- FreeSWITCH-users - FreeSWITCH Mailing Lists

freeswitch binaries w/o IPv6 anywhere (for w2k)?
by Pekka Kurki 05 Dec '09

05 Dec '09

thanks, --pekka--

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Setting up a FreeSwitch system on a pfSense firewall???
by mailinglist 05 Dec '09

05 Dec '09

Has anybody done this? I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there's any help to be found on FreeSwitch? Anybody that can give pointers to a good step-by-step instruction? I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider. Of course transferring calls between the two, as well as group calls would be a nice benefit.

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Eavesdrop error?
by Lars Zeb 05 Dec '09

05 Dec '09

I tried to use eavesdrop today and it did not work. The error message in the log is: [ERR] mod_dptools.c:334 Usage: [all | <uuid>] I simply dialed 881010, trying to eavesdrop on extension 1010. Is this incorrect? http://pastebin.freeswitch.org/11363 Thanks Lars

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Bridging to a non SIP based system
by Phillip Jones 05 Dec '09

05 Dec '09

Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port & CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port & CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil

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errors installing wanpipe drivers
by Neil Patel 05 Dec '09

05 Dec '09

Hi All, I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: > cd wanpipe-3.5.6.5/ > make openzap ... make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make -C api/libstelephony clean make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make[1]: *** No rule to make target `clean'. Stop. make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make: *** [all_lib] Error 2 The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. Thanks, Neil

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Playing an rtp stream
by Phillip Jones 04 Dec '09

04 Dec '09

Hi there, It it possible do something like: <extension name="rtp"> <condition field="destination_number" expression="^2127776252$"> <action application="answer"/> <action application="playback" data="rtp://192.563.41.246:27378"/> </condition> </extension> Basically I have need to connect to incoming calls listen to an existing rtp stream - I know the IP and port. Any hints on achieving this would be much appreciated. Thanks Phil

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Voicmail - message only
by Peter P GMX 04 Dec '09

04 Dec '09

Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter

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odbc FLAG_MULTI_STATMENTS
by Frank ＠ Impact 04 Dec '09

04 Dec '09

"GREAT SCOTT!!! Cannot execute batched statements! If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS" I realize a bit off of list topic. But I do have mysql 3.51.18 and higher but for the life of me , I cannot seem to get the DSN config setup so that the odbc connector seems to tell FS that it can do multi statements. Anyone have any insight on how and where to set this flag?

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Option to hang-up both legs in a bridge
by Nik Middleton 04 Dec '09

04 Dec '09

Hi, Is there an option to hang-up both call legs in a bridge when one leg hangs up? In my lua script I only ever see the hang-up for the call I'm in, not for the bridged b leg. That said, I can see both a hang-up and un bridge event being fired for the B leg. However my issue is that the A leg is still up, and if I've called 2 Pots numbers, the phone network will maintain the bridge. Is my only option to subscribe to the unbridge event and fire a hang-up event using the 'other leg' UID? Regards,

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FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
by Jerry Richards 04 Dec '09

04 Dec '09

I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing <enable-3pcc> to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry

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