[Freeswitch-users] Number of codecs offerred in SDP
David Ponzone
david.ponzone at gmail.com
Wed Jun 30 06:25:28 PDT 2010
This looks like you have disable-transcoding set to true.
David Ponzone Direction Technique
email: david.ponzone at ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis
à l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion non autorisée est interdite. Tout message électronique est
susceptible d'altération. IPeva décline toute responsabilité au titre
de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes
pas destinataire de ce message, merci de le détruire immédiatement et
d'avertir l'expéditeur.
Le 30/06/2010 à 15:07, Mark Campbell-Smith a écrit :
> Updating the configuration did not help.
>
> I'm not telling FS to use any other codecs other than what I have
> specified below, so I'm not sure what I have done wrong. Below is the
> full trace:
>
>
> ------------------------------------------------------------------------
> INVITE sip:1020 at 192.168.1.120 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae
> From: 1000 <sip:1000 at 192.168.1.120>;tag=2c7a518d12f9370eo0
> To: <sip:1020 at 192.168.1.120>
> Call-ID: 316156b9-f6b413d2 at 192.168.1.121
> CSeq: 102 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="1000",realm="192.168.1.120",nonce="77836a7a-8447-11df-93a6-
> d9ad5b204ca2",uri="sip:
> 1020
> @192.168.1.120
> ",algorithm
> =
> MD5
> ,response
> =
> "c4a5e5cb60676366b8bcfb1329f3fc08
> ",qop=auth,nc=00000001,cnonce="6a0220d0"
> Contact: 1000 <sip:1000 at 192.168.1.121:5060>
> Expires: 240
> User-Agent: Linksys/PAP2T-5.1.6(LS)
> Content-Length: 451
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura, replaces
> Content-Type: application/sdp
>
> v=0
> o=- 17789112 17789112 IN IP4 192.168.1.121
> s=-
> c=IN IP4 192.168.1.121
> t=0 0
> m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:96 G726-40/8000
> a=rtpmap:97 G726-24/8000
> a=rtpmap:98 G726-16/8000
> a=rtpmap:100 NSE/8000
> a=fmtp:100 192-193
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
>
> ------------------------------------------------------------------------
> send 343 bytes to udp/[192.168.1.121]:5060 at 13:00:32.882684:
>
> ------------------------------------------------------------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae
> From: 1000 <sip:1000 at 192.168.1.120>;tag=2c7a518d12f9370eo0
> To: <sip:1020 at 192.168.1.120>
> Call-ID: 316156b9-f6b413d2 at 192.168.1.121
> CSeq: 102 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19
> 14-49-15 -0500
> Content-Length: 0
>
>
> ------------------------------------------------------------------------
> 2010-06-30 23:00:32.899423 [DEBUG] sofia.c:5975 IP 192.168.1.121
> Rejected by acl "domains". Falling back to Digest auth.
> 2010-06-30 23:00:32.914917 [NOTICE] switch_channel.c:776 New Channel
> sofia/internal/1000 at 192.168.1.120
> [77a3ab82-8447-11df-93a7-d9ad5b204ca2]
> 2010-06-30 23:00:32.943135 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 192.168.1.120) Running State Change CS_NEW
> 2010-06-30 23:00:32.945232 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/1000 at 192.168.1.120) State NEW
> 2010-06-30 23:00:32.989778 [DEBUG] sofia.c:4293 Channel
> sofia/internal/1000 at 192.168.1.120 entering state [received][100]
> 2010-06-30 23:00:32.997500 [DEBUG] sofia.c:4304 Remote SDP:
> v=0
> o=- 17789112 17789112 IN IP4 192.168.1.121
> s=-
> c=IN IP4 192.168.1.121
> t=0 0
> m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:96 G726-40/8000
> a=rtpmap:97 G726-24/8000
> a=rtpmap:98 G726-16/8000
> a=rtpmap:100 NSE/8000
> a=fmtp:100 192-193
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
>
> 2010-06-30 23:00:33.002369 [DEBUG] sofia_glue.c:3877 Audio Codec
> Compare [PCMU:0:8000:30]/[G729:18:8000:20]
> 2010-06-30 23:00:33.005967 [DEBUG] sofia_glue.c:3877 Audio Codec
> Compare [PCMU:0:8000:30]/[PCMU:0:8000:20]
> 2010-06-30 23:00:33.008248 [DEBUG] sofia_glue.c:3877 Audio Codec
> Compare [PCMU:0:8000:30]/[GSM:3:8000:20]
> 2010-06-30 23:00:33.010721 [DEBUG] sofia_glue.c:3924 Substituting
> codec PCMU at 30i@8000h
> 2010-06-30 23:00:33.019002 [DEBUG] sofia_glue.c:2462 Set Codec
> sofia/internal/1000 at 192.168.1.120 PCMU/8000 30 ms 240 samples
> 2010-06-30 23:00:33.038714 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf
> send/recv payload to 101
> 2010-06-30 23:00:33.040992 [DEBUG] sofia.c:4451
> (sofia/internal/1000 at 192.168.1.120) State Change CS_NEW -> CS_INIT
> 2010-06-30 23:00:33.043182 [DEBUG] switch_core_session.c:1027 Send
> signal sofia/internal/1000 at 192.168.1.120 [BREAK]
> 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 192.168.1.120) Running State Change CS_INIT
> 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/1000 at 192.168.1.120) State INIT
> 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:83
> sofia/internal/1000 at 192.168.1.120 SOFIA INIT
> 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:117
> (sofia/internal/1000 at 192.168.1.120) State Change CS_INIT -> CS_ROUTING
> 2010-06-30 23:00:33.048752 [DEBUG] switch_core_session.c:1027 Send
> signal sofia/internal/1000 at 192.168.1.120 [BREAK]
> 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/1000 at 192.168.1.120) State INIT going to sleep
> 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 192.168.1.120) Running State Change CS_ROUTING
> 2010-06-30 23:00:33.048752 [DEBUG] switch_channel.c:1474
> (sofia/internal/1000 at 192.168.1.120) Callstate Change DOWN -> RINGING
> 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/1000 at 192.168.1.120) State ROUTING
> 2010-06-30 23:00:33.055456 [DEBUG] switch_channel.c:1333
> (sofia/internal/1000 at 192.168.1.120) Callstate Change RINGING -> ACTIVE
> 2010-06-30 23:00:33.055456 [DEBUG] mod_sofia.c:140
> sofia/internal/1000 at 192.168.1.120 SOFIA ROUTING
> 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:77
> sofia/internal/1000 at 192.168.1.120 Standard ROUTING
> 2010-06-30 23:00:33.055456 [INFO] mod_dialplan_xml.c:331 Processing
> 1000->1020 in context default
> Dialplan: sofia/internal/1000 at 192.168.1.120 parsing
> [default->Local_1000_1019] continue=false
> Dialplan: sofia/internal/1000 at 192.168.1.120 Regex (FAIL)
> [Local_1000_1019] destination_number(1020) =~ /^(10[01][0-9])$/
> break=on-false
> Dialplan: sofia/internal/1000 at 192.168.1.120 parsing
> [default->Mobile_1020s] continue=false
> Dialplan: sofia/internal/1000 at 192.168.1.120 Regex (PASS)
> [Mobile_1020s] destination_number(1020) =~ /^(102[0-9])$/
> break=on-false
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action
> set(dialed_extension=1020)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action
> export(codec_string=GSM)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action
> set(codec_string=GSM)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action
> bridge(user/${dialed_extension}@${domain})
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action set_user(1000@$
> {domain})
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action answer()
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action sleep(1000)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action
> system(/usr/local/freeswitch/scripts/sms.pl ${smsaccount}
> ${smspassword} ${smsnumber} 'You have one new voicemail from ${effec$
> <action application=)
> 2010-06-30 23:00:33.110926 [DEBUG] switch_core_state_machine.c:119
> (sofia/internal/1000 at 192.168.1.120) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-06-30 23:00:33.114917 [DEBUG] switch_core_session.c:1027 Send
> signal sofia/internal/1000 at 192.168.1.120 [BREAK]
> 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:341
> (sofia/internal/1000 at 192.168.1.120) State ROUTING going to sleep
> 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1000 at 192.168.1.120) Running State Change CS_EXECUTE
> 2010-06-30 23:00:33.128360 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/1000 at 192.168.1.120) State EXECUTE
> 2010-06-30 23:00:33.129661 [DEBUG] mod_sofia.c:233
> sofia/internal/1000 at 192.168.1.120 SOFIA EXECUTE
> 2010-06-30 23:00:33.134566 [DEBUG] switch_core_state_machine.c:157
> sofia/internal/1000 at 192.168.1.120 Standard EXECUTE
> EXECUTE sofia/internal/1000 at 192.168.1.120 set(dialed_extension=1020)
> 2010-06-30 23:00:33.157698 [DEBUG] mod_dptools.c:843
> sofia/internal/1000 at 192.168.1.120 SET [dialed_extension]=[1020]
> EXECUTE sofia/internal/1000 at 192.168.1.120 bridge(user/1020 at mydns.dyndns.org
> )
> 2010-06-30 23:00:33.248458 [DEBUG] switch_ivr_originate.c:1956
> variable string 0 = [presence_id=1020 at mydns.dyndns.org]
