July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with setting up a variable in <variables> section of a gateway
by Dmitry R 10 Jul '12

10 Jul '12

Hi guys, I'm trying to set a channel variable effective_caller_id_number in gateway's <variables> section and it seems not working. However, if I set it up with application set command in dialplan it works fine. What is the difference between these two ways? My gateway section looks like this: <include> <gateway name="CanUS"> <param name="username" value="64721"/> <param name="password" value="supersecretpasword"/> <param name="from-user" value="64721"/> <param name="from-domain" value="11.22.33.44"/> <param name="proxy" value="11.22.33.44"/> <param name="expire-seconds" value="800"/> <param name="register" value="true"/> <param name="retry-seconds" value="60"/> <param name="context" value="public"/> <param name="caller-id-in-from" value="true"/>  <variables> <variable name="effective_caller_id_name" value="Gateway"/> <variable name="effective_caller_id_number" value="64721"/> </variables> </gateway> </include> and working dialplan strings look like this: <action application="set" data="effective_caller_id_number=64721" /> <action application="bridge" data="sofia/gateway/CanUS/$1" /> Thank you, Dmitry

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Sending INVITE without sdp
by Srini K 10 Jul '12

10 Jul '12

Hi,FreeSWITCH can receive INVITE with no sdp by configuring <param name="enable-3pcc" value="true"/> in sip profile.Is there any way to send Invite out from FreeSWITCH without sdp? ThanksSrini

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Codec Negotiation
by Stephen Wilde 10 Jul '12

10 Jul '12

Is this message flow valid? FS ---------------- MyProvider ---> INVITE (w/SDP G729, G711a, G711u) <--- 100 Trying <--- 183 Session Progress (w/SDP G729, G711a, G711u) <--- 200 OK (w/SDP G729, G711a, G711u) What is strange to me is that my provider send back a 183 Session Progress with all codec list of Freeswitch INVITE (I expect only the choosen codec). It's this message flow correct? Stephen

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Attended transfer via ESL
by Limit 10 Jul '12

10 Jul '12

I'm trying to make an attended transfer through EventSocket, using sendmsg with att_xfer command, but having problems with it. ESL function: execute("att_xfer", args, b_uuid); It creates new calls as normal dialplan att_xfer, but problem occured when one of three sides hangs up. Sometimes it goes as planned, but more often all channels hangup automatically. att_xfer called from dialplan work as needed, looks like there is some problem with EventSocket call specifically . Maybe I must set some variables before to use it properly? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Attended-transfer-via-ESL-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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G_729 Virtual Server Audio Quality Problem
by Josh Foshee 09 Jul '12

09 Jul '12

Let me start off with the environment that I am using. I have a virtual server that is hosted on Linode.com. They are running Xen Server VPS. The Freeswitch version I am using is FreeSWITCH Version 1.2.0-rc2+git~20120629T194106Z~5f09b40381+unclean~20120630T202857Z I am using mod_com_g729 with 20+ license installed. I have two Sip Profiles setup (internal and external). The internal with authentication is on port 5060 and external with no authentication on port 5080. They have different codec values for each. The Sip provider that I use is Flowroute.com The Leg from Flowroute to Linode I will call Leg A. The Leg from Linode to phones I will call Leg B. The problem that I have is that I can setup the pathway with the codec PCMU. I can place a call and both parties sound find and no audio problems. I can change the Leg A to g729 and still have no audio problems. I then change Leg B to g729 and the remote party (the one on the PSTN network) will hear the site traffic as choppy. The local party will hear quality issues not as much. It is a constant problem on the remote party side though. I then change Leg B back to PCMU and the quality is great. I change it back to G729 and it goes back to bad audio quality. All testing was done between 1am to 2am but this was also reproducible during the daytime. I'm pretty sure it is not a bandwidth problem. I have noticed a problem in the console that when one call is connected and stable this keeps scrolling over and over. http://pastebin.freeswitch.org/19466 Any ideas as to the cause of the audio problems with G729?

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FreeSWITCH Weekly News and Notes
by Michael Collins 09 Jul '12

09 Jul '12

Greetings all! We are back to a full week after taking last Wednesday off. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_11?cc12-0709>we have Dave Kompel scheduled to talk to us on a subject that has been of much interest in recent weeks: firewall security on Windows deployments of FreeSWITCH. Many of you know that we have fail2ban<http://wiki.freeswitch.org/wiki/Fail2ban?cc12-0709>available for Linux/Unix based installs, but what about Windows <http://xkcd.com/528?cc12-0709>? There are indeed techniques for Windows systems to update the Windows firewall to prevent things like the friendly scanner <http://www.freeswitch.org/node/343?cc12-0709>. Recall: the friendly scanner is an abuse of a SIP testing tool called SIPVicious<http://blog.sipvicious.org?cc12-0709>. People use it for nefarious purposes, so having the ability to lock down the source IP address on the fly is a powerful technique to employ. We look forward to this discussion <http://xkcd.com/841/>. We are happy to report that we have a new GOLD sponsor at ClueCon: Flowroute<http://www.flowroute.com?cc12-0709>! The folks at Flowroute have graciously offered to make sure that our attendees are well taken care of. First, they are supplying the gift bags that all of our attendees will receive. In those bags will be lots of goodies from Flowroute and our other sponsors<http://www.cluecon.com/sponsors?cc12-0709>. A special item will be a $25 gift card for Flowroute. Additionally, Flowroute is sponsoring a pizza party on Monday night, August 6 at Gino's<http://www.ginoseast.com?cc12-0709>in downtown Chicago! We hope you all will be able to join us. Only a few more weeks until we're all together again! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Why FS recreates indexes when it starts
by Jimmy Godbout 09 Jul '12

09 Jul '12

Hi, I'd like to know why does FS tries to recreate its indexes when it restarts. When using a clustered solution, FS will then fail because the DB is still running and the indexes a still there. ____________________________________________________________ Send any screenshot to your friends in seconds... Works in all emails, instant messengers, blogs, forums and social networks. TRY IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if2 for FREE

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Different Voices for FS
by toddb＠toddbailey.net 09 Jul '12

09 Jul '12

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Call recording quality problems when both legs have different codecs
by Jayesh Nambiar 09 Jul '12

09 Jul '12

Hi, I am using mod_callcenter and have enabled recording_template in my queues for all my calls to be recorded. Now there is a problem in quality of recorded file when freeswitch answers the call in one codec and connects with the agent on a different codec. For eg: The incoming call comes in and get connected with G711u codec. The caller is listening to MoH while waiting in queue. The agents have G722, G711u both enabled on their phones and the sofia has outbound-codec-prefs as G722,PCMU. So when the agent answers, the call is connected with G722 codec. So in this case, the recorded file generated plays as if the call is getting fast-forwarded. The audio in the call is very fast. Whereas if I change the codec of Agent's phone to G711u only the recorded file quality is perfect. I don't think it is related to callcenter module anywhere but since I am using it in callcenter I have given this example. I think its more related to the record function. I am running latest GIT version and using mod_shout for recording calls in mp3 format. I had the same problems when recording in wav format also. Any help regarding this is really appreciated. I have attached two files; one which has PCMu on first leg and 722 on other and the second one has PCMu on both legs. Thanks in advance. --- Jayesh

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spandsp error reading frame
by Jean-Marc Hyppolite 09 Jul '12

09 Jul '12

Hello, I am trying to use mod_spandsp to detect call progress. I haven t been successful so far. I am getting the following error messages. ================================================ 2012-03-03 23:55:08.398205 [DEBUG] mod_spandsp_dsp.c:379 (sofia/outbound/.@...) Starting tone detection for '1' 2012-03-03 23:55:08.398205 [INFO] mod_spandsp_dsp.c:411 (sofia/outbound/.@...) initializing tone detector 2012-03-03 23:55:08.398205 [DEBUG] switch_core_media_bug.c:462 Attaching BUG to sofia/outbound/.@... 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:422 (sofia/outbound/.@...) error reading frame 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:447 (sofia/outbound/.@...) destroying tone detector ================================================ Any help would be appreciated. Thanks Jean-Marc. N.B. Sorry if some people received this e-mail twice. I am not sure this email went through the first time.

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