August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Syntax of ${domain}
by Henk Oegema 18 Aug '08

18 Aug '08

What is the difference between something@$${domain} and something${domain} ? Both seem to work . Rgds Henk

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Can self control switch_channel_set_name() ?
by Lee JJ 18 Aug '08

18 Aug '08

Hello : My client is 1526@internal (210.243.126.69 <1526(a)internal%20(210.243.126.69>) call 9992@inter2 (210.243.126.72) , but the channel name turn into calle domain ( sofia/ "inter2"/ ?? ) This should not be right . How to avoid it ? or Can self control switch_channel_set_name() ? 2008-08-18 11:53:08 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/inter2/1526(a)210.243.126.69[32f88fbd-e1ae-4c7d-9178-6152b3c9b85f] 2008-08-18 11:53:08 [NOTICE] mod_dptools.c:601 answer_function() Channel [sofia/inter2/1526(a)210.243.126.69] has been answered thanks !

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conference.freeswitch.org via ipv6 (Testing)
by Brian West 18 Aug '08

18 Aug '08

If you have native ipv6 or a tunnel to ipv6 connectivity you can now call sip:888@[2001:470:7:ea::2]:5060 to join the freeswitch conference. I have also added to svn internal-ipv6.noload you can just cp it to internal-ipv6.xml and restart FreeSWITCH it will launch an ipv6 profile. bridge data="sofial/internal-ipv6/888@[2001:470:7:ea::2]:5060" http://www.tunnelbroker.net If you need/want ipv6 tunnels. I'll be working on ipv6 howto's over the next week. Thanks, Brian West sip:brian@freeswitch.org

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Problem using originate
by B Karthik 17 Aug '08

17 Aug '08

Hi, With bridge app, Freeswitch is able to place calls to multiple extensions simultaneously. But with originate, is it possible? If Yes, can anyone give me an example. The timeout option with originate also does not seem to work correctly. Timeout is not working (rings like a normal call and does not terminate within the given timeout) if i do something like this - newSession.originate(session, "user/1001(a)192.168.1.1",5); However it works if another extension is added with a comma - newSession.originate(session, "user/1001@192.168.1.1,user/9999@192.168.1.1 ",5); In the second ex, It does not call 9999 but the call disconnects exactly after 5 seconds. I also tried the same with event_socket using api commands directly. The results are same. OS: Windows XP SP2 Compiler: Visual Studio 2008 Thanks. B Karthik

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Mod_perl build error
by Sheeju Alex 17 Aug '08

17 Aug '08

Dear All, I get undermentioned error when I tried to build mod_perl, could anyone please let me know the fix. ------------------- ERROR ------------------------- making all mod_native_file making all mod_perl Creating mod_perl.so... /usr/bin/ld: cannot find -lgdbm collect2: ld returned 1 exit status gcc -w -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -pipe -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm -I/usr/lib64/perl5/5.8.8/x86_64-linux-thread-multi/CORE -DEMBED_PERL -I/usr/src/freeswitch-1.0.1/src/include -I/usr/src/freeswitch-1.0.1/libs/libteletone/src -fPIC -Werror -g -ggdb -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -o .libs/mod_perl.so -shared -Wl,-x mod_perl.o -Wl,-E -Wl,-rpath -Wl,/usr/lib64/perl5/5.8.8/x86_64-linux-thread-multi/CORE freeswitch_perl.o mod_perl_wrap.o perlxsi.o -L/usr/local/lib /usr/src/freeswitch-1.0.1/.libs/libfreeswitch.so /usr/src/freeswitch-1.0.1/libs/apr/.libs/libapr-1.a -lrt -L/usr/src/freeswitch-1.0.1/libs/srtp -lncurses /usr/lib64/perl5/5.8.8/x86_64-linux-thread-multi/auto/DynaLoader/DynaLoader.a -L/usr/lib64/perl5/5.8.8/x86_64-linux-thread-multi/CORE -lperl -lresolv -lnsl -lgdbm -ldb -ldl -lm -lcrypt -lutil -lpthread -lc -Wl,--rpath -Wl,/usr/local/freeswitch/lib make[5]: *** [mod_perl.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_perl-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 --------------------------------------------------------------- Also I have a question with mod_perl, As I am Asterisk user I am figuring out the way to call perl script from dialplan. exten => 88,1,Answer() exten => 88,2,AGI(check.pl) exten => 88,3,Goto(enter,s,4) I guess after installing mod_perl I would be able to do above function in freeswitch as well, is it something like this? <extension name="perl"> <condition field="destination_number" expression="^88$"> <action application="answer"/> <action application="perl" data="check.pl"/> </condition> </extension> looking forward for your reply! Thanks, Sheeju

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Caller id number in dialplan
by Ivan C Myrvold 17 Aug '08

