October 2011 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Epoch current time difference
by Glaucio Vaz de Mello 10 Jan '12

10 Jan '12

Hello, I noticed a difference between FreeSWITCH time and system clock time. Is this a standard behaviour? or something is wrong in my system? More details of the issue: Inside FreeSWITCH I look for current epoch time using this function: time_t now = switch_epoch_time_now(NULL); Then I look for system clock current time using: #include <time.h> time_t now = time(NULL); Between both results I noticed a difference of 9 seconds. This can also be viewed using Unix commands. Unix date command from console: # date +%s 1314741992 Then, I start a call and make a DB query: # sqlite3 /var/freeswitch/db/core.db "SELECT call_created_epoch FROM calls;" 1314742000 Then I repeat unix date command: # date +%s 1314741994 Any comments are welcome. Regards, Glaucio

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[Freeswitch-dev] 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

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[Freeswitch-dev] 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

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[Freeswitch-dev] 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 --

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[Freeswitch-dev] 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

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[Freeswitch-dev] 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++) {

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[Freeswitch-dev] 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

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Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
by Visciano Giovanni 24 Oct '11

24 Oct '11

Preconditions on sip profile ---------------------------- inbound-codec-prefs NOT set outbound-codec-prefs NOT set inbound-proxy-media=TRUE inbound-bypass-media=FALSE inbound-late-negotiation=TRUE + inherit_codec=TRUE TEST scenario ------------- SIP phone send INVITE with SDP to FS INVITE SDP[codelist=PCMA, PCMU, G722, G729] --> FS FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR, L16] The "ep_codec_string" channel variable is loaded with [PCMA, PCMU] EXPECTED -------- In "proxy-media + late-negotiation" scenario: ep_codec_string=[PCMA,PCMU,G722,G729] PROBLEM ------- According to the documentation (http://wiki.freeswitch.org/wiki/Codec_negotiation) "ep_codec_string: This variable is only available if late negotiation is enabled on the profile. It's a readable string containing all the codecs proposed by the calling endpoint." and "inherit_codec=true (only applicable when late-negotiation is enabled) will take the codec negotiated when the B leg answers and pass it to the A leg so it also uses the same codec (if the A leg supports it); otherwise, it will use whatever it can from its own list" The variable ep_codec_string doesn't contain ALL the codecs proposed by the calling endpoint. It contains the intersection between the codecs proposed by the calling and the codecs loaded in FS. In proxy-media / no-media configuration the called endpoint doesn't have the calling codec list so he can't negotiate using all the capabilities available. (I'm working on a test endpoint module that should work in proxy and no-media configuration ... mod_sofia vs mod_sofia works because they look at the remote sdp variable) Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string() Regards, Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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[Freeswitch-dev] building on windows 7/64 - follow up
by marco blaisou 24 Oct '11

24 Oct '11

As a follow up of previous post, the issue is confirmed to be located at the prebuilt phase of libsofia_sip_ua_static lib ie at *if not exist "$(ProjectDir)..\..\sofia-sip\win32\gawk.exe" cscript /nologo "$(ProjectDir)..\util.vbs" Get http://files.freeswitch.org/downloads/win32/gawk.exe"$(ProjectDir)..\..\sofia-sip\win32\" cd /D "$(ProjectDir)..\..\sofia-sip\win32\" set AWK="$(ProjectDir)..\..\sofia-sip\win32\gawk.exe" "autogen.cmd"* What seems to happen is that after gawk.exe is loaded, the prebuilt is trying to go on while concurrently the anti-virus/protection is checking the gawk.exe -> this creates a mess and concludes with a failure and then missing headers and so on. *What I've done is simply to preload gawk.exe in the target directory, declared it as safe to the AV/Firewall and only then launch the VS built.* So as a suggestion, loading gawk ahead (ie at thbeginning of the build) could be a way btw. The previous discussed recommendations (changes) related to SDK 7.1 are valid

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[Freeswitch-dev] FS-2877 patch
by Beeton, Carolyn (Carolyn) 24 Oct '11

24 Oct '11

Anthony, I see that the patch from FS-2877 has been committed (thanks! I hope it helps to address the BLF issues others have seen). I did have a comment on that issue that there was a typo in the patch which I don't think was corrected in the commit. The typo will cause the database to be dropped and rebuilt on every run. This would fix it: diff --git a/src/mod/endpoints/mod_sofia/sofia_glue.c b/src/mod/endpoints/mod_sofia/sofia_glue.c index e6076e8..a6de2f5 100644 --- a/src/mod/endpoints/mod_sofia/sofia_glue.c +++ b/src/mod/endpoints/mod_sofia/sofia_glue.c @@ -5974,7 +5974,7 @@ int sofia_glue_init_sql(sofia_profile_t *profile) switch_cache_db_test_reactive(dbh, test_sql, "DROP TABLE sip_subscriptions", sub_sql); free(test_sql); - test_sql = switch_mprintf("delete from sip_dialogs where hostname='%q' and expires <> -9999 or rpid='' or from_tag=''", mod_sofia_globals.hostname); + test_sql = switch_mprintf("delete from sip_dialogs where hostname='%q' and expires <> -9999 or rpid='' or sip_from_tag=''", mod_sofia_globals.hostname); switch_cache_db_test_reactive(dbh, test_sql, "DROP TABLE sip_dialogs", dialog_sql); Carolyn

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