- FreeSWITCH-users - FreeSWITCH Mailing Lists

Playing a file again and again
by Faisal Maqsoodi 06 Dec '08

06 Dec '08

Hi, Is there any built-in function, like playback, which plays a file again and again unless interrupted. I want to use a simple function not FIFO. Faisal

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Windows is slow?
by Per Møller 06 Dec '08

06 Dec '08

I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1 on a Windows Vista machine using the precompiled .msi - actually the same machine). Using the default configuration files, and using 2 Snom 360 phones I dialed from extension 1000 to extension 1001. On the Mac, 1001 starts ringing instantly, but under Windows it takes 1-2 seconds before it starts ringing. It seems to be in the dialplan the time is spent. From the time I see this line on the console: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in context default Until the next thing happens it always takes at least 1 full second, but on the Mac it happens instantly. Why is the Windows build this much slower? Is it a known problem? I get the feeling that the majority of the FS community is Unix based, which is fine by me, but I would really like to know just how well supported and stable the Win32 build is and if this is currently a viable way to go, or if I should stick to Linux/BSD/Mac for production use? // Per

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How to setup TLS between two Freeswitch servers
by shehzad p 06 Dec '08

06 Dec '08

I am wondering how to setup two freeswitch servers to route call with TLS configured between them. As shown in wiki http://wiki.freeswitch.org/wiki/SIP_TLS, I created two certificates on one freeswitch, and changed SIP profile by enabling tls in it, then Starting freeswitch it just opens port 5061 (for TLS ), But when i route the call from that FS server, it uses the its general ports (5060 and 5080) for call. Where i am missing something?, A doubt is about where to place which certificate. -- View this message in context: http://www.nabble.com/How-to-setup-TLS-between-two-Freeswitch-servers-tp208… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Provider: Junction Networks
by mehdix 06 Dec '08

06 Dec '08

I've got a problem with configuring a SIP trunk from Junction Networks with FS: it only works for a few minutes then the line is dropped. I tried Unlimitel with no problem. Any Ideas? Thanks -- View this message in context: http://www.nabble.com/Provider%3A-Junction-Networks-tp20859688p20859688.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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How to force FS using the hosts entry in outgoing gateway calls
by Jon Bruel 06 Dec '08

06 Dec '08

For the configuration of a gateway I need to use a specific proxy domain name before the server (Covergence SBC with a BroadWorks Application Server behind) accepts calls. The twist is that the right proxy name points the wrong IP-address (the voicemail server for the account). I have tried to overrule this by adding a host entry (Linux). When I ping to the domain name I get the right address (the one from the host table), but the FS uses the address from the DNS lookup, not the address from the host table. What can I do to force the FS using the entry from the host table? Thanks /Jon.

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Re: [Freeswitch-users] How to force FS using the hosts entry in outgoing gateway calls
by Jon Bruel 05 Dec '08

05 Dec '08

Thanks Anthony. Using the parameters: <param name="proxy" value="domain name"/> <param name="register-proxy" value="ip address"/> Returns error 900, and a 'ngrep port port-number' indicates that its doesn't try to register at all. I have now let the server look at a local DNS where I have added a "wrong" A-record. That solves the issue, but your solution would be cleaner. The version is: trunk 10220. /Jon

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Re: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma A104
by Evgeniy Zolotov 05 Dec '08

