems to quit after playing for a while , breaking through only transiently. It doesn't seem good to the party listening to MOH. This doesn't seem to happen when VAD is disabled though. I had to enable VAD as I was constantly hearing hissing background noise during conversation if VAD was not enabled. Are there any settings I can adjust in Freeswitch to disable VAD when wav files are bing played in general. Any help is greatly appreciated Thanks in advance Regards Jyotshna

2 1