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change sounds file format of voicemail
by Madovsky 02 Jul '11

02 Jul '11

Where can I force voicemail to use sounds other than .wav ? Thanks

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Could not connect thru GSMOpen
by ankIT WALiA 01 Jul '11

01 Jul '11

Hi all, I have properly compiled the gsmopen module. Added the Dialplan <extension name="gsmopen"> <condition field="destination_number" expression="^1002$"> <action application="bridge" data="gsmopen/interface1/xxxxxxxxx"/> </condition> </extension> ---------------------------------------------------- And the gsm conf file <configuration name="gsmopen.conf" description="GSMopen Configuration"> - <global_settings> <param name="debug" value="8"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="hold-music" value="$${moh_uri}"/> <param name="destination" value="9999"/> </global_settings>  - <per_interface_settings> - <interface id="1" name="interface1"> <param name="hold-music" value="$${moh_uri}"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="destination" value="5000"/> <param name="alsacname" value="plughw:1"/> <param name="alsapname" value="plughw:1"/> <param name="controldevice_name" value="/dev/ttyACM1"/> </interface>  </per_interface_settings> </configuration> ------------------------------------------------------------------ I am using a Nokia handset in India --------------------------------------------------------------------- I am getting the below error on gsm_load. gsm reload 2011-06-30 19:03:52.716491 [WARNING] mod_gsmopen.cpp:1860 rev exported[(nil)|37 ][WARNINGA 1860 ][interface1][-1, 0, 0] STARTING interface_id=1 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2999 rev exported[(nil)|37 ][ERRORA 2999 ][interface1][-1, 0, 0] snd_pcm_open failed with error 'No such file or directory' on device 'plughw:1', if you are using a plughw:n device please change it to be a default:n device (so to allow it to be shared with other concurrent programs), or maybe you are using an ALSA voicemodem and slmodemd is running? 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2891 rev exported[(nil)|37 ][ERRORA 2891 ][interface1][-1, 0, 0] Failed opening ALSA capture device: plughw:1 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1931 rev exported[(nil)|37 ][ERRORA 1931 ][interface1][-1, 0, 0] alsa_init failed 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1932 rev exported[(nil)|37 ][ERRORA 1932 ][interface1][-1, 0, 0] STARTING interface_id=1 FAILED 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:3146 rev exported[(nil)|37 ][ERRORA 3146 ][interface1][-1, 0, 0] ALARM on interface interface1: ----------------------------------------------------------------------------------------------------------------- I could not understand what does the error message trying to say? I am connected to friend's mobile with datacable on com1 port. I jsut wanted to check SMS. Please help. Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

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Generating ringback if 180 arrives after 183
by Matteo 01 Jul '11

01 Jul '11

Hi all, I've the following scenario from my upstream provider : FS calls provider Provider sends easrly media with 183 announcing that the call is going to be trasferred Provider sends 180 without sdp In my dialplan I'm using ringback variable to let FS generate ringback to internal network, and if I receive 180 without 183 is working ok. What I'm trying to achieve is to let ringback work even if we've already received the 183 with sdp. I've tryed a lot of dialplan magic without luck. Maybe we need to add this support into FS itself ? What I see seems that we need to break the bridge or at least "inject" the tone stream into the internal leg of the call. To be fair, the exact sequence is: FS calls provider provider sends 183/sdp annoucing the call trasnfer provider sends update which stops the media (media=inactive) provider sends 180 ringing (we must generate here) maybe the correct way is to handle the update, put fs back into ringing state and proceed from there? Is feasible or we can get the same result with some dialplan magic? any hint is appreciated :) regards, matteo

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How to send Outbound Caller ID Name out Sangoma A101 PRI
by Robert Hadley 01 Jul '11

01 Jul '11

Hi FS Users, Is there a way to send the outbound caller ID name when using FreeTDM, wanpipe, and a Sangoma A101 PRI? I am setting the effective_caller_id_name and _number and the number goes out but not the name. I also tried setting the origination_caller_id_name on the bridge but that didn't work either. I receive the incoming caller_id_name in a separate Facility message. Thanks, Robert

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mod_limit
by Tomasz Kopacki 01 Jul '11

01 Jul '11

Hi, Im trying to set mod_limit but i dont understand what are realm and id parameters for. I have 2 users groups - Group A and Group B. Group A has two domains and Group B has three domains. I want to set limit 100cc on group A and 150cc on group B. I did this: <action application="limit" data="hash realm groupA 100 /> <action application="limit" data="hash realm groupB 150 /> Will it works ?

