September 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ESL: uuid events with filtering
by Jayesh Nambiar 16 Sep '11

16 Sep '11

Hi, Is it possible to filter specific events for a particular uuid through ESL. I am using it from a PHP application as follows: $sock->SendRecv("events plain ALL"); $sock->SendRecv("filter unique-id 12345"); $sock->sendRecv("filter Event-Name CHANNEL_ANSWER"); $sock->sendRecv("filter Event-Name CHANNEL_HANGUP"); $sock->sendRecv("filter Event-Name CHANNEL_HANGUP_COMPLETE"); In the case above, I start seeing the events coming from CHANNEL_ANSWER and thereafter all events for that uuid is received. Even if I write: $sock2->SendRecv("events plain ALL"); $sock2->SendRecv("filter unique-id 12345"); $sock2->sendRecv("filter Event-Name CHANNEL_HANGUP_COMPLETE"); I get all the events belonging to uuid 12345 starting from CHANNEL_ANSWER event name. I also tried doing "myevents 12345 plain" and then applying filters but it still doesnt work as expected. I tried googling a bit but could'nt find much help. Anybody pointing me to a proper direction in this regards is highly appreciated. Thanks, --- Jayesh

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Re: [Freeswitch-users] changing default FS for G729
by Mi Ke 16 Sep '11

16 Sep '11

Please disregard my below request - the problem was a buggy endpoint. FS works as expected. Thanks / Mike ----- Original Message ----- From: Mi Ke Sent: 09/15/11 10:39 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] changing default FS for G729 Hi ALL ! I'm trying to change default frame size from 30 to 40 bytes for G729 (commercial version) but setting correspondent codec prefs to G729@40i (vars.conf) gives no effect. Codec still starts with 30 bytes frames: 2011-09-15 19:00:37.538174 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16@8000hz 1 channels 30ms Is it possible to do that with mod_com_g729 ? Thanks, Mike

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changing default FS for G729
by Mi Ke 15 Sep '11

15 Sep '11

Hi ALL ! I'm trying to change default frame size from 30 to 40 bytes for G729 (commercial version) but setting correspondent codec prefs to G729@40i (vars.conf) gives no effect. Codec still starts with 30 bytes frames: 2011-09-15 19:00:37.538174 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16@8000hz 1 channels 30ms Is it possible to do that with mod_com_g729 ? Thanks, Mike

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Answer issue on inbound call
by Max Alex 15 Sep '11

15 Sep '11

Hi, I have installed freeswitch latest version with sangoma card A200 as well, I have installed and configured freetdm module with wanpipe drivers for freeswitch, We are properly receiving the inbound calls in public context and then we are routing that call to 1001 extension, it is properly routing to 1001 as well, but we have one issue while routing on 1001. Here is the issue description. I am calling from my cell phone to that DID number of pri line, and then it will start ringing on 1001 extension, When 1001 extension start ringing the call is answered on my cell phone, it is something like freeswitch preanswer or autoanswer the call, how can i stop this answer call when it is ringing on 1001 extension, Waiting for good reply. Thanks, Max Alex

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calling Skype-users and receiving calls from Skype-users
by bill evergreen 15 Sep '11

15 Sep '11

Hello, i would like to know, if it's possible from FreeSWITCH to call Skype-users and to receive calls from Skype-users. As I understand, with "SipTheeSkype" that is to some extent possible. But I am not sure whether or not on the Skype client, some addtional software is required (what is not the thing i'd like) What approach would you recommend to go? Thank's a lot for any help! Bill

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DTMF Info
by Antonio Teixeira 15 Sep '11

15 Sep '11

Good Morning. I have a soft phone that uses SIP INFO to deliver the DTMF keys to FS. Now the problem is that FS always reported INFO is not expected on this channels so off to the wiki i go and i set in the profile dtmf-type = info. sofia.c:6439 IGNORE INFO DTMF(2) (This channel was not configured to use INFO DTMF!) So anyone can help me out with this ? Best Regards Antonio Teixeira

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Lua API for CouchDB
by Chad Phillips -- Apartment Lines 15 Sep '11

15 Sep '11

For those interested in connecting to CouchDB (http://couchdb.apache.org) from FreeSWITCH, I've written a Lua API (http://www.lua.org) for it, available here: https://github.com/thehunmonkgroup/luchia With Lua being the recommended scripting language for FreeSWITCH, this seems an excellent way to work with Couch. For those that have the LuaRocks package manager installed, it's a simple 'luarocks install luchia' to install. I'm interested in exploring CouchDB as the datastore for a distributed, fault tolerant voicemail system -- I think it's attachment feature and bi-directional synchronization are well suited for the task. Chad

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DrayTek 2820n routers and FreeSWITCH
by Gavin Henry 15 Sep '11

15 Sep '11

Just a quick note to all to say if you have any one-way audio issues or SIP stability issues with these routers and FreeSWITCH then downgrade to firmware 3.3.3 and all issues disappear. This is what DrayTek recommend. Thanks, Gavin. -- http://www.suretecgroup.com http://www.surevoip.co.uk

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One Way Audio - Auto Change RTP port?
by Dan Lan 15 Sep '11

15 Sep '11

Hi, I run into a weird situation. My media gateay handle voice call with 2 different RTP ports for send & receive Here is what happened. (ps: both gateway and FS are all on public IP, no NAT involved) 1. Incoming call INVITE from gateway to FS Connection Information (c): IN IP4 100.100.100.100 (This is my media gateway IP address) Media Description, name and address (m): audio 5294 RTP/AVP 18 0 4 2. FS response with session progress with media information Connection Information (c): IN IP4 200.200.200.200 Media Description, name and address (m): audio 22428 RTP/AVP 0 3. I start to see some RTP traffic exchange between FS and GW from FS (22428) --> GW (5294) from GW (5292) --> FS (22428) please note: the GW use two DIFFERENT PORT for RTP, one for sending and one for receiving 4. For a while (about 5 secs, I think) The RTP flow change on FS side to become, (there is no RTCP packet during the time) from FS (22428) --> GW (5292) from GW (5292) --> FS (22428) In other word, the FS now sending RTP to 5292 instead of 5294 (which was intended in INVITE SDP message) And, of course, I cannot hear the voice on GW side after this. Anyone encounter this before? Are there any paramaters that might involved in this auto changing RTP port behavior of FS? Any direction for me is appreciated, I will play around with this, and post back my result to community. Dan Lan

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call forwarding how to test
by Aakash 14 Sep '11

14 Sep '11

Hi, I am trying to call forward from one extension to another extension.I configured default.xml and features.xml by following below link http://wiki.freeswitch.org/wiki/Call_Forward_Example Can anyone help me how do i test or activate the call forwarding in the extension. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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