- FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference fails with speex codec
by Peter P GMX 08 Apr '09

08 Apr '09

I want to use a low bandwidth codec. But whenever I try to use speex I get an error in the conference. We have FS trunk 1288. Switching back to PCMx it works again. Is there any problem with speex and DTMF or with transcoding? Best regards Peter 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2624 sofia_glue_negotiate_sdp() Set Remote Key [1 AES_CM_128_HMAC_SHA1_32 inline:HU0NdX8n18lnRuEKhmJ1O4zSBaolz3wDtOwWIjy8] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:1917 sofia_glue_build_crypto() Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline:r98Rf+0lojftJOPPW9GZon5SZgB6Kg7FsED4cQV3] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[G722:9:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[PCMU:0:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[PCMA:8:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[GSM:3:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2738 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:99:16000:20]/[SPEEX:98:8000:20] 2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2786 sofia_glue_negotiate_sdp() Substituting codec SPEEX@20i@16000h 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error Opening File [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav] [System error : No such file or directory.] 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799 conference_function() Cannot ask the user for a pin, ending call2009-04-08 00:43:24 [NOTICE] mod_conference.c:4800 conference_function() Hangup sofia/internal/723328(a)sip.mydomain.com [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]

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Re: [Freeswitch-users] High CPU load but only few sessions
by B Karthik 08 Apr '09

08 Apr '09

It could be due to registrations. I am currently trying to troubleshoot this problem. I used a sipp scenario to authenticate with fs and register about 2000 different accounts (absolutely no calls made on the test setup). Memory usage increases continuously and does not decrease at all and crosses more than 1 GB in a few hours. On the other hand, there is another fs setup with bypass media turned on and no registrations and is up for almost 45 days without restart and has consumed only about 95 MB of memory and twice as much virtual memory. B Karthik

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need help getting ISDN talking to Cisco 3845
by John Wehle 08 Apr '09

