November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Listen to a file, while recording?
by Dennis 02 Dec '08

02 Dec '08

hi, i would like to be able to listen to conversations, while they are ongoing. this should not happen over a phone. i would like to be able to have a link or something in my admin-area, where i can click, if i want to listen to a conversation. i thought about to start a record with socket inbound on a specific uuid and while the recording is done, i would like to play the file. but i have the feeling, that this is not possible, while the file is not finished. is there a way to stream the audio over the web? thanks for your help.

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SIP INVITE timeout
by Gabriel Kuri 01 Dec '08

01 Dec '08

I have a phone that is registered to FS but is no longer available (Internet connection down, phone turned off, etc.). The registration still exists in the sip_registrations table (not expired yet), but the phone is not reachable on the network. According to my dialplan, if the bridge to the phone fails after 20 seconds, the call should be forwarded to a different box for handling (see dialplan below). <extension name="My Extension"> <condition field="destination_number" expression="1213XXXXXXX"> <action application="info"/> <action application="set" data="originate_timeout=20"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="${sofia_contact(default/1213XXXXXXX(a)mydomain.net"/> <action application="bridge" data="sofia/default/1213XXXXXXX(a)box.mydomain.net"/> </condition> </extension> If the phone is down and not responding to the INVITEs, it appears my carrier is canceling the SIP INVITE to FreeSWITCH after about 10 seconds. My timeout is 20 seconds. Is there anyway to deal with this situation, without going back to my carrier and asking them to increase their timeout on an INVITE? Call Progress: Carrier -> FS (INVITE) FS -> Carrier (100 Trying) <10 seconds pass while FS is attempting to contact the phone> Carrier -> FS (CANCEL) FS -> Carrier (200 OK) FS -> Carrier (487 Request Terminated) Carrier -> FS (ACK) Thanks ... Gabe

5 7

Leg A terminated by Leg B on a uuid_bridge
by Simon Tang 01 Dec '08

01 Dec '08

Hello, I'm using event socket outbound, and have an issue where, after a bridge ends and is terminated by Leg B, Leg A is also terminated. Here's the call flow: 1. Call comes in (Leg A), session created, play welcome message. 2. From this session, originate and dial out using api originate 3. After the target answers (Leg B), bridge the 2 calls using api uuid_bridge 4. Leg B hangs up. 5. Leg A will be terminated. After step 4, Leg A is terminated. I do not want Leg A to hang up. I've tried setting "hangup_after_bridge=false" prior to the call, and that doesn't work. Having said that, I tried a similar test which does not end Leg A's call after Leg B hangs up, but I can't use this solution because, functionally, does not accomplish what I want it to do (i.e., I want to perform some actions on Leg B prior to the bridge, like send some DTMF tones, playback some messages, etc). I did not need to set the "hangup_after_bridge" variable (default should be false anyway). 1. Call comes in (Leg A), session created, play welcome message. 2. From this session, do a bridge by doing an execute bridge. 3. The target answers (Leg B) 4. Leg B hangs up. 5. Leg A will still be active. Any ideas would be appreciated. Thanks! Simon

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Multi FS behind same NAT, PRACK goes to wrong port
by x y 01 Dec '08

01 Dec '08

Hy! There are two different FS behind the same NAT, and there were Reigstration Failures about one or to times a day. The gateway status turned down, then I got 503 error codes. Then I set up the ext-ip to STUN, as the wiki requests it. Now I facing the next problem: Start the call, all goes right, INVITE goes to port 1352, then after 183 Session progress from port 1352, the PRACK package goes to 5060 instead of 1352, wich messes up the call procedure. Is there anyway to force PRACK to the port to the INVITE has been sent before? Cheers, Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!

3 6

Failed to create a udp socket on port 5060
by Faisal Maqsoodi 01 Dec '08

01 Dec '08

I m trying to play some basics on FreeSwitch plus twinkle softphone. Following error msg is displayed on the softphone: Failed to create a udp socket on port 5060. Address already in use & following on freeswitch window: X Error: BadWindow (invalid Window parameter) 3 Major opcode: 20 Minor opcode: 0 Resource id: 0x30001be Plz help me solving this. faisal

5 6

Configuring different codecs for internal and external calls.
by Maxim Karp 01 Dec '08

01 Dec '08

Hello, How would I configure global_codec_prefs and outbound_codec_prefs for the following: All calls between freeswitch extensions GSM. All calls going out from an extension on Freeswitch using GSM (between the endpoint and freeswitch) and then out to a SIP gateway using PCMA or PCMU. Calls coming in from another SIP gateway to freeswitch using PCMU or PCMA and then transcoded to a local extension using GSM. Is this it? <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/> Thanks, Maxim.

2 2

Sound file as ringback
by Jan Kubr 30 Nov '08

30 Nov '08

Hi, I would like to play a sound file while bridging the call. The wiki says that the ringback variable can be set to a file name as well. However, I haven't been able to get this to work (as I thought it would). <action application="set" data="ringback=ring.wav"/> <action application="bridge" data="user/jan@$${domain}" /> I have try different format of files (from 8KHz mono wavs to MP3s, all of which play fine via playback) and some caused the bridge to be finished immediately (with NO_USER_RESPONSE), some make it generate crazy beeping, but none is played while the phone is ringing on the other end. No error or anything obvious to me in the log file (even in debug mode). Can someone please tell me how this is supposed to be used? Cheers, Jan

2 1

playAndGetDigits via event socket
by Peter P GMX 30 Nov '08

30 Nov '08

has anybody already used playAndGetDigits via event socket and has a working example? Best regards Peter

1 0

Linksys/Cisco SPA400 (ATA 4 Line FXO) Now Documented in Wiki
by Karl Vesterling 29 Nov '08

29 Nov '08

Folks; I've just taken the time to document the Sipura, err, Linksys, errr Cisco SPA400 4 line FXO Analog Telephone Adapter in the Wiki. http://wiki.freeswitch.org/wiki/SPA400_FreeSwitch_HowTo If anyone uses these ATA's and has questions about it let me know and I'll see if I can answer them in the Wiki. Of course, if you know something about it that I haven't documented, by all means, document it in the Wiki. ;-) **Note: MikeJ seems to think that I'm overlooking something pertaining to the registration of the SPA400 without a password. Evidently this pertains to accepting blind auth, but I'd need to know more about that before knowing how to piece it in. Best Regards, Karl J. Vesterling kjv(a)ken-ton.com 202-448-3009 x0

1 0

Answering machine detection
by Keith Wood 29 Nov '08

29 Nov '08

Hi, Does anyone know if it is possible to config freeswitch to do answering machine detection? I see that there is a tone_detect command. Is it possible to use it to achieve a certain level of answering machine detection? Thanks, Keith

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