- FreeSWITCH-users - FreeSWITCH Mailing Lists

Where FreeSWITCH writes some files
by Andy Spitzer 15 Dec '08

15 Dec '08

Woof! It appears that FreeSWITCH writes freeswitch.history freeswitch.log freeswitch.pid freeswitch.xml.fsxml to the -log directory. Is there a way to put the files other than freeswitch.log into the -db directory instead? In my environment we archive and rotate everything in the log directory (which includes logs beside FreeSWITCH's), and these other FreeSWITCH files are getting rotated. Yeah, I can explicitly exclude them, but to me it seems those really belong in the -db directory anyway, as they are inherently data needed for the current executable of FreeSWITCH, and not logs. --Woof!

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how to handle returned sip 302 dialplan
by Chav Paskov 15 Dec '08

15 Dec '08

Good morning everybody, I would like to know what would be the best way to process the info in returned back sip 302 message. Let me be a bit more specific. i'm sending INVITE as pasted below to z.z.z.z *INVITE* sip:17402740539@z.z.z.z:5000 SIP/2.0^M Via: SIP/2.0/UDP z.z.z.z:5000;branch=z9hG4bK215b.e6ae7e74.0^M Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bKaa6257694041475e0659eeb0019a0bbc^M *From: "Bob Caller" <sip:6144102508@y.y.y.y:5060>;tag=CCC7.1F37^M To: <sip:17402740539@x.x.x.x:5060>^M *Call-ID: 15821337-3367080415-112396(a)msc-01.voice.example.net^M CSeq: 1 INVITE^M Contact: <sip:6144102508@y.y.y.y:5060>^M Content-Type: application/sdp^M Max-forwards: 69^M Session-expires: 3600;Refresher=uac^M Supported: timer^M Content-Length: 200^M ^M 1192 v=0^M o=iserver 16123 16124 IN IP4 y.y.y.y^M s=sip call^M c=IN IP4 64.112.188.81^M t=0 0^M m=audio 55824 RTP/AVP 18 0 101^M a=sendrecv^M a=fmtp:101 0-15^M a=rtpmap:101 telephone-event/8000^M a=ptime:20^M the reply from z.z.z.z back is sip 302 SIP/2.0 *302 Moved Temporarily*^M Via: SIP/2.0/UDP z.z.z.z:5000;branch=z9hG4bK215b.e6ae7e74.0^M Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bKaa6257694041475e0659eeb0019a0bbc^M From: "Bob Caller" <sip:6144102508@y.y.y.y:5060>;tag=CCC7.1F37^M To: <sip:17402740539@x.x.x.x:5060>;tag=1087415105^M Call-ID: 15821337-3367080415-112396(a)msc-01.voice.example.net^M CSeq: 1 INVITE^M *Contact: <sip:2145551234@64.112.188.84>;npdi^M *User-Agent: eXosip/3.1.0^M Content-Length: my question is: Is it possible to send the call to z.z.z.z , receive the SIP 302 , process the data in Contact field and redirect to the new destination contained in *Contact: <sip:2145551234@64.112.188.84>;npdi^M *without closing the session. i red something about <action application="set" data="continue_on_fail=true"/> but i'm not sure how to use it. Any ideas on this matter will be highly appreciated. Best Regards Chav

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Newbie Questions
by Saeed Ahmed 15 Dec '08

15 Dec '08

Hi, I am very new Freeswitch. Till now I've some experience with Asterisk. Can someone explain me the following things: 1. Can I connect my TDM switch to Freeswitch? My switch can speak Dss1/ss7, SS7 is more preferable 2. Is freeswitch similar to nextone SBC? (http://www.nextpointnetworks.com/) 3. Does freeswtich support codec translation? These are very basic questions at start. When I'll go deeper into it then there could be more questions. Kind Regards Saeed

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Recording Pause
by Robert Clayton 15 Dec '08

15 Dec '08

All, I was thinking since there is no direct functionality for the person being recorded to pause the recording could this be done indirectly. For example if using the functionality to record only when voice is present could the audio stream be preprocessed to evaluate the dtmf where if a specific key was pressed the audio stream would no longer be passed or passed as blank to the recording functionality so an implicit pause would be created. And when the person recording wished to continue pressing a key would signal FS to pass the audio again therefore reestablishing recording to the same file? Bob

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Re: [Freeswitch-users] Multiple mod_portaudio channels
by Giovanni Maruzzelli 15 Dec '08

15 Dec '08

You can try making the lynx appear to Operating System like 8 single soundcards (mono 1in 1out). You can do this on Linux via the /etc/asound.conf Then for each soundcard you created, you can start one portaudio channel Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039 On Mon, Dec 15, 2008 at 11:56 AM, Clifford <freeswitch(a)ptmm.com> wrote: > I need to set up a system that would allow multiple mod_portaudio channels > to be configured. > > The calls would come in over SIP and any calls would be assigned a specific > port on the multi-port sound card. > > I would probably set it up so specific extensions go to specific ports, for > example 1001 would be port 1 and 1008 would be port 8 on the sound card. > > The documentation on the mod_portaudio module is very slim. In fact I would > say it is nearly undocumented. Is it capable of handling multiple audio > device ports (such as an 8-port LYNX sound card)? > > If so how would it be interfaced/configured? > > Thanks, > > Clifford > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)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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Interrupting read application with DTMF
by Jan Kubr 15 Dec '08