08 Apr '09

Our FreeSWITCH setup has an existing T1 using RBS to talk to a digital modem pack in a Cisco 3845. I'm interested in changing from RBS to ISDN. I changed both sides, restart things, and see FreeSWITCH report: 2009-04-07 18:53:15 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:54:40 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:55:36 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries 2009-04-07 18:55:45 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:46 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:47 [NOTICE] Span:0 Q.921() I frame in invalid state ignored 2009-04-07 18:55:48 [NOTICE] Span:0 Q.921() I frame in invalid state ignored I've attached the configs and Cisco debug below. This is using the native ISDN support in FreeSWITCH with a Sangoma A104d on FreeBSD 6.4. I unfortunately don't currently speak ISDN (though I'm starting to pick up a little as a result of this exercise) ... suggestions / hints regarding what's going on and how to resolve it would be welcomed. -- John ------------------------------ wanpipe2.conf ------------------------------- [devices] wanpipe2 = WAN_AFT_TE1, Comment [interfaces] wbg1 = wanpipe2, , TDM_VOICE, Comment [wanpipe2] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 5 PCIBUS = 5 FE_MEDIA = T1 FE_LCODE = B8ZS FE_FRAME = ESF FE_LINE = 2 TE_CLOCK = MASTER TE_REF_CLOCK = 1 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 2 TDMV_DCHAN = 0 TDMV_HW_DTMF = YES [wbg1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES ------------------------------ zaptel.conf --------------------------------- #Sangoma A104 port 2 [slot:5 bus:5 span:2] <wanpipe2> span=2,0,0,esf,b8zs bchan=25-47 dchan=48 ------------------------------ openzap.conf -------------------------------- [span zt] ; A104D FE 2 1-6 MICA name => Cisco Digital Modem trunk_type => t1 number => 2487 b-channel => 25-47 d-channel => 48 --------------------------- openzap.conf.xml ------------------------------- <pri_spans> <span id="2">  <param name="q921loglevel" value="info"/> <param name="q931loglevel" value="info"/> <param name="mode" value="net"/> <param name="dialect" value="national"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> ------------------------------ Cisco config -------------------------------- controller T1 1/0 framing ESF linecode b8zs cablelength short 220 pri-group timeslots 1-24 interface Serial1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice modem isdn calling-number 2487 no cdp enable ------------------------------ Cisco debug --------------------------------- #show isdn stat Global ISDN Switchtype = primary-ni ISDN Serial1/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x807FFFFF Number of L2 Discards = 2, L2 Session ID = 117 Total Allocated ISDN CCBs = 0 Apr 7 22:53:44.264: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:53:44.264: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:53:44.312: ISDN Se1/0:23 Q921: User RX <- UAf sapi=0 tei=0 Apr 7 22:53:44.312: %CSM-5-PRI: add PRI at 1/0:23 (index 0) Apr 7 22:53:44.312: %ISDN-6-LAYER2UP: Layer 2 for Interface Se1/0:23, TEI 0 cha nged to up Apr 7 22:53:47.268: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:53:47.268: ISDN Se1/0:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0 prepnet-rt# Apr 7 22:53:57.336: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:53:57.336: ISDN Se1/0:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0 prepnet-rt# Apr 7 22:54:11.692: ISDN Se1/0:23 Q921: User RX <- SABMEp sapi=0 tei=0 Apr 7 22:54:11.692: ISDN Se1/0:23 Q921: User TX -> UAf sapi=0 tei=0 Apr 7 22:54:21.760: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:54:21.760: ISDN Se1/0:23 Q921: User TX -> RRf sapi=0 tei=0 nr=0 Apr 7 22:54:51.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:52.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:53.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:54.760: ISDN Se1/0:23 Q921: User TX -> RRp sapi=0 tei=0 nr=0 Apr 7 22:54:55.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:54:55.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:56.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:57.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:58.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:54:59.760: %CSM-5-PRI: delete PRI at 1/0:23 (index 0) Apr 7 22:54:59.760: %ISDN-6-LAYER2DOWN: Layer 2 for Interface Se1/0:23, TEI 0 c hanged to down Apr 7 22:54:59.760: ISDN Se1/0:23 Q931: Ux_DLRelInd: DL_REL_IND received from L 2 Apr 7 22:55:04.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:55:04.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:05.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:05.872: ISDN Se1/0:23 Q921: User RX <- RRp sapi=0 tei=0 nr=0 Apr 7 22:55:06.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:07.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:08.760: ISDN Se1/0:23 Q931: Ux_DLRelInd: DL_REL_IND received from L 2 Apr 7 22:55:13.760: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME Apr 7 22:55:13.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:14.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:15.760: ISDN Se1/0:23 Q921: User TX -> SABMEp sapi=0 tei=0 Apr 7 22:55:15.772: ISDN Se1/0:23 Q921: User RX <- UAf sapi=0 tei=0 Apr 7 22:55:15.772: %CSM-5-PRI: add PRI at 1/0:23 (index 0) Apr 7 22:55:15.772: %ISDN-6-LAYER2UP: Layer 2 for Interface Se1/0:23, TEI 0 cha nged to up Apr 7 22:55:15.772: ISDN Se1/0:23 Q921: User TX -> INFO sapi=0 tei=0, ns=0 nr=0 Apr 7 22:55:15.772: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:16.772: ISDN Se1/0:23 Q921: S7_T200_EXPIRY: VA = 0, VS = 1 Apr 7 22:55:16.772: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=0 nr= 0 Apr 7 22:55:16.772: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:17.772: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=0 nr= 0 Apr 7 22:55:17.772: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:17.784: ISDN Se1/0:23 Q921: User RX <- RRf sapi=0 tei=0 nr=1 Apr 7 22:55:45.773: ISDN Se1/0:23 Q921: User TX -> INFO sapi=0 tei=0, ns=1 nr=0 Apr 7 22:55:45.773: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:46.773: ISDN Se1/0:23 Q921: S7_T200_EXPIRY: VA = 1, VS = 2 Apr 7 22:55:46.773: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=1 nr= 0 Apr 7 22:55:46.773: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 Apr 7 22:55:47.773: ISDN Se1/0:23 Q921: User TX -> INFOp sapi=0 tei=0, ns=1 nr= 0 Apr 7 22:55:47.773: ISDN Se1/0:23 Q931: RESTART pd = 8 callref = 0x0000 Restart Indicator i = 0x87 ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Re: [Freeswitch-users] Two or more simultaneous calls not working
by Traun Leyden 08 Apr '09

