February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Skypiax- how to know which card to use?
by Doc 10 Mar '10

10 Mar '10

Hey guys, I am attempting to setup Skypiax and Freeswitch on Ubuntu Hardy. I've had a couple of previous problems which updating to SVN trunk release seemed to resolve, but I've got one lingering one to ask about. When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS picks up and rings the desired extension. No problems there. What happens though is, the person that is on the extension in the office can hear the calling party just fine, but the calling party hears stuttered choppy sound. I've already tested my internet connection, and through QOS have assigned my test box 512K up and 512K down, so on one test call bandwidth should not be an issue. What do you think causes the stuttering on my audio for the calling party? Do you think it might be a codec? I have everything setup right now to use g711u. I'd like to use something smaller and higher quality, but I'm not sure what is a better option. I had made some phone calls as listed above and they tested out just fine, but in that same time frame I also recompiled Alsa drivers with the Skypiax dummy file, so do you think there might be something in that causing the stutter? Looking for some advice. Thanks, Doc

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How to tie context to a gateway?
by Nikolay Kondratyev 10 Mar '10

10 Mar '10

Hi all, I have several gateways in the external profile. Let's say GW1 and GW2. I'd like to process calls from the GW1 in the context C1 and calls from GW2 in the context C2. Parameter "context", as far as I understand works for the whole profile, not for individual gateways in the profile. How do send calls from GW1 into context C1? What will be a good practice to do that? Thanks in advance, Nikolay.

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actition after a set time during call
by Todd 10 Mar '10

10 Mar '10

Hey List- I want to have nibblebill pause after a certain time during a call. I was wondering what the best way to put this into the dialplan is? Still kind of new to this.. <action application="nibblebill" data="pause"/> Is the action I want to implement 2 minutes into a call. what is the best way to do this? I have the nibblerate set in the individual extension XMLs and the nibblerate heartbeat set in the nibble.conf.xml Thanks

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IVR and Erlang
by Mark Sobkow 09 Mar '10

09 Mar '10

I'm having a little trouble with Erlang and IVRs. Specifically, the IVR configuration file is auto-loaded before the initialization code has a chance to tell Freeswitch to get it's configs from Erlang. As a result, it's not querying Erlang for the configuration file. Also, because there is no mod_ivr, I can't just do a "reload mod_ivr" after the Erlang communications has been initialized. Any idea how to get Freeswitch to load it's IVR configs from Erlang? -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

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FScomm
by Madovsky 08 Mar '10

08 Mar '10

Hi, is http://wiki.freeswitch.org/wiki/FSComm available yet ? because QT framework link has broken and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root) Thanks Franck

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FSComm basic issue
by Suneel Papineni 08 Mar '10

08 Mar '10

Hi, I am trying to use FSComm with Freeswitch and facing following issues. 1. Using pre-build binary (windows), when the application is started FSComm is getting Registered properly. When I tried to make a call, UI displays Dialing... but unable to see any SIP (INVITE) messages in wireshark traces. After sometime UI displays with message "Call with (destination number) failed with reason DESTINATION_OUT_OF_ORDER though destination number is registered with another FSComm" 2. Also I am unable to see any logs generated in the log folder. Downloaded the latest source code (Freeswitch 1.0.5 latest updated as on 26/02/10 at 4am) and tried to build FSComm. Build was succeeded. Application (FSComm) also started and displayed with UI. When I try to change the preferences, it has thrown Porta Audio Error saying "Error Querying Audio Devices" even though proper audio devices are present. Also it doesn't create folders like "conf", "mod". Even after copying all the required dll's and mod files (as specified in FSComm wiki pages), application is throwing the same error. I am using Windows XP machine. Built a Debug & Release version with 32-bit option. If someone has built FSComm for windows environment and is working fine, could you please let me know if there are any additional things I need to do to make it work. Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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415 Unsupported Media Handling
by Jerry Richards 05 Mar '10