15 Dec '08

Hi, I have been having some troubles with the read application for quite a while which I haven't been able to solve yet. I have Freeswitch connected to a SIP gateway to accept calls from a landline-like number. For the incoming calls I have a simple testing dialplan: <action application="read" data="1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 #"/> <action application="phrase" data="spell,${res}"/> The behavior I have a problem with is that the read app should terminate when I press a digit and the execution should jump to the next action - meaning the playback of the file should be interrupted. The problem is that when I call the public number from my cell phone this works only about 50% of the time. In the other cases I need to wait for the wav file to be played (or press the digit two or three times). When using a SIP phone it always works. Today I tried to convert the wav file the read app plays to the GSM format and found out it fixed the problem! Now I can almost always interrupt the read app with DTMF from my cell phone. Doing the same from my SIP phone doesn't work well though when the file is GSM. Can someone explain me what is going on here and what is the right approach? I'm on revision 10751. I've tried to set a few configuration variables based on suggestions from this list, but it didn't make any difference. Thanks, Jan Kubr

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Speed Dial Emulation
by jflowers 15 Dec '08

15 Dec '08

How do I emulate a speed dial setup. That is, from extension 1003 I dial just a 1 ( or 2, or 3 etc.) and nothing else and freeswitch dials a PSTN number. Is there software to do this? -- View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21016167.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Multiple mod_portaudio channels
by Clifford 15 Dec '08

15 Dec '08

I need to set up a system that would allow multiple mod_portaudio channels to be configured. The calls would come in over SIP and any calls would be assigned a specific port on the multi-port sound card. I would probably set it up so specific extensions go to specific ports, for example 1001 would be port 1 and 1008 would be port 8 on the sound card. The documentation on the mod_portaudio module is very slim. In fact I would say it is nearly undocumented. Is it capable of handling multiple audio device ports (such as an 8-port LYNX sound card)? If so how would it be interfaced/configured? Thanks, Clifford

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Openzap ERROR can't dial
by Franziska Röhler 15 Dec '08

15 Dec '08

Hello, Im a newbie in FS and my English isnt very good but I try to explain my problem. Hopefully you can understand me. :-) I have a Linux-Server with a Digium Wildcard TE110P. I install and configure openzap (PRI/ISDN). But when I make an outbound call, I have an error. FS cant dial. [ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING] I have this warnings too, when no call is done: 2008-12-15 10:11:14 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6 2008-12-15 10:11:15 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 7 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:1 (1:1) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:2 (1:2) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:3 (1:3) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:4 (1:4) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:5 (1:5) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:6 (1:6) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:7 (1:7) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:8 (1:8) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:9 (1:9) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:10 (1:10) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:11 (1:11) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:12 (1:12) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:13 (1:13) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:14 (1:14) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:15 (1:15) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:16 (1:17) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:17 (1:18) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:18 (1:19) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:19 (1:20) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:20 (1:21) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:21 (1:22) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:22 (1:23) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:23 (1:24) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:24 (1:25) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:25 (1:26) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:26 (1:27) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:27 (1:28) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:28 (1:29) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:29 (1:30) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:30 (1:31) has alarms [YELLOW] 2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:31 (1:16) has alarms [YELLOW] 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:1 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:2 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:3 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:4 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:5 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:6 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:7 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:8 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:9 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:10 2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for channel 1:11 What do I wrong? I hope you can help me! Here my configuration: OPENZAP.CONF [span zt] name => OpenZap number => 1 trunk_type => e1 b-channel => 1-15, 17-31 d-channel => 16 OPENZAP.CONF.XML <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <pri_spans> <span id="1"> <param name="mode" value="user"/> <param name="dialect" value="euro"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> ZT.CONF [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 DEFAULT.XML <extension name="outgoing_pri"> <condition field="destination_number" expression="^([0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="openzap/1/A/${dialed_ext}"/> </condition> </extension> When I have forget to display some Configuration to , you can tell me!

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Bug in Q921.c? AVAYA-PBX issue
by Helmut Kuper 15 Dec '08

15 Dec '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04 server in a non-root environment. We experienced a timer problem which led to this FS console error message: [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries During anylizing this we found that q921 T203 is never reset when link is in state "Multiple Frame Mode Established" and SABME frames are received by FS. So it must timeout regardless if SABME frames are received or not. Additionally we found that the default T203 value (10 sec) was too short for AVAYA (it has to be >=19 sec) To fix the problem we changed two things in q921.c: Change T203 default value from 10 sec to 20000 sec Line 406: trunk->T203Timeout = 20000; Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each received SABME frame Line 1996: Q921T203TimerReset(trunk, tei); After recompiling FS the Error disapeared. Next week we will do some calls over the link to make sure there are no other side effects. Is it planned to make the q921 timeouts configurable in openzap.conf or in openzap.conf.xml? best regards Helmut PS: My openzap configs: openzap.conf [span wanpipe PRI_1] trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="7"/>    </settings> <pri_spans> <span name="PRI_1">  <param name="q921loglevel" value="debug"/> <param name="q931loglevel" value="debug"/> <param name="mode" value="user"/> <param name="dialect" value="Oh this is not my dialect"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> Very interesting here is, that the "dialect" parameter doesn't seem to have an effect on FS. I use that one above without any errors or warning and I guess that was not intended. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB =geXp -----END PGP SIGNATURE-----

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