08 Apr '09

> > > Hello, > > I wrote an application using FreeSWITCH version 1.0.3, with mod_python and > a > 64 bit box on Red Hat. > The app works fine when one person dials in, but when a second person dials > in, the first call stops and waits until the second call is finished. It's > really strange - if the first call is right in the middle of playing a > prompt, it will just stop, and there will be dead air. As soon as the 2nd > call hangs up, the prompt for the first call starts playing right where it > left off. > > I previously had FreeSWITCH installed on a 32 bit CentOS box, and this was > not happening. > > Does anybody have any idea what the cause of this could be? > > Thanks! I'm running mod_python fine with 64-bit (Ubuntu 8) and fs svn 12793, and have not seen that problem. Which version of python? Did you build it yourself or was it from a package? If its from a package, please provide the version of Red Hat you are using. One possible cause is that python was not compiled with multi-threading support. I don't know how to check that however .. googled around and didn't find anything. If you fire up your python interpreter and type "import threading" do you get an error?

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Two or more simultaneous calls not working
by Adam Wilt 08 Apr '09

08 Apr '09

Hello, I wrote an application using FreeSWITCH version 1.0.3, with mod_python and a 64 bit box on Red Hat. The app works fine when one person dials in, but when a second person dials in, the first call stops and waits until the second call is finished. It's really strange - if the first call is right in the middle of playing a prompt, it will just stop, and there will be dead air. As soon as the 2nd call hangs up, the prompt for the first call starts playing right where it left off. I previously had FreeSWITCH installed on a 32 bit CentOS box, and this was not happening. Does anybody have any idea what the cause of this could be? Thanks!

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Re: [Freeswitch-users] problems with Faktortel (AU) and multiple D ID's and extensions
by jim＠evolutiontel.net 08 Apr '09

08 Apr '09

Hi David, Have seen a similar issue reported on whirlpool recently with another provider, essentially if the ITSP does not forward the To: header with the correct terminating DID you will not be able to determine the extention to route the call to. Am I correct in saying you only have one Faktortel account with 2 DIDs attached? Regards, Jim - original message - Subject: [Freeswitch-users] problems with Faktortel (AU) and multiple DID's and extensions From: David Robinson <pawzlion(a)gmail.com> Date: 07/04/2009 13:50 I have been trying to setup 2 DID's to route to 2 extensions but whenever I try it, the second configured DID always routes to the first extension. In my public.xml I have the following: <include> <context name="public"> <extension name="DID 1"> <condition field="destination_number" expression="^(0746029000)$"> <action application="transfer" data="1000 XML default"/> </condition> </extension> <extension name="DID 2"> <condition field="destination_number" expression="^(0746029002)$"> <action application="transfer" data="1001 XML default"/> </condition> </extension> .... rest of file continues here ... While in my default.xml I have this: <include> <context name="default"> <extension name="David"> <condition field="destination_number" expression="^(0746029000)$" continue="on-true"> <action application="set" data="dialed_ext=$1" /> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/1000@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> <extension name="Jake"> <condition field="destination_number" expression="^(0746029001)$" continue="on-true"> <action application="set" data="dialed_ext=$2" /> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/1001@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> <extension name="outgoing - voicepulse"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=0746029000"/> <action application="bridge" data="sofia/gateway/voicepulse/$1"/> </condition> </extension> .. file continues here ... I got my new friend swk to try and diagnose the problem and using ngrep he found with ngrep that the incoming call to the second extension looked like this: U 203.161.130.133:5060 -> 10.0.0.12:5080 INVITE sip:gw+voicepulse@10.0.0.12:5080;transport=udp SIP/2.0..Via: SIP/2.0/UDP 203.161.130.133:5060;branch=z9hG4bK75f53071;rport..From: "0451282630" <sip: 0451282630(a)203.161.130.133>;tag=as555c5b50..To: <sip:gw+voicepulse@124.254.94.41:5080;transport=udp>..Contact: <sip:0451282630@203.161.130.133>..Call-ID: 7 47befb63a2def723e6796294853cc22@203.161.130.133..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 07 Apr 2009 08:18:23 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Content-Type: application/sdp..Content-Length: 290....v=0..o=root 1244 12 44 IN IP4 203.161.130.133..s=session..c=IN IP4 203.161.130.133..t=0 0..m=audio 13806 RTP/AVP 18 3 101..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap :3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv.. He says that the INVITE line should have a DNIS (not sure what that is) in that field to indicate which number to route it to but that for some reason, my provider (Faktortel in Australia) is not supplying that information. Does anyone know whether the problem is really at my provider's end or at my end, and if it's at my end, where ? thanks, pawz _______________________________________________ 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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Hi Load, but calls still perfect
by Nik Middleton 07 Apr '09