2 1

t38-passthru sofia param
by Spencer Thomason 01 Jul '11

01 Jul '11

Hello all, I have a few questions regarding the t38-passthru sofia param. One of the ways we are using FreeSWITCH is as a simple B2BUA in our SBCs. The topology looks like this: Freeswitch /\ \/ FS and Asterisk boxen -> OpenSIPS -> PSTN The Freeswitch instance is setup with inbound late negotiation and bypass media = true and the dialplan consists of a very simple bridge statement that sends a call back to the proxy. The problem is that when someone sends a fax and the GW send a t.38 reINVITE, Freeswitch is sending the original SDP (PCMU in this case) back to Leg A and the fax then fails using PCMU. Is this parameter needed when bypass media is already enabled? Thanks, Spencer

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(no subject)
by sharad garg 01 Jul '11

01 Jul '11

Hi to all, This is just let to know this forum that we have completed the work on Alarm Service. All this is done using our great Freeswitch only. The GUI also is designed for the same. You can have a view at the attached PPT for the details. Thanks to Freeswitch developers & community off course who helped us in this project directly or indirectly. Regards Sharad

2 1

inbound trunk from Linksys 3102?
by Bret Watson 01 Jul '11

01 Jul '11

Hi All, one of the few things that really worked on my * setup was the inbound trunk from the 3102... * config was trunk name 1-pstn disallow=all allow=ulaw canreinvite=no context=from-trunk dtmfmode=rfc2833 host=dynamic incominglimit=1 nat=never port=5061 qualify=yes secret=xxxxxxxx type=friend username=1-pstn How do I translate this into freeswitch? Thanks! Bret

1 0

Re: [Freeswitch-users] please help!!! how to set flag 'endconf' in bridging conference
by ran zhang 01 Jul '11

