December 2013 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Query regarding calling "reloadxml"
by Nezam Uddin 11 Dec '13

11 Dec '13

I am using xml_rpc web api from my web application I am creating extensions programmatically and then i want to call reloadxml/reloadacl so that those changes are picked up by the server. I have tried myip:myport/webapi/reloadxml and even myip:myport/webapi/sofia?profile internal rescan reloadxml but both these commands do not seem to reload my xmls.However,when i click "Reload XML" on the fusionpbx sip management portal i can see the new extensions working.So whats the proper command to fire which would trigger the reloadxml properly which would make my new extensions work. Thanks -- *Regards,* Nezam Uddin

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[Freeswitch-dev] How to create extensions programmatically through .net ?
by Nezam Uddin 09 Dec '13

09 Dec '13

I have recently setup the fusionpbx server with freeswitch and my scenario is off a mobile app registration which triggers a creation of an extension.My backed logic is based on .net framework. Can anyone direct me whether this can be done using Freeswitch.Managed.dll or it is simply done via web api?? Thanks again -- *Regards,* Nezam Uddin

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Re: [Freeswitch-dev] How to get value of the sample?
by Anton 08 Dec '13

08 Dec '13

Anthony Minessale <anthony.minessale <at> gmail.com> writes: > > > Not enough to go on from that .... > You probably want something more like. > > > { > int16_t *samples = (int16_t *) frame->data; > int x = 0; > > for (x = 0; x < frame->datalen / 2; x++) { > printf("[%d] ", samples[x]); > } > > printf("\n\n"); > > } > > > > > > > > > On Sat, Nov 9, 2013 at 4:28 PM, Anton <e322 <at> ya.ru> wrote:Hallo. Why, when I'm trying to get a sample of the sound, I always get the > same value -1? > For example, in the mod_avmd.c is the string: > frame = switch_core_media_bug_get_read_replace_frame(bug); > After this I add the following code > switch_log_printf( > SWITCH_CHANNEL_LOG, > SWITCH_LOG_NOTICE, > "data[0]=%d\n", ((int16_t *)(frame->data))[0] > ); > Then trigger the module mod_avmd and get a result in the console: > data[0]=-1; > And so on for all the modules that contain the string: > frame = switch_core_media_bug_get_read_replace_frame(bug); > independently of the level of sound in the channel. > Why is that? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting <at> freeswitch.orghttp://www.freeswitchsolutions.com > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp:// www.cudatel.com > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp:// wiki.freeswitch.orghttp://www.cluecon.com > FreeSWITCH-dev mailing listFreeSWITCH-dev <at> lists.freeswitch.orghttp:// lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- devhttp://www.freeswitch.org > > > -- Anthony Minessale IIFreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wireAIM: anthmMSN:anthony_minessale <at> hotmail.comGTALK/JABBER/ PAYPAL:anthony.minessale <at> gmail.com > IRC: irc.freenode.net #freeswitchFreeSWITCH Developer Conferencesip:888 <at> conference.freeswitch.orggoogletalk:conf+888 <at> conference.freeswitch.org > pstn:+19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting <at> 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-dev mailing list > FreeSWITCH-dev <at> lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > I ran some tests, including one that led You. Even tried the following conversion types: char * samples = (char *) frame-> data; and anyway, the result is a -1 for all x, from 0 to frame->datalen-1 My task was to remake the module mod_avmd, ie file mod_avmd.c. And this problem has put me into a dead end, to the point where I moved my job to another server. In other operating environments do not have this problem. I wrote about this in more detail in the Issues: http://jira.freeswitch.org/browse/FS-5944

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[Freeswitch-dev] endless recusive process fork during build of latest stable in centOS 6.4
by Nuno Reis 06 Dec '13

06 Dec '13

Hi guys. I've just tried to compile the latest stable 1.2v branch and I'm getting stuck on an endless process fork as seen here<http://pastebin.freeswitch.org/21724>(ps fax output) This is a freeswitch RPM production pipe system on a centOS 6.4 x86_64. The latest version successfully compiled here was commit (456f69a62b2fa4df7ac8438b30ba68e74c2a04fc). Should i open another Jira for this? It looks like FS-5956 already addresses this issue or something quite similar. Looking forward to hear from you. BR, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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[Freeswitch-dev] Accessing Custom Header in REGISTER packets
by Mark Tsai 04 Dec '13

04 Dec '13

Hi All, In my REGISTER packets that's sent towards a FreeSwitch server, I am adding a custom header "X-CustomHeaderName: auth_token". The goal for my project is to access that token and conduct a verification through another API before allowing the REGISTER to pass. What's the best way to accomplish this? thanks, Miz

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[Freeswitch-dev] FreeSWITCH erroneous RTP timestamps in front of sofsip_cli client?
by Jean-Paul Iribarren 04 Dec '13

