May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_managed session.recordFile() problem
by babak yakhchali 23 May '10

23 May '10

Hi all somebody save me plz! I'm using the code below to simply record messages: Session.Answer() ... if (Session.Ready()) { Session.Execute("sleep", "100"); Session.StreamFile(..., -1); Session.Execute("playback", "tone_stream://%(500,500,480,620)"); Session.RecordFile(..., 60000, 500, 5); } it's working on my pc with no problem but when I upload it to the server(which is much much much stronger than my pc) first 5 6 sec of message is not recorded thanks

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Re: [Freeswitch-users] confirm 155a8a12b77a3382db745b073f0d8f6a37c95f5d
by Alejandro Duque Rueda 22 May '10

22 May '10

Hello, I am working with freeswitch on Open wrt. I have 2 modules that doesn't exists information in the wiki: mod_portaudio_stream and mod_alsa. Someone can tell me for what are those modules, and if someone have worked with them. Thank you

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Conference command "relate", nospeak/nohear
by Peter Olsson 22 May '10

22 May '10

Hello everyone, I have a small question about the mod_conference relate command. What's the expected behaviour of "nohear"? Is it supposed to work as nospeak, but in the other direction? When running "conference xx relate member_x member_y nospeak", the audio from member_x to member_y is muted, but when using nohear I didn't notice any difference at all (I could still hear audio everywhere). Is it supposed to work the opposite way, so when using nospeak|nohear as parameter, the audio from x to y is muted, and also the other way around? /Peter

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Re: [Freeswitch-users] Quick question on Originate through api
by Kristian Kielhofner 22 May '10

22 May '10

AFAIK the % let's Freeswitch know it's a local domain and doesn't need to do any DNS lookups, etc. While it was never really necessary to use % it made a lot sense for various reasons. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Sent: Fri May 21 12:07:17 2010 Subject: Re: [Freeswitch-users] Quick question on Originate through api not necessary .... ----- Original Message ----- From: "Peter Olsson" <peter.olsson(a)visionutveckling.se> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Friday, May 21, 2010 11:52 AM Subject: Re: [Freeswitch-users] Quick question on Originate through api For sofia/internal/ you should use a % as a separator I think, instead of @-sign. /Peter ________________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] för Phillip Jones [pjintheusa(a)gmail.com] Skickat: den 21 maj 2010 17:41 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] Quick question on Originate through api Hi there, In my mind - where 6775557767 is a registered user, the following two are the same: apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}user/6775557767 ' apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767(a)121.763.76.76 ' i.e. user/ and sofia/internal are the same thing. However on my system the user/ works great but sofia/internal loops call back through in the public context. Can someone tell me the difference between these options? What am I missing? Thanks Phil !DSPAM:4bf6abb932934294689129! _______________________________________________ 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 _______________________________________________ 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

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CDRs for Gateways
by Code Ghar 22 May '10

22 May '10

Hi David I looked at what you suggested. Before your suggestion, in /usr/local/freeswitch/conf/autoload_configs/cdr_csv.conf file, I had the following lines   After your suggestion, I changed it to   and then did a reloadxml. Now all CDRs were created in /usr/local/freeswitch/log/cdr-csv/Master.csv and CDRs for inbound calls from unregistered sources (i.e. external gateways) were also created in this file. Thanks for your help. Code, I think you have a major issue or configuration error with your FS setup. FS, as any B2BUA, generates a CDR for any inbound leg, either from a registered user or from a non-registered gateway. It doesn't care where it comes from, it just does it. I can barely imagine a softswitch that would not write the CDR matching a call that was routed through it. The CDR for the outbound leg is optional. Can you confirm you are looking in Master.csv ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 21/05/2010 à 05:25, Code Ghar a écrit : > Using mod_cdr_csv, it is easy to create CDRs for registered users. > However, how can we get CDRs for external gateways? For example, > FreeSWITCH is configured to receive calls from one gateway and send > them out through another. In this scenario, when neither inbound nor > outbound gateway registers itself to FreeSWITCH, how can we generate > and store CDRs for inbound gateway? > > Similarly, let's say a SIP provider sends a call meant for a > registered user. For example, if a cell phone calls 6175550000, it > is routed through the telephone network to the SIP provider. This > provider routes the call to FreeSWITCH, which looks up in its dial > plan that this number should be routed to extension (or registered > user) 1000. There are no CDRs generated for either the SIP > provider's gateway or the registered user in this case. > > In both these scenarios, if a call is inbound to FreeSWITCH but has > not been generated by a registered user, mod_cdr_csv does not create > CDRs. I have even tried to set the following parameter in > cdr_csv.conf.xml but it didn't help. > > <param name="legs" value="ab"/> > > My question is this: can mod_cdr_csv generate CDRs for inbound calls > from sources other than registered users? If it can't, how can this > be achieved? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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mod_dptools/redirect and 300 Multiple Choices
by Carlos Romero 21 May '10

