November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Grandstream GXP 2000
by Jonas Gauffin 18 Dec '11

18 Dec '11

Hello I have problems with TCP and GXP2000 phones. They work fine using UDP, but no calls arrive if I switch to TCP. I've upgraded to the latest GS firmware today to see it that helped, but it didn't. Have anyone else had problems with TCP? Regards, Jonas

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FreeTDM does not work
by Valery Kalinin 16 Dec '11

16 Dec '11

I upgraded freeswitch from git. Nothing changed in the configuration! Suddenly stopped working freetdm. Why? my configs: freetdm.conf [span zt pri] name => pri trunk_type => E1 group => e1group b-channel => 1-15 d-channel => 16 b-channel => 17-31 wanpipe.conf [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 freetdm.conf.xml <configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="7"/> </settings> <libpri_spans> <span id="1" name="pri"> <param name="node" value="network"/> <param name="switch" value="euroisdn"/> <param name="l1" value="alaw"/> <param name="dp" value="international"/> <param name="debug" value="all"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </libpri_spans> </configuration> log: 2011-11-27 14:35:12.284837 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 3 2011-11-27 14:35:12.284847 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 4 2011-11-27 14:35:12.284898 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 1 2011-11-27 14:35:12.284906 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 2 2011-11-27 14:35:12.285290 [WARNING] ftmod_zt.c:333 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 1 fd 50 (Invalid argument)] 2011-11-27 14:35:12.285523 [WARNING] ftmod_zt.c:333 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 3 fd 50 (Invalid argument)] 2011-11-27 14:35:12.285634 [ERR] mod_freetdm.c:2465 Error finding FreeTDM span id:1 name:pri Card Digium TE121 Please HELP!

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Question Answertime variable?
by Thomas Hoellriegel 16 Dec '11

16 Dec '11

Hi all, Is there a possibility to determine the caller time for a channel at the end of a call? For example: in asterisk exists a variable: ${ANSWEREDTIME} I like to store the Answertime in a Database for example: Dailly minutes to call a Cellphone. Can your help plese? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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Outbound calling with recorded message.
by Dave 08 Dec '11

08 Dec '11

Hi all, I have been using freeSWITCH for a while for inbound calls in which a person registers for a seminar via IVR. I simply use the DID, CID number and name to identify the person and put them in the Database. That part of the application is working wonderfully. The issue I'm presented with now is that we need automate making calls to a few of these registrants, after each event, whos caller_id_name comes in as "Unknown" or "Wireless Caller", Play a recording to let them know we need the name, and that they can record the information right then, or call the office. I am limited to using a Windows 7 machine. What would be the best tool to use to automate these calls with the use of an IVR? Dave Goodwin

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Need help
by Sam 08 Dec '11

08 Dec '11

Hi, I need some help with an idea I am trying to implement with Freeswitch. I am trying to work with a partner company which will forward a sip call to my freeswitch server and my freeswitch server is suppose to playback annoucements to the caller and then once they are finished listening they should be forwarded back to the partner company so that the call can continue without my freeswitch being in the call path. So does any one have any suggestions/ideas on the best way to implement this?

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In Bypass Mode Freeswitch Changes SDP?
by Jerry Richards 05 Dec '11

05 Dec '11

Hello, I noticed an interop issue when using a Bria softphone and inbound-bypass-media is true in the sip_profile. When answering an inbound call, the Bria softphone's SDP Answer includes the list of codecs it supports (audio and video). I noticed that, even in bypass media mode, Freeswitch filters out all of the codecs in the list after the first one. I think this is not right. According to RFC 3264 (Section 7 Offerer Processing of the Answer), it is valid to provide a list of codecs in an SDP Answer and the endpoint SHOULD use the first one (but it might not). Why is Freeswitch altering the SDP in the 200 OK? I think it should send the SDP unmodified. Thanks, Jerry

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get cepstral volume lower
by Madovsky 05 Dec '11

05 Dec '11

I'm trying the SSML example from mod_cepstral wiki <action application="speak" data="cepstral|david|<prosody volume=13>Thank you for holding</prosody>"/> but the voice doesn't say the sentence but says something like "slash blablabla...". Any idea why I can't embed SSML tag in the cepstral sentence ? Thanks

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mod_rtmp: "Read error" when attempting calls from FS to RTMP client.
by peely 03 Dec '11

03 Dec '11

Hi, I have a strange issue when I attempt to bridge a call from FS out to an RTMP client. As soon as the call is offered up to the client, the call is disconnected immediately i.e at the "ringing" stage. The strange thing is that as soon as I issue an "fsctl loglevel 9" then everything is fine. I don't know whether this is because of any additional load, as nothing above "ERR" is printed in each case. When the call is cleared immediately then I am getting "[ERR] rtmp.c:713 Read error" in the logs. mod_rtmp is compiled from today's git. Is there a possibility something is trying to happen too quickly in the handshake before data has been received from the client? Outbound calls from client to FS work flawlessly and incoming calls too, so long as I seem to have debug logging enabled. I'm not sure if it's relevant, but I'm bridging to the UUID of the session AKA "<action application="bridge" data="rtmp/cd421c76-1614-11e1-be92-397d873fd9b8"/>" Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-atte… Sent from the freeswitch-users mailing list archive at Nabble.com.

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google voice connection going offline (mod_dingaling)
by Federico Beffa 02 Dec '11

02 Dec '11

Hi All, I've setup mod_dingaling to connect to my google voice account according to the instructions on the wiki http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_t… with two modifications: <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> I use the same IP settings for SIP (sofia) with no problem. The setup works fine for the first hour or two, but then I can see from another google account that the connection established by freeswitch goes offline. If I try anyway to make an outbound call, then the other end does not ring (even if I wait for a long time). However, after some minutes the freeswitch account returns online and works again for a couple of hours. I'm behind a nat and firewall. Is there any port which I have to open to keep freeswitch online? As far as I understand only the google XMMP server needs to listen to port 5222 and port 5269. Or, is there any kind of "keep alive" timer available? The google talk client running in the same environment works flawlessly. Thanks for any advise. Fede

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is 100rel still a bad idea?
by Dave Horton 01 Dec '11

01 Dec '11

I saw a few threads from a couple years back indicating the enabling reliable provisional responses could cause a FS crash. In fact, I am working with a customer who was told some time ago (by "freeswitch support", not sure exactly who they are referring to) to turn off 100rel (this while they were troubleshooting an unrelated issue). Just looking for an update on this, because I prefer to run my SIP gear with 100rel supported --- is there still a known bug (or even, suspected issue) with 100rel support on FS?

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