01 Jul '11

I attached the section in my dialplan that handles the bridging conference, when the first user (just say user 99) dials '20', he will invite user 10 to join conference, the conference will only be established if user 10 accepts the invitation, after this, other users can join the conference by dialing '20'. I want that if the first user (user 99 in this case) or the user been invited (user 10 in this case) leaves the conference, no matter how many people are still in the conference, it will close down the conference. so i'm trying to use the 'endconf' flag, but apparently it is not valid syntax for bridging conference as I get a config error while running. If I take out the '+flags{endconf} ', i wont get a config error while running, but then conference will only close down when there is 1 person left. I have also tried creating 2 difference conference profiles, one profile has the 'endconf' set in the 'member-flags', one profiles doesnt have 'endconf' set, so user99 and user20 joins the 20@profile1 conference, and other users joins the 20@profile2 conference, that doesnt seem to work neither. I have pasted conference.conf.xml file as well for ur review. <include> <context name="test"> <extension name="global" continue="true"> <condition> </condition> </extension> <extension name="conf"> <condition field="destination_number" expression="^(20)$"> </condition> <action application="answer"/> <action application="conference" data="bridge:20@default+flags{endconf}:user/10"/> <action application="conference" data="20@default"/> </condition> </extension> </context> </include> On 6/30/2011 6:16 PM, Michael Collins wrote: > Please put this information on pastebin and reply to the list so that > we can all discuss it. > -MC > > On Thu, Jun 30, 2011 at 1:51 PM, ran zhang <rzhang(a)gosilverplus.com > <mailto:rzhang@gosilverplus.com>> wrote: > > Mr Collins: > > I attached the section in my dialplan that handles the > bridging conference, when the first user (just say user 99) dials > '20', he will invite user 10 to join conference, the conference > will only be established if user 10 accepts the invitation, after > this, other users can join the conference by dialing '20'. > > I want that if the first user (user 99 in this case) or > the user been invited (user 10 in this case) leaves the > conference, no matter how many people are still in the conference, > it will close down the conference. so i'm trying to use the > 'endconf' flag, but apparently it is not valid syntax for bridging > conference as I get a config error while running. If I take out > the '+flags{endconf} ', i wont get a config error while running, > but then conference will only close down when there is 1 person left. > > I have also tried creating 2 difference conference > profiles, one profile has the 'endconf' set in the > 'member-flags', one profiles doesnt have 'endconf' set, so user99 > and user20 joins the 20@profile1 conference, and other users joins > the 20@profile2 conference, that doesnt seem to work neither. I > have pasted conference.conf.xml file as well for ur review. > > <include> > <context name="test"> > <extension name="global" continue="true"> > <condition> > </condition> > </extension> > > <extension name="conf"> > <condition field="destination_number" expression="^(20)$"> </condition> > <action application="answer"/> > <action application="conference" data="bridge:20@default+flags{endconf}:user/10"/> > <action application="conference" data="20@default"/> > </condition> > </extension> > </context> > </include> > > > <configuration name="conference.conf" description="Audio Conference"> >  > <advertise> > <room name="3001@$${domain}" status="FreeSWITCH"/> > </advertise> > >  >  > <caller-controls> > <group name="default"> > <control action="mute" digits="0"/> > <control action="deaf mute" digits="*"/> > <control action="energy up" digits="9"/> > <control action="energy equ" digits="8"/> > <control action="energy dn" digits="7"/> > <control action="vol talk up" digits="3"/> > <control action="vol talk zero" digits="2"/> > <control action="vol talk dn" digits="1"/> > <control action="vol listen up" digits="6"/> > <control action="vol listen zero" digits="5"/> > <control action="vol listen dn" digits="4"/> > <control action="hangup" digits="#"/> > </group> > </caller-controls> > >  > <profiles> >  > <profile name="default"> >  > <param name="domain" value="$${domain}"/> >  > <param name="rate" value="8000"/> >  > <param name="interval" value="20"/> >  > <param name="energy-level" value="300"/> > >  > <param name="member-flags" value="dist-dtmf"/> > >  >  >  >  >  >  > >  >  >  >  >  >  >  >  >  >  >  >  >  > <param name="caller-id-name" value="$${outbound_caller_name}"/> >  > <param name="caller-id-number" value="1234"/> >  >  >  > <param name="comfort-noise" value="true"/> >  >  >  > </profile> > > > > > > > > > > > > > > > > > > > > > On 6/30/2011 1:07 PM, Michael Collins wrote: >> Can you pastebin exactly what you are doing to establish the >> call? Including any relevant dialplan entries. Also, if you have >> modified conference.conf.xml we would like to see that also. >> >> -MC >> >> On Thu, Jun 30, 2011 at 10:59 AM, ran zhang >> <rzhang(a)gosilverplus.com <mailto:rzhang@gosilverplus.com>> wrote: >> >> hi all: >> >> I'm trying to creating a conference, so when first member >> enters the >> conference, he has to invite another members >> and have at least 1 other member to join to have the >> conference established, so i'm >> using bridging conference. >> >> I need this conference to be terminated when the original >> creator of the >> conference leaves no matter how many members are still left >> in the conference. >> >> i'm trying to set 'endconf' flag in a bridging conference using >> 'bridge:confname+flag{endconf}:user/10', >> so it wil invite user extension 10, but its giving me config >> error while >> running. >> >> can someone tell me what to do to solve this problem or get >> around? the key >> is i only want the original member to be able to terminate >> the conference >> when he leaves, not other members assuming there are at least >> 2 members. >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@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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NAT Traversal on SFLphone - FS not Auto Changing port
by Avi Marcus 30 Jun '11

30 Jun '11

http://pastebin.freeswitch.org/16627 I've got a Linksys ATA behind NAT and a softphone behind NAT. Both seem to register with the same UDP-NAT string and both have the same type of contact string.. but on one FS is rewriting the RTP IP to work properly - the 1102 works, but on 1000 I don't seem to be getting any audio. Is there some hidden parameter I can't see affecting this? Both are dialing the same extension 9664 default MOH stuff... This softphone actually seems to have a responsive gui in linux. If I can get the audio to actually work, that would great! (Oh, it has tls/srtp too, it seems) Thanks! <http://pastebin.freeswitch.org/16627> -Avi Marcus

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