August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Software phone
by Ashish Mishra 04 Aug '13

04 Aug '13

Hello, in order to use a softphone do i need to install freeswitch on the client machine also...or simply i can download the softphone and can connect to a server machine on which freeswitch is already installed...???

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How to configure DTMF detection using 'info' and 'rfc2833' both?
by Thomas Lee 04 Aug '13

04 Aug '13

How to configure DTMF detection using 'info' and 'rfc2833' both?Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-configure-DTMF-detecti… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Resource used on server for conferencing
by Mahendra Bhegade 03 Aug '13

03 Aug '13

Hi, I am using flowroute to access incoming calls and then conference additional phones using flowroute SIP gateway. I would like to understand what resources are being used on the server that is hosting the conference calls. Now if flowroute is handling all the call what resources are being consumed on the freeswitch server. CPU, bandwidth, what else ? If these are cellphone versus landlines does it make any difference ? Please share some insight.

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Mod_directory: "connecting call" sound?
by Brian Foster 03 Aug '13

03 Aug '13

I can't find it anywhere in the wiki or in the config files, but is there a way to specify a "connecting call" sound? Example: A caller enters the directory application, he/she selects an entry. When they select the entry, it goes straight to ringing. Before that call is connected, can we specify that it plays a sound such as "One moment while I connect your call"? I realize this might be a feature request, in which case I'll probably take it upon myself to rummage through the source code and come up with a patch to go with a JIRA ticket. Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana

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Restrict inbound codecs for specific user
by Steven Ayre 03 Aug '13

03 Aug '13

Is there any variable that can be set in the user directory to restrict/override the profile's inbound codec preferences for a specific user account? I'm aware I can do this with late negotiation and setting absolute_codec_string, but is prefer to reject calls without a supported dialplan before they hit the dialplan and go to a bridge. -Steve

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Restrict inbound codecs for specific user
by Steven Ayre 03 Aug '13

03 Aug '13

Is there any variable that can be set in the user directory to restrict/override the profile's inbound codec preferences for a specific user account? I'm aware I can do this with late negotiation and setting absolute_codec_string, but is prefer to reject calls without a supported dialplan before they hit the dialplan and go to a bridge. -Steve

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Naming xml files
by Richard Mace 03 Aug '13

03 Aug '13

Hi, In the freeSWITCH 1.2 book, it talks about xml files, such as an incoming route being called 01_ and then a name such as iptel.xml Is the "01_" that important/a good idea/needed or is it just so that it is displayed as the first entry in the directory when browsing the files? Thanks in advance Richard

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Resources used on server for conferencing
by Mahendra Bhegade 03 Aug '13

03 Aug '13

Hi, I am using flowroute to access incoming calls and then conference additional phones using flowroute SIP gateway. I would like to understand what resources are being used on the server that is hosting the conference calls. Now if flowroute is handling all the call what resources are being consumed on the freeswitch server. CPU, bandwidth, what else ? If these are cellphone versus landlines does it make any difference ? Please share some insight. Mahendra

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Best practices for memory pool usage in modules?
by Nathan Neulinger 02 Aug '13

02 Aug '13

I'm looking into memory leak issues with mod_skinny, and it seems clear that it's repeatedly allocating memory when it doesn't really need to hang onto it permanently. Since it has both a long standing connection to each device (listener) and has requests coming in regularly that currently are allocating memory and appear to be leaking it since the listener stays around. Would having the code allocate a new temporary sub-pool for each request that comes in be appropriate, or would it be better to just use regular malloc for those allocate/use/destroy cases? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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No audio on either of an established call
by Peter 02 Aug '13

02 Aug '13

I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end. The set up is as follows: Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000) Using Freeswitch 1.2.12 on CentOS (installed via RPM) Freeswitch has two interfaces: external - 10.1.1.206 internal - 10.10.10.206 Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch. The SIP negotiation is working as I can call either Linphone client. I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch. I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio. Any pointers on how to track down the issue? Thanks Peter

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