17 Aug '08

I am testing out conditions in my dialplan, but am unable to get a match on caller_id_number, and I wonder why. I have to numbers registered from my PAP2, with User ID 100, and 200. If I have this in my dialplan, I thought I should get a match if I call any number from any of these two users: <extension name="Outbound_ip24"> <condition field="destination_number" expression="^(.+)$"> <condition field="caller_id_number" expression="^(100)$"> <action application="bridge" data="sofia/$${domain}/ 140%imyrvold.dyndns.org"/> </condition> <condition field="caller_id_number" expression="^(140)$"> <action application="bridge" data="sofia/$${domain}/ 100%imyrvold.dyndns.org"/> </condition> </condition> </extension> But FreeSWITCH can not find a route to either of 100 nor 140. If I remove the inner conditions, and dial any number, my 140 telephone will ring: <extension name="Outbound_ip24"> <condition field="destination_number" expression="^(.+)$"> <action application="bridge" data="sofia/$${domain}/ 140%imyrvold.dyndns.org"/> </condition> </extension> Why don't I get a match on caller_id_number? Ivan

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Re: [Freeswitch-users] help getting FXO inbound to bridge to
by Ian Henderson 17 Aug '08

17 Aug '08

Michael, Thanks very much for remembering about my request. The card was a recent eBay purchase so I was keen to confirm it worked sooner rather than later. Consequently, I built and installed Asterisk, keeping the Zaptel install unchanged. I managed to get that working on the inbound FXO with a small "fix", but have yet to try the workaround in FS. I'm hoping that the fix will translate. Basically, for testing I am using a Netgear TA612V as the FXS ports that I'm connecting to. Asterisk was also initially failing, but by debugging through the SIP I noticed that the Caller Id was coming in with 2 lots of double quotes, i.e. ""tav612v line 1"". It would appear that both the Cisco 7940 and another phone, a Welltech LP201, were rejecting the SIP request due to the double double quotes, with, from memory, a 400 error. I probably didn't have enough debugging configured in FS to see this issue. I tried overriding the callerid in Asterisk, and the SIP request was successful and the call connected. As I said, I haven't had the chance to reconfigure back to FS and apply the same fix. Happy to update the list when I do and confirm, or seek further guidance. Thanks again for getting back to me, Ian. > Ian, > > My apologies for not following up sooner. Where you able to get past this > issue? I just got my hands on a NxtVox TDM400 compatible card so I will > hopefully be playing with it soon. I'm sure that what you are trying to do > is not at all unusual and should be easily accomplished with the correct > setup. I will work to get that setup properly documented. > -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com

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Documentation suggestion
by Felix Rabinovich 17 Aug '08

17 Aug '08

I signed up to this list when the discussion was already ongoing, so I can't reply-to the existing messages. But this is in response to Michael Collins' comments about documentation plans. I am very technical (developer / system administrator), but I am fairly new to telephony, and I am trying to set up a PBX at home, and then set up similar PBX at my wife's small business. What I am missing most, is "use case" documentation beyond Installation and getting started. Installation and Getting started are quite good. There are some updates that contradict previous information, but I got the system up and running in a few hours. Two inbound providers, two outbound. Now I have some other use cases that I want to implement. For example, * how to set-up IVR * how to check voicemails with passcode or the system recognizes caller-id and doesn't require passcode * for certain incoming caller-ids ("important client") forward the call to predefined number ("cell phone") * how to load custom music-on-hold Obviously, this is just an example. I am sure other people have different use cases. I don't have PSTN line at home and only one ATA/extension that is connected to the server. I tried softphone / 2nd extension just to make sure that everything works, but it is not a priority for me. For others, transferring a call from one extension to another might be the most important thing. In my wife's office there will be a fax line (probably PSTN that will serve as fallback / emergency line) I know that all these scenarios are very trivial to set up. But I can't find any guides that would show me how to do this. Thanks Felix

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iptel.org
by Michael S Collins 16 Aug '08

16 Aug '08

Just curious: does anyone have experience with iptel.org? Has anyone connected FS to their service? Lastly, they have a lot of basic info about voip and sip on their site. It looks pretty good but I wanted a second opinion before I add any links from our wiki to their site. Thanks, MC Sent from my iPhone

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Video conferencing in freeswitch
by David A. Horner 16 Aug '08

16 Aug '08

Hey everyone, So I've been thinking about how to get video conferencing going with freeswitch. When I first started, I was thinking about developing a video capture module for freeswitch....though then I'd have to write custom RTP packetizer/depacketizer... VLC already provides network RTP streams of video data....like webcams. I hear that the freeswitch conference currently supports video sip clients now, however, I don't yet own one to test with. Could someone leverage the current features in freeswitch to use a VLC network RTP stream? I guess the thing that is missing is something to broadcast the video stream urls to each of the conference participants and start the vlc streaming/viewing automatically. Anyways, just thoughts.... --Dave http://dave.thehorners.com/

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