07 Apr '09

Hi Guys, I'm no Linux guru, but today I inadvertently had 1000+ call attempts going through FS, load according to TOP was 16.5. Calls were still absolutely perfect. Can I throw out the rule book on load ? CPU was ~45% on each core. (dual) Regards,

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Skype interaction commands on skypiax
by Giovanni Maruzzelli 07 Apr '09

07 Apr '09

Hi all, background: mod_skypiax is Skype compatible endpoint that allows incoming and outbound calls to/from the Skype network and SkypeOut service. It's seen by FS like other endpoints, so you can originate from sofia, bridge to skypiax, and connect the call to a landline number via SkypeOut service, for eg. skypiax endpoint use a Skype client to interact with the Skype network (see the wiki page for more details http://wiki.freeswitch.org/wiki/Skypiax) The news are: now you can send commands to the skype client related to a skyiax interface, both from the FS command line and programmatically (socket/API/esl/whatever) http://wiki.freeswitch.org/wiki/Skypiax#API_and_CLI_Commands This allow you to use directly the entire power of the Skype API ( https://developer.skype.com/Docs/ApiDoc ), for eg to send chat messages, interact with the buddy list, etc etc. Typing "console loglevel 9" at the FS command line allows you to see the Skype API answers from the Skype client instance. So, in short: you bring loglevel to 9 (so you can see the Skype API messages going back and forth), you use "sk" or "skypiax" to send Skype API commands to the Skype client instance. This way you can prototype extensions to the current mod_skypiax, that can then be implemented in C directly into the mod_skypiax source code. Please, let me know of extensions you would like to be integrated into the mod_skypiax code ;-). Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039

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Problem listening to PCMU and ul recordings
by Drew Ozier 07 Apr '09

07 Apr '09

Hi, I've been using the record_session feature and wish to use PCMU or ul audio format, but when I try to play back the audio in either format, it sounds high-pitch and fast as if it is playing back at 2x speed. I looked at the waveform recorded in PCMU and ul versus what it looks like when I record as wav, and it seems like it is only recording every-other sample (which would explain the pitch and speed). My vars.xml is set to PCMU as the global codec pref and the outbound codec pref. I am recording in stereo (one channel per leg of the call), but I'm not messing with any other recording parameters. Incedentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio file, because I'm getting calls off a T1 (actually off of an AudioCodes Mediant 1000 that is converting the T1 to SIP for me), and I'd like to record precicely what is coming off the wire. I'd be happy to send any configuration files, I'm just currently at a loss for how to proceed. Thanks, Drew Ozier

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Problem listening to PCMU and ul recordings
by Drew Ozier 07 Apr '09

07 Apr '09

Hi, I've been using the record_session feature and wish to use PCMU or ul audio format, but when I try to play back the audio in either format, it sounds high-pitch and fast as if it is playing back at 2x speed. I looked at the waveform recorded in PCMU and ul versus what it looks like when I record as wav, and it seems like it is only recording every-other sample (which would explain the pitch and speed). My vars.xml is set to PCMU as the global codec pref and the outbound codec pref. I am recording in stereo (one channel per leg of the call), but I'm not messing with any other recording parameters. Incidentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio file, because I'm getting calls off a T1 (actually off of an AudioCodes Mediant 1000 that is converting the T1 to SIP for me), and I'd like to record precisely what is coming off the wire. I'd be happy to send any configuration files, I'm just currently at a loss for how to proceed. Thanks, Drew Ozier

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