05 Dec '08

Greetings! Question about possibility of the use FreeSWITCH for work with T1/E1 streams under Sun Solaris 10 a bit clears up (Solaris 11 is in condition of alpha-version and not suitable for the industrial use). But answers carry more negative sense. Start of T1/E1 under Sun Solaris has 2 stages: a) start of wanpipe's interface; b) make FreeSWITCH for Sun Solaris. a) For Linux Sangoma recommends (http://wiki.sangoma.com/wanpipe-freeswitch-install) installation of new interface wanpipe-3.3.14.tgz, which has beta-status and placed at ftp://ftp.sangoma.com/linux/current_wanpipe. This installation and tests were successful. In this release (2007 y.) by Sangoma was added TDM API - native interface for FreeSwitch (YATE) — which is absent in all previous releases of WANPIPE. In the Linux release 3.3.14 this variant was named as «TDM API», in older releases - «TDM Voice». In this case FreeSWITCH works very good with E1/T1 cards A-101/102/104/108 without installation in zaptel system, and spans in configuration files of FreeSWITCH declared as [wanpipe#]. But for Sun Solaris we found only drverftp://ftp.sangoma.com/Solaris/Beta/SVwanpipe-i386-5.10.pkgbeta, which dated 2007: > NAME=sangoma.com Wanpipe Driver > VERSION=1.1.0,REV=2007-07-16, Packet Svwanpipe-i386-5.10.pkg uses Svzaptel-i386-5.10.pkg, which wasn't developed by Sangoma, but by little free communityhttp://www.solarisvoip.com/, source codes of this packet is herehttps://svn.sunlabs.com/svn/solaris-asterisk/zaptel-solaris/trunk/ The analysis of source codes shows that this project develops very slowly (there are no updates for about year), has very limited functionality and supports ( unlike original zaptel ) very limited list of cards ( only one ;) - Digium Wildcard TE110P T1/PRI). Svwanpipe-i386-5.10.pkg supports only 64-bit Sun Solaris (on CD, which we get with Sangoma's cards, presents 32-bit driver and PDF document about installation under Sun Solaris — but it dated 2001 — 2002 yy). At first we have checked up installation with TDM Voice + zaptel (like for Asterisk) under Linux. We configured PRI spans as [zt] ... - such installation works good and we could do calls : > originate openzap/1/A/20000 &sleep(3) * evidently that call retranslates from span 1 to span 2, connected with cross-cable, and goes to extension 20000. b) About making FreeSwitch: Under Sun Solaris 10 with GCC makes FreeSWITCH core and most modules, except openzapand some others, because Sun Solaris 10 has GCC 3.4, but ./configure for openzap requires compatibility with ANSII 99. We have checked 3 different methods of makinf FreeSWITCH : 1. set up GCC 4.0.2 from CSW-repositaries (and all reguired for GCC *.pkg); 2. set up SunStudio 12 and do make cc/CC; 3.cross-compile FreeSwitch for Sun Solaris under Linux with CC-options. In testing we used assembly SunStudio as 32-bit application (64-bit comes to the end with mistakes of assembly of some libraries, it is possible to correct for it easily, but we did not begin to specify it). For testing cards A-104 we have repeated the same installation and configuring as under Linux (wanpipe TDM Voice + zaptel) on 4 different 64-bit servers under Sun Solaris 10: * on 3 servers (2 of them manufactured by SUN) executed successfully : # wanrouter start 4 wanrouter# interfaces was created; leds on spans, connected with cross-cable, becomes GREEN, i.e. synchronization T1/E1 presents (no alarms). On these servers FreeSWITCH correctly makes: > load mod_openzap * but when we make : > originate openzap/1/A/20000 &sleep(3) - for 2 connected with crosss-cable spans (1 & 2) FreeSWITCH transmits PRI message (chan 1/31), but chan 2/31 doesn't receive this message (unlike under Linux) and call breaks after timeout. On 4-th server wanpipe doesn't even starts by mistake some IOCTL (i.e. at a command for device). Three «working» servers is on AMD Opteron Dual Core 2214(F), fourth is on ntel Xeon 3210. So, we supposes Svwanpipe-i386-5.10.pkg or Svzaptel-i386-5.10.pkg from Sangoma checked up a little and on some one processor and doesn't heave up even on beta, as they are declared, at the best on alpha What's the reason of error??? * wanpipe doesn't work under Solaris? * wrong working signalling with zaptel? (but same configuration works good under Linux) * wrong working of FreeSWITCH, which was built correctly, but their work was violated? * * Thanks, Evgeniy.

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Re: [Freeswitch-users] DTMF from cell phones
by Jan Kubr 05 Dec '08

05 Dec '08

> > no solution. I have a similar problem, when calling Freeswitch from my > > cell phone (via a SIP provider), sometimes DTMF is not recognized >> The important thing to note is that when using >> a SIP softphone (X-Lite) I have never had this problem, DTMF is > So i guess that using latest version with a few changes in your config > should work unless there's any other issue related to your sip provider > ( PSTN / Media Gateway ), on this case you can get some captures of > sip/rtp traffic to check SDP and rtp Marks. I tried trunk and the values for the variables (all except rtp-timer-name=none are already default in trunk), but only two things are different: 1. When I press a key, the read app seem to always terminate, but not always the dtmf is captured in a variable. 2. The read app seems to ignore the variable name parameter: calling it with "1 1 104.wav choice_181152 10000 #" doesn't put the digit in variable_choice_181152, but to dmtf_digit. Why is that? > If it is coming from the sip provider as rfc 2833 dtmf, they are > probably doing inband detection and failing at it. If you look at an > rtp dump you can confirm this. If this is the case, there is nothing > you can do on the FreeSWITCH side and the provider will have to fix it. But the call goes through the same SIP provider even when using the soft phone and there it works fine. The difference might be that then it is SIP to SIP within the same provider.. How do I do the RTP dump? Also I should have mentioned that DTMF is not captured only DURING the file is being played. It is always captured correctly when I wait until the playback is finished. Does this sound familiar? I thought this would be somet obvious misconfiguration on my side. Jan

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Event_Socket: How to determine the ogininater uuid before an outgoing call is set up?
by Peter P GMX 05 Dec '08

05 Dec '08

I am building an IVR application where an incoming call is initiating an outgoing call. When I pass a "variable_other_uuid" (the uuid of the incoming channel) at originate time, I am able to reference to the incomig call, once the outgoing call is set up. So the outgoing call can see the uuid of the incoming call who has originated the outgoing call. This is needed for bridging the 2 calls together. However I want to control also the call setup process (see, if the outgoing call is ringing etc.). At call setup time, when I parse the channel_originate ,channel_outgoing and channel_progress events, I cannot see any reference to the incoming call (variable_other_uuid is not set). I suspect that variables are only passed once the outgoing channel is set up. Has anybody an idea, how I may get the uuid of the originating uuid in the outgoing call at call setup? Best regards Peter

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key tone trigger event during call
by Frank ＠ Impact 05 Dec '08

05 Dec '08

Is there any dialplan instructions that could be added that would sit and listen during a call for a tone (a key press, say 2) and when FS hears that tone, then FS can broadcast another key tone (say 6) back to the channels? -Frank

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