05 Mar '10

I have two types of devices, one supports text/html MESSAGE content and one that only supports text/plain MESSAGE content. When I send an IM from the first to the second, the second replies with 415 Unsupported Media Type (as shown below). Section 8.1.3.5 of RFC 3261 says the sender should retry using the media type acceptable to the receiver (in this case: plain/text). The problem I have is that Freeswitch doesn't pass the error back to the sender (nor does it retry itself using plain/text). So the IM is lost. Does anyone see the reason why the error is not being handled correctly? ------------------------------------------------------------------------ send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446: ------------------------------------------------------------------------ MESSAGE sip:5398@192.168.72.141:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre Max-Forwards: 70 From: "5382 on 141" <sip:5382@192.168.72.141>;tag=66661130 To: "5398" <sip:5398@192.168.72.141:5062> Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0 CSeq: 127444135 MESSAGE Contact: <sip:mod_sofia@192.168.72.141:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: text/html Content-Length: 63 <font size="2" face="Arial">hello this is Jerry from Teo</font> ------------------------------------------------------------------------ recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390: ------------------------------------------------------------------------ SIP/2.0 415 Unsupported media type Via: SIP/2.0/UDP 192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168.72.14 1 From: "5382 on 141" <sip:5382@192.168.72.141>;tag=66661130 To: "5398" <sip:5398@192.168.72.141:5062> Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0 CSeq: 127444135 MESSAGE Date: Fri, 26 Feb 2010 18:43:24 GMT User-Agent: MobilityGateway-2.0.34078 Server: MobilityGateway-2.0.34078 Accept: text/plain Content-Length: 0 Best Regards, Jerry

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about pizza demo
by Madovsky 05 Mar '10

05 Mar '10

Hi, I'm trying to change the language of pizza demo script. is it need to change only words inside addItemAlias() ? Many thanks Franck

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How to exchange media when I developed new endpoint module?
by 杨江骅 03 Mar '10

03 Mar '10

Hi, I am developping a new endpoint module, now I can make an inbound call and execute IVR. When I make an outbound call and bridge the inbound leg and outbound leg, I receive remote alerting and pickup remote phone but there isn't any voice exchange. So how to exchange media next? ---------------------------------------------------------------------- gtalk: spiritonly(a)gmail.com

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Freeswitch/Openzap dials out on disconnected FXO line
by Robert Hadley 02 Mar '10

02 Mar '10

When dialing out, Freeswitch/Openzap is not detecting that an analog FXO channel is disconnected and tries dialing out on the channel anyway. No error is reported. The call doesn't timeout until a minute later. Shouldn't Freeswitch/Openzap skip over a disconnected channel to the next connected channel? I have configured a Sangoma A200 FXO card as a FXO span. [span wanpipe FXO] name => PSTN Line 1 number => 4253491059 fxo-channel => 2:3 name => PSTN Line 2 number => 4253491058 fxo-channel => 2:4 The wanpipe driver does detect and report when a CO line is connected or disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as reported in the log. /var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3: FXO Line is disconnected! FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT [ALARM_TRAP][3:1] STATE [DOWN] /var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4: FXO Line is connected! FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT [ALARM_CLEAR][3:2] STATE [DOWN] I have the dialplan configured to use the next available port in the FXO span (there will be more than 2 channels later). <extension name="OutgoingFXO"> <condition field="destination_number" expression="^7(\d+)$"> <action application="bridge" data="openzap/FXO/a/$1"/> </condition> </extension> Here is a portion of the log when that shows dialing out on a disconnected analog FXO channel. EXECUTE sofia/internal/5410@192.168.72.45:5060 bridge(openzap/FXO/a/93491045) 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound channel OpenZAP/3:1/93491045 2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7] 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269 (OpenZAP/3:1/93491045) State Change CS_NEW -> CS_INIT 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/3:1/93491045 [BREAK] 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1 from DOWN to DIALING 2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486 OpenZAP/3:1/93491045 does not support the proxy feature, disabling. 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread starting. 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state handler on 3:1 for DIALING 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/3:1/93491045) Running State Change CS_INIT 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/3:1/93491045) State INIT 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045) State Change CS_INIT -> CS_ROUTING 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/3:1/93491045 [BREAK] 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/3:1/93491045) State INIT going to sleep 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/3:1/93491045) Running State Change CS_ROUTING 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/3:1/93491045) State ROUTING 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045 CHANNEL ROUTING 2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/3:1/93491045 [BREAK] 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/3:1/93491045) State ROUTING going to sleep 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/3:1/93491045) State CONSUME_MEDIA 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep Thanks, Robert

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