October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

progressinband=yes in freeswitch?
by Peter P GMX 29 Oct '10

29 Oct '10

Hello, we have some delays with media setup with Aastra SIP over DECT phones. Aastra recommends using "progressinband=yes" for Asterisk so that the media to the phone is already setup during ringing phase. Are there any parameter in Freeswitch to support that? If yes: As we have registered more than one phone (Aastra SIP/DECT + Snom) to one extension, do we expect any problems in that case when media is already setup to both phones during ringing time? Best regards Peter

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microsecond timestamps
by François Delawarde 29 Oct '10

29 Oct '10

I've just seen that there are some channel variables (Caller-Channel-Created-Time, ...) having timestamps in micro-seconds. It is quite cool to see such resolution, but I was just wondering if there was a practical use for it. Does anyone actually uses such fine granularity in their CDRs or elsewhere? Do carrier environments have such requirements (and why?)? François.

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transfer related question
by Meftah Tayeb 29 Oct '10

29 Oct '10

hi, if i do blind or atended transfer i like my phone to pass through a special dialplan extension befaure thay refer to the other phone any work arround this? because i wanna falback to my phone if the transfer fail using *1 in dp could do it, but using the phone transfer button no idea thank you -- Meftah Tayeb inum: +883510001288000 Phone: +13602276297 Fax: +12538020313

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FS incoming calls Queue ACD implement
by Johnson 29 Oct '10

29 Oct '10

Hi FS Group, I am a new vanilla. Just want to realize Queue ACD function in Freeswitch. Is any one know how to configure or create the script to process incoming calls put into defined Queues without using mod_FIFO. Any additional prompting is very appreciated. Johnson -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-i… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Same caller id comes in freeswitch
by abubacker 29 Oct '10

29 Oct '10

Dear all , I faced an issue which comes rarely , when the freeswitch process the calls in a queue it would give the same customer caller id to the agents $ fifo list <bridge fifo_name="test_fifo" bridge_start="2010-10-29 11:03:51" bridge_start_epoch="1288330431"> <caller uuid="161fc7e8-e31e-11df-9a13-f1c4ac37d22d" caller_id_name="" caller_id_number="1"></caller> <consumer> <uuid>1b4c7090-e31e-11df-9a28-f1c4ac37d22d</uuid> <outgoing_uuid>c21108c2f348bea840a8bab0dbc3458a</outgoing_uuid> </consumer> </bridge> <bridge fifo_name="test_fifo" bridge_start="2010-10-29 11:03:58" bridge_start_epoch="1288330438"> <caller uuid="164f03b4-e31e-11df-9a14-f1c4ac37d22d" caller_id_name="" caller_id_number="2"></caller> <consumer> <uuid>1c7e4614-e31e-11df-9a2c-f1c4ac37d22d</uuid> <outgoing_uuid>906dab93f19c508ba53ef7628ee7ce3b</outgoing_uuid> </consumer> </bridge> above output says that the caller_id_number 1 and 2 are connected with the agent 1000 and 1001 but in the twinkle it says both 1 and 1 are connected with the agent 1000 and agent 1001 why it is showing bad caller id Note: Don't know whether attachment is allowed ( sorry if it has been prohibited ) -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer: http://www.bksystems.co.in/email-policy

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Memory problems
by Babstar 28 Oct '10

28 Oct '10

I have been using Freeswitch (git updated regularly) for about 9 months on a very low load ubuntu server. I have noticed that the after a period of time (sometimes only a few days) I am unable to make calls from attached SIP phones. Logging into the server I find freeswitch is taking upto 70% of the memory (measured by top), and I can hear the disk being thrashed (swapping, I assume). Restarting Freeswitch fixes all the problems & it then uses only 5% of them memory. How can I collect more debuging information to help fix the problem?

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FS Conference Call Recordings
by Michael Collins 28 Oct '10

28 Oct '10

Hello all! I just wanted to let everyone know that I have started putting the recordings for the conference calls on the main conference page: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls I've got them in mp3, ogg, and wave formats. Enjoy! -MC

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Running FS on Segate dockstar
by Jian Ren 28 Oct '10

28 Oct '10

Hi, Did anyone compile FS successfully on dockstar? What about skypopen? Thanks! Jian

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Re: [Freeswitch-users] Freeswitch and Digium
by Michael Collins 28 Oct '10

28 Oct '10

Grant, I believe you might be looking in the wrong place for information. I would start here: http://wiki.freeswitch.org/wiki/FreeTDM Note that if you are using a Digium-compatible card then you will need to install the DAHDI drivers first and make sure that part is working. (If you are able to run dahdi_tool and see your ports then you've got the DAHDI stuff done and can move on to the FreeTDM setup.) -MC On Thu, Oct 28, 2010 at 6:03 AM, Grant Gray <grant(a)mrit.co.za> wrote: > > > Hi All, > > > > I am trying to get a digium TDM410 card to work. I have looked at the > following tutorials: > > > > http://wiki.sangoma.com/wanpipe-linux-drivers > > http://wiki.sangoma.com/wanpipe-api-freetdm-linux > > > > and a few others. > > > > The problem I am having is that this command does nothing: > > > > *export FREETDM_MOD_DIR=/usr/local/freetdm* > > * * > > it does not create the directory as required. Is this normal on Centos or > is there another command? > > > > I also tried this command of one of the forum posts: > > > > make dahdi DAHDI_DIR=/usr/src/dahdi ( > http://wiki.sangoma.com/Topic-1273333936389) > > > > I poked around further and found that the export –p command shows that the > command is there: > > > > declare -x FREETDM_MOD_DIR="/usr/local/freetdm" > > > > So, I am wondering why its not working. I also cannot find a command like > /sbin/export so i am lost. > > > > what I am really trying to achieve is the installation of the tdm410 card > on freeswitch. Any tutorials would be really great. > > > > Regards, > > > > Gigg > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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issue with freetdm
by ovvenkat 28 Oct '10

28 Oct '10

Hello, I am using sangoma pri card A101 with freeSwitch. Today, I moved from openzap to freetdm. I have followed * http://wiki.sangoma.com/wanpipe-freeswitch-ftdm-installation* page to install freeSwitch and freedtm. After installing freeSwitch and Sangoma, I am getting following error in * fs_cli 2010-10-27 17:57:03.228457 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883 sng_isdn->s1:L1 Rx Error 2010-10-27 17:57:03.228457 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871 sng_isdn->s1: Resetting L1 link * What is these mean? I have followed wiki page and installed all the components. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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