September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

What means by *.SPEEX?
by Евгений Золотов 09 Sep '08

09 Sep '08

We have some number of questions in one letter: We make registration seance's audiostream simultaneously into a 2 different files, which have an extensions "wav" and "SPEEX". We use for this extension about such sort: <extension name="20024"> <condition field="destination_number" expression="^20024$"> <action application="set" data="bypass_media=false" /> <action application="set" data="RECORD_ANSWER_REQ="true" /> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2008, Altron Inc." /> <action application="set" data="RECORD_SOFTWARE=FreeSwitch" /> <action application="set" data="RECORD_ARTIST=Registration" /> <action application="set" data="RECORD_COMMENT=!!!" /> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.SPEEX"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="sofia/internal/${dialed_ext}@192.168.2.107:5062" /> <action application="answer" /> </condition> </extension> As result we receive simultaneously 2 files ( audio files are enclosed to the letter) $ ls -l 2008* -rw-r--r-- 1 olej olej 26880 Сен 5 14:13 2008-09-05-14-13-08_20024_1010.SPEEX -rw-r--r-- 1 olej olej 107752 Сен 5 14:13 2008-09-05-14-13-08_20024_1010.wav 1. File *.wav, as one would expect, contains RIFF header with metadata and can be played back with any matching tool, for example we used "sox" *.SPEEX file, in that sort as it is written with FreeSWITCH, contains, as we think looking for its size, compressed by speex codec. Is it really so? Project SPEEX community develops own tools (speexenc, speexdec) plaining, that their speex stream is located into OGG container and thus can be converted into *.wav or another ( with speexdec) and played back ( for example, with sox). 2. As it seems, *.SPEEX file can be played back with another extension by using FreeSWITCH playback application. Is it really so? If it isn't so, than with which 3-rd part tool we could play back *.SPEEX file for the control? 3. And final question: <action application="set" data="RECORD_STEREO=true"/> - does it pack 2 channels from A-leg and B-leg of bridge into one *.wav stream? Can we register only A-leg or B-leg with record_session application?? Thanks in advance! Evgeniy/

3 4

Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
by Anthony Minessale 09 Sep '08

09 Sep '08

nonetheless, can you please see my bugnote in jira. ===== can you edit /usr/local/freeswitch/conf/autoload_configs/iax.conf.xml and set the debug param to 1. Then reproduce the case where it's not working. We can see if the mod_iax is even getting the digits. ===== also be sure to sync to the current release by entering "make current" from the build root. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 11, 2008 9:36:22 AM Subject: Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi Anthony, I'm fairly sure that it's not an IAX problem. IAX2 messages are properly acked, sound (between IAX2 and SIP) is ok, etc, etc. The problem is (only) with DTMF, as far as I can tell. It's more likely to be in core or sofia_sip? Thanks, Birgit On 11/03/08 13:20, Anthony Minessale wrote: > Just to warn you we do not use mod_iax all that much. It was used as a > tool to test the core of FreeSWITCH in the early stages of development > and we just kinda left it hanging around since it does at least work. > We had to add a bunch of code to the stock iaxclient lib to make it > threadsafe etc because the lib is designed to handle 1 or 2 calls and we > are using it in a b2bua type env where hundreds of concurrent calls may > be possible. That said, since we more or less forked the code to avoid > generating support issues for the other soft-phones that use iaxclient, > you may want to check if there were any bugs in the real version of the > library that match your issue. > > http://sourceforge.net/projects/iaxclient > > > We've had little to no demand for IAX2 so we have not focused very much > on it. We do have strategies for creating a more robust and perhaps even > an "improved" implementation of the IAX2 protocol. Alas, it's pretty > much purely an academic exercise because there is little reason to > implement it. > > > > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> > To: Freeswitch-users(a)lists.freeswitch.org > Sent: Monday, March 10, 2008 10:52:40 AM > Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable > > Hi, > > We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). > > I'm experiencing problems with DTMF: > The IAX2 DTMF frames aren't always transferred into RTP packets. > Sometimes none of the frames are translated, sometimes only the first > one is omitted. All the IAX2 DTMF packets are ack-ed. > > Our setup: > PhoneFromHere.com <http://PhoneFromHere.com> applet (IAX2 protocol) > -> FreeSWITCH (ns2.westhawk.co.uk <http://ns2.westhawk.co.uk>) > -> SIP to SystemX (sofia/default/123(a)66.212.134.192) > > > I searched on jira.freeswitch.org <http://jira.freeswitch.org> for a bug > that could explain this, but > failed, so I submitted this as a bug, see > http://jira.freeswitch.org/browse/FSCORE-109 > > Has anyone else experienced the same or a similar problem? > > Thanks, Birgit > -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk> _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