04 Dec '13

Hi FS wizards, first message on this list... Apologies if this post belongs to another section (please tell me if this is the case). First, a big thank you to all members of the team for this amazing piece of software. Great job here. Now, on for my issue: I am currently testing a Linux sofsip_cli SIP client on an embedded device, connected to a Win32 FreeSWITCH version, and I can't get correct RX audio on my platform. I ran the same test with a linphonec client on the same embedded device for comparison purposes, and the rx audio is good. Both clients end up using the same RTP configuration (PCMU/8000). Preliminary conclusion: my sofsip_cli client is buggy. But there is more... I ended up checking data exchanged between endpoints using Wireshark, and I was surprised to see that in front of sofsip_cli, FreeSWITCH seems to timestamp TX RTP packets incorrectly: it sends 160 samples/RTP packet, but the timestamps of consecutive TX packets are increased in 320 units steps, and FS sends 25 RTP packets per second on average. Consecutive RTP packets actually contain consecutive PCMU data (from the audio point of view), resulting in a sloooow, chopped rx audio coming out of the loudspeaker of my device. Using Wireshark embedded player to play the VoIP audio results in the same slow / chopped audio, which seems to prove that FS actually sent bad RTP. On the other hand, when I use linphonec in the very same conditions, I can see that FreeSWITCH sends 50 RTP packets per second on average. Each RTP packet contains 160 samples, and timestamps of consecutive are increased in 160 units steps, resulting in correct RX audio coming out of the loudspeaker of my device. FS logs in console window are the same for both cases, below is what I get for the sofsip_cli case: 2013-11-21 09:35:00.549253 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1009(a)10.0.0.57 [527f855e-3856-4805-b72c-da59b3a6d292] 2013-11-21 09:35:04.502530 [INFO] mod_dialplan_xml.c:558 Processing 1009 <1009>->1001 in context default 2013-11-21 09:35:04.533781 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *1 execute_extension::dx XML features 2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *2 record_session::C:/Program Files/FreeSWITCH/recordings/1009.2013-11-21-09-35-04.wav 2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *3 execute_extension::cf XML features 2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *4 execute_extension::att_xfer XML features 2013-11-21 09:35:04.565032 [NOTICE] switch_ivr_originate.c:2699 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2013-11-21 09:35:04.565032 [NOTICE] switch_ivr_originate.c:2699 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2013-11-21 09:35:04.580658 [INFO] mod_dptools.c:3201 Originate Failed. Cause: USER_NOT_REGISTERED 2013-11-21 09:35:04.580658 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1009(a)10.0.0.57! 2013-11-21 09:35:04.580658 [NOTICE] mod_dptools.c:1225 Channel [sofia/internal/1009(a)10.0.0.57] has been answered 2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1055 New Channel loopback/app=voicemail:default 10.0.0.57 1001-a [f2aa95f1-0631-4590-9fa4-c4dc1c96c88e] 2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1053 Rename Channel loopback/app=voicemail:default 10.0.0.57 1001-a->loopback/voicemail-a [f2aa95f1-0631-4590-9fa4-c4dc1c96c88e] 2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1055 New Channel loopback/voicemail-b [fc478cfd-e932-450a-9f50-2a5a76acc907] 2013-11-21 09:35:05.611948 [NOTICE] mod_loopback.c:947 Pre-Answer loopback/voicemail-a! 2013-11-21 09:35:05.611948 [NOTICE] mod_dptools.c:1260 Pre-Answer loopback/voicemail-b! 2013-11-21 09:35:11.549676 [NOTICE] sofia.c:715 Hangup sofia/internal/1009(a)10.0.0.57 [CS_EXECUTE] [NORMAL_CLEARING] 2013-11-21 09:35:11.565301 [NOTICE] switch_ivr_bridge.c:733 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2013-11-21 09:35:11.565301 [NOTICE] mod_loopback.c:553 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] 2013-11-21 09:35:11.580927 [NOTICE] switch_core_session.c:1598 Session 5 (loopback/voicemail-a) Ended 2013-11-21 09:35:11.580927 [NOTICE] switch_core_session.c:1602 Close Channel loopback/voicemail-a [CS_DESTROY] 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1598 Session 4 (sofia/internal/1009(a)10.0.0.57) Ended 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1602 Close Channel sofia/internal/1009(a)10.0.0.57 [CS_DESTROY] 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1598 Session 6 (loopback/voicemail-b) Ended 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1602 Close Channel loopback/voicemail-b [CS_DESTROY] I ran my tests by calling an unregistered extension (1001), therefore resulting in FS sending voice messaging audio, and hanging up on client after a few seconds. I have compared the SIP handshakes for both clients, and I can't see any significant difference (except that linphonec supports several codecs, while sofsip_cli only supports G.711/PCMU). Regarding FS config (this may or may not be relevant), I had to tweak it a bit to support sofsip_cli. I added the two following lines in the dialplan for the relevant extensions (file default.xml): <action application="set" data="disable_rtp_auto_adjust=true"/> <action application="set" data="zrtp_secure_media=false"/> I also changed the two following lines to add video codecs in file vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM,H263,H263-1998,H264"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,H263,H263-1998,H264"/> The rest of the config is plain vanilla. The (small) Wireshark traces of both tests can be downloaded from Box: - linphonec trace (FS TX RTP ok): file linphone.dac.pcap is available at https://app.box.com/s/63p8ayrr3dc6w1d9wm33 - sofsip_cli trace (FS TX RTP ko): file ssc.pcap is available at https://app.box.com/s/z1cde37xg5teh9d690oy My device has IP address 10.0.0.186, and FS is installed on my PC (running Win XP Pro) at 10.0.0.57. My FreeSWITCH version is "1.5.6b+git~20131101T202135Z~2589bf7750~32bit (git 2589bf7 2013-11-01 20:21:35Z 32bit)" Sorry for the long post (I tried to give all relevant info), and thank you in advance for your comments and help on this weird issue. -- JPI

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[Freeswitch-dev] Connect to SIP Trunking
by Roamer Roamer 02 Dec '13

02 Dec '13

Hi all, I'm new to freeswitch and want to learn SIP development, I have a question: what's protocol is using for the SIP PBX(freeswitch, asterisk, sipx) connect to SIP Trunking? In freeswitch source code, which part source code is for connect to SIP trunking ? Thank you in advance.

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