21 May '10

what do you mean that you don't have access to it? you have the code On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote: > not without intervention from the sofia devs, > Sofia sip does that deep down and we don't have access to it. > even when we supply it as a one big string it renders it out depending on > those settings. >

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mobisents
by Jan Berger 21 May '10

21 May '10

Hi, Is anyone working on a FreeSWITCH RA for mobicents? I just realised that mobicents already have sccp, tcap, inap, camel, map + a few other things. They already have a Asterisk RA. Jan

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Quick question on Originate through api
by Phillip Jones 21 May '10

21 May '10

Hi there, In my mind - where 6775557767 is a registered user, the following two are the same: apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}user/6775557767 ' apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767(a)121.763.76.76' i.e. user/ and sofia/internal are the same thing. However on my system the user/ works great but sofia/internal loops call back through in the public context. Can someone tell me the difference between these options? What am I missing? Thanks Phil

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ISDN to SIP via Freeswitch - 408 Timeout and Terrible line noise
by samwise 21 May '10

21 May '10

Hi, I've just joined a team which had an Asterisk 1.6.0.1 desktop machine with a Digium TE122 card running CentOS 4.7 to bridge calls from, as far as I can tell, an E1 PRI line (with 10 DDI numbers) and software SIP UAs. Unfortunately, the machine died recently and will no longer boot from hard disk. So I've taken the card out and placed it in a HP Compaq ProLiant ML350 G3 Server with 1 GB RAM running CentOS 5.4 (Final). I've set it up to run the Freeswitch 1.0.6 release with the OpenZAP module enabled and pre-requisites dahdi-linux-complete-2.3.0+2.3.0 and libpri-1.4.10.2. All these components were built from source packages. I installed the default configuration files and configured everything as far as I could by following the instructions on the Freeswitch wiki. I've managed to get as far as successfully being able to dial one of the DDI numbers and getting the call to be delivered to an X-Lite SIP UA I have registered with Freeswitch. Unfortunately, I have two issues: 1) I have configured a SIP user account (Greg) but when I try to register X-Lite, it consistently returns a Registration error: 408 - Request Timeout. However, if I leave X-Lite alone, most of the time it will eventually (after quite some time) re-register with the server of it's own accord, allowing me to try a test call from the PSTN. 2) Whilst the SIP UA can then hear the call fine, there is a *lot* of noise on the PSTN end of the call - so much so that it's basically unusable, even though I can hear some things from the SIP end. Some info on the kernel/CPU included belom and some possibly useful logs / config files available here: openzap.conf: http://pastebin.freeswitch.org/13019 openzap.conf.xml: http://pastebin.freeswitch.org/13020 Greg.xml: http://pastebin.freeswitch.org/13021 61522x.xml: http://pastebin.freeswitch.org/13022 freeswitch.log: http://pastebin.freeswitch.org/13023 My question is where to start looking to diagnose these problems? I can provide whatever logs may be appropriate though I can't see any major problems in the freeswitch.log currently. With regards to the line noise issue, I read in the OpenZAP FAQ that unlike Asterisk, Freeswitch does not include a software echo canceller. Could that be related to the line noise? Should I attempt to install OSLEC to improve the quality? Thanks for any help - I'm fairly new to the world of telephony tecnologies like ISDN, Sam. # uname -a Linux bob 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386 GNU/Linux # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping : 6 cpu MHz : 864.011 cache size : 256 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 mmx fxsr sse up bogomips : 1728.02

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kamailio and freeswitch integration
by Daniel-Constantin Mierla 21 May '10

21 May '10

Hello, I put together a tutorial about using kamailio (openser) and freeswtich together: the proxy takes care of authentication and registration, freeswitch of media services, here is a link: http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms Is it ok to upload it to FS wiki so others can add to it? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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