8 11

How to do load balancing in freeswitch
by Krunal Patel 09 Sep '08

09 Sep '08

Hi, Here is the scenario: **Whatever call comes to freeswitch will be forwarded to terminators. **Now I would like to do load balance to send call to terminators as per there weight. For example there are two terminator having weight 75 & 25. So 75 % calls should be forwarded to first terminator & 25% should be forwarded to another. So please help me to resolve this. Thanks, Krunal Patel

1 0

FXS problem
by Luis Manuel Zuccolo 08 Sep '08

08 Sep '08

Hi, I'm usign an openvox card with 2 fxo modules and 1 fxs module. When I dial from fxs extension only take the first digit dialed and tried to route the call. This digit is not in dialplan, then the call is hung up. This happend only in FreeSWITCH snapshot (2008-08-31 and 2008-09-6). In release 1.0.1 work well. It's look like that don't take the option digit-timeout configured to 2000 in openzap.conf.xml Am I' missing something else? Any help about this would be appreciated. Thanks in advance. Luis Zuccolo Console logs: (FreesSWITCH release 1.0.1) 2008-09-06 18:31:13 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/3:1/45662530 [2f5b3615-e144-4ce0-9e75-7fd987373e3b] 2008-09-06 18:31:13 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing OpenZAP->45662530@default 2008-09-06 18:31:14 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/1:1/45662530 [f19ddc6a-ab75-4e5c-b881-d00918d4ef86] 2008-09-06 18:31:14 [WARNING] mod_openzap.c:1113 on_fxo_signal() Unhandled type for channel 1:1 2008-09-06 18:31:18 [NOTICE] mod_openzap.c:1097 on_fxo_signal() Channel [OpenZAP/1:1/45662530] has been answered 2008-09-06 18:31:18 [NOTICE] switch_ivr_originate.c:1298 switch_ivr_originate() Channel [OpenZAP/3:1/45662530] has been answered 2008-09-06 18:31:25 [NOTICE] mod_openzap.c:1218 on_fxs_signal() Hangup OpenZAP/3:1/45662530 [CS_EXECUTE] [NORMAL_CLEARING] (FreeSWITCH snapshots) 2008-09-06 18:38:04 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel OpenZAP/3:1/4 [a269cba9-9e5a-427a-8fb4-48df1ee490ad] 2008-09-06 18:38:04 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing OpenZAP->4 in context default 2008-09-06 18:38:04 [NOTICE] switch_ivr.c:1098 switch_ivr_session_transfer() Transfer OpenZAP/3:1/4 to enum[4@default] 2008-09-06 18:38:04 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-09-06 18:38:04 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup OpenZAP/3:1/4 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-09-06 18:38:04 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 11 (OpenZAP/3:1/4) Ended 2008-09-06 18:38:04 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel OpenZAP/3:1/4 [CS_HANGUP] My configs file are: FreeSWITCH 1.0.1 => /etc/openzap.conf/openzap.conf FreeSWITCH snapshot => ../conf/openzap.conf [span zt] name => OpenZAP number => 1 fxo-channel => 1 [span zt] name => OpenZAP number => 2 fxo-channel => 2 [span zt] name => OpenZAP number => 3 fxs-channel => 3 conf/autoload_configs/openzap.conf.xml config: <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="9"/> </settings> <analog_spans> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-analog-option" value="3-way"/> <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-analog-option" value="3-way"/> <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </analog_spans> </configuration> Yahoo! Cocina Recetas prácticas y comida saludable http://ar.mujer.yahoo.com/cocina/

2 1

Re: [Freeswitch-users] SIP, NAT and Amazon EC2
by Novak Joe 08 Sep '08

08 Sep '08

Hi, When I first tried setting up freeswitch on ec2 I attempted to do this through the quick and dirty wiki: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install and the ec2 wiki http://wiki.freeswitch.org/wiki/Amazon_ec2 This didn't work out for me. You also need to take a careful look at http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall then, for sip, setup a security group with the following ports enabled (easiest to do with elasticfox http://developer.amazonwebservices.com/connect/entry.jspa?externalID=609 ): udp 16384:32768 udp 4569 udp 5060 tcp 5060 udp 5080 tcp 5080 tcp 8000 udp 8000 make sure you make the security group, and apply it before you boot the instance. after this, the quick and dirty install guid worked just fine for me. you'll just need to tweak the default dialplan to your needs. cheers Hi Brian, Did you have any luck with the EC2 instances. I tried a few things, but I just cant get RTP to run through it. PLease let me know. Best Damon Brian West wrote: > Let me launch mine in 32bit and see what I can do over the weekend... > I had this working without a problem. ;/ > > /b > > On Sep 6, 2008, at 12:09 AM, Damon Brown wrote: > > >> Ive Tried the following with no success: >> >> internal.xml >> >> <param name="ext-rtp-ip" value="75.101.142.208"/> >> <param name="ext-sip-ip" value="75.101.142.208"/> >> >> Ive also tried just changing the vars.xml file. I also tried >> forwarding the incoming rtp connections to my test pc. all with no >> audio success. I am sure im doing something wrong I jsut cant find >> it. >> >> I found another suggestion on the wiki of creating a "double nat" >> that listens on 5090. That didnt change anything either, here is >> that information: >> >> <settings>

4 3

Setting hangup hook from Perl?
by Nic Wolff 08 Sep '08

08 Sep '08

How do I set a hangup hook function in a Perl application called from the dialplan? After looking at freeswitch.pm I tried these: freeswitch::setHangupHook( &on_hangup ); $session->setHangupHook( &on_hangup ); but both failed with errors. Thanks - Nic

3 5

post-hangup processing
by Novak Joe 08 Sep '08

08 Sep '08

Hi, I am new to freeswitch, and have successfully set it up with mod_python, and put together some IVR dialogues. It seems amazingly easy to use, however I have a question about running post-processing after a hangup. I would like to run some time-consuming post processing on the recordings I make, after I hang up and I'm not sure how best to approach this. I would prefer not to use the async module as I am not so familiar with twisted, and I note that at the moment I can hangup up the call and have no problem processing the data I've recorded, however I also note that if I don't hang up the call, my sip client stays on the line until the processing finishes. I tried to send session.hangup("1") before I run the post processing commands, but this doesn't seem to have any effect. It seems there must be a way to get around using twisted, but maybe not? Cheers

2 1

The last CHANNEL_DESTROY event was not sent by FS !
by Erol Akarsu 08 Sep '08

08 Sep '08

I am having a issue on CHANNEL_DESTROY event. I have sent 20 originate calls and receiving only 19 CHANNEL_DESTROY events instead of 20. Why am I not getting that event? Also, I have got 20 CHANNEL_HANGUP events, that is correct I have registered this event through event_socket Could you please help me on this? This is very critical in my application Thanks Erol Akarsu

3 4

SIP, NAT and Amazon EC2
by Damon Brown 08 Sep '08

08 Sep '08

Hello All, I have installed freeswitch on a computing cloud (Amazon EC2). My main network configurations are: (FreeSwitch ARI LAN ) <---> (WAN) <------{INTERNET} -------> (WAN) <----> (SIP Soft or Hard Phone LAN) I have pointed all of my clients to the external (5080) sip port .... No matter what I try STUN, Port Forwarding or the "Use Asterisk Method" settings in the directory, I get no audio. My guess would be the RTP side, but at this point I feel like I have done everything I can and still be a FreeSwitch Newb. Can anyone provide any in-site? Seems like a great system ... more directly the idea of no hardware dependency! Thanks in advance Damon

3 10

running FS in the Amazon cloud / other scalability options
by Brian B 08 Sep '08

08 Sep '08

Our FS app will be under near-zero load much of the time, and then spike with our large (highly customized) conference call applications. So, we'd like to investigate running it in an easily scalable environment, a la the Amazon cloud. Three questions 1) Are the current images available here *current*, can someone tell from the version numbers? I looked for version history on the wiki but couldn't find it. The amazon command line command: ec2-describe-images -x all -v |grep freeswitch yields the following three (repeated in the verbose mode) <imageLocation>rbuilder-online/freeswitch-0.0.1-x86_11952.img.manifest.xml</imageLocation> <imageLocation>freeswitch-fedora-6-svn-rev-9178/image.manifest.xml</imageLocation> <imageLocation>rbuilder-online/freeswitch-0.0.1-x86_11963.img.manifest.xml</imageLocation> IMAGE ami-999672f0 freeswitch-fedora-6-svn-rev-9178/image.manifest.xml 811137716590 available public x86_64 machine IMAGE ami-1be30672 rbuilder-online/freeswitch-0.0.1-x86_11952.img.manifest.xml 099034111737 available public i386 machine IMAGE ami-dae306b3 rbuilder-online/freeswitch-0.0.1-x86_11963.img.manifest.xml 099034111737 available public i386 machine 2) If not, does anyone want to help set up a current Amazon image? I'll provide the Amazon account if you can install FS in it. 3) Does anyone have other recommendations for services that can burst to high usage, without paying for high usage 24x7? I don't see VPS providers that can burst to much RAM/CPU for the few hours/month we'll need it. Thanks! -- =========================== Brian Burt 415-308-4258 mobile (best) freeswitch(a)brian-burt.com ===========================

6 7