> 2010-06-30 23:00:33.260573 [NOTICE] switch_channel.c:776 New Channel
> sofia/internal/sip:1020 at 192.168.1.123:5060
> [77d7e3b6-8447-11df-93a8-d9ad5b204ca2]
> 2010-06-30 23:00:33.303487 [DEBUG] mod_sofia.c:3883
> (sofia/internal/sip:1020 at 192.168.1.123:5060) State Change CS_NEW ->
> CS_INIT
> 2010-06-30 23:00:33.304564 [DEBUG] switch_core_session.c:1027 Send
> signal sofia/internal/sip:1020 at 192.168.1.123:5060 [BREAK]
> 2010-06-30 23:00:33.327507 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:1020 at 192.168.1.123:5060) Running State Change
> CS_INIT
> 2010-06-30 23:00:33.332010 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/sip:1020 at 192.168.1.123:5060) State INIT
> 2010-06-30 23:00:33.334621 [DEBUG] mod_sofia.c:83
> sofia/internal/sip:1020 at 192.168.1.123:5060 SOFIA INIT
> send 1183 bytes to udp/[192.168.1.123]:5060 at 13:00:33.351944:
>
> ------------------------------------------------------------------------
> INVITE sip:1020 at 192.168.1.123:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKFDpKe5v6j2QcQ
> Max-Forwards: 69
> From: "1000" <sip:1000 at 192.168.1.120>;tag=Z51Qve55SUHta
> To: <sip:1020 at 192.168.1.123:5060>
> Call-ID: 4f442091-feea-122d-448a-00e04c0312e9
> CSeq: 132833080 INVITE
> Contact: <sip:mod_sofia at 192.168.1.120:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19
> 14-49-15 -0500
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla, include-session-description, presence.winfo, message-summary,
> refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 246
> X-FS-Support: update_display
> Remote-Party-ID: "1000"
> <sip:1000 at 192.168.1.120>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1277875447 1277875448 IN IP4 192.168.1.120
> s=FreeSWITCH
> c=IN IP4 192.168.1.120
> t=0 0
> m=audio 27386 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:30
>
> It fails after this with Not Acceptable Here /
> INCOMPATIBLE_DESTINATION
>
> On Mon, Jun 28, 2010 at 5:15 PM, David Ponzone <david.ponzone at gmail.com
> > wrote:
>> Please, retry with a genuine config (the default one would be a wise
>> choice).
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: 01 74 03 18 97
>> gsm: 06 66 98 76 34
>> Service Client IPeva
>> tel: 0811 46 26 26
>> www.ipeva.fr - www.ipeva-studio.com
>> Ce message et toutes les pièces jointes sont confidentiels et
>> établis à
>> l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion
>> non autorisée est interdite. Tout message électronique est
>> susceptible
>> d'altération. IPeva décline toute responsabilité au titre de ce
>> message s'il
>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire
>> de ce
>> message, merci de le détruire immédiatement et d'avertir
>> l'expéditeur.
>>
>>
>>
>> Le 28/06/2010 à 07:37, Mark Campbell-Smith a écrit :
>>
>> Hi All,
>>
>> I'm not really sure if I got a firm answer for this one. Is the only
>> way to ensure that transcoding is performed is by using
>> late-negotiation? Why aren't all my codecs sent in the INVITE
>> message
>> to the B-leg (extension 1020)?
>>
>> Thanks!
>>
>> On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith
>> <mcampbellsmith at gmail.com> wrote:
>>
>> FS version and codecs are shown below, but my config file are
>> probably
>>
>> quite old. But I guess they should still work?
>>
>> All codecs are loaded, and the call works if late negotiation is set
>>
>> on profile internal.
>>
>> As I wrote above:
>>
>> The call setup is extension 1000 calls extension 1020
>>
>> 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
>>
>> FS as the A-leg codec
>>
>> 2. Extension 1020 only supports GSM codec. The call fails with Not
>>
>> Acceptable Here.
>>
>> I forgot to write that Extension 1000 does not support GSM (I want to
>>
>> force transcoding). Is that why FS is filtering out GSM on the b-
>> leg?
>>
>> freeswitch at internal> version
>>
>> FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500)
>>
>> freeswitch at internal> show codecs
>>
>> type,name,ikey
>>
>> codec,ADPCM (IMA),mod_voipcodecs
>>
>> codec,G.711 alaw,CORE_PCM_MODULE
>>
>> codec,G.711 ulaw,CORE_PCM_MODULE
>>
>> codec,G.722,mod_voipcodecs
>>
>> codec,G.723.1 6.3k,mod_g723_1
>>
>> codec,G.726 16k,mod_voipcodecs
>>
>> codec,G.726 16k (AAL2),mod_voipcodecs
>>
>> codec,G.726 24k,mod_voipcodecs
>>
>> codec,G.726 24k (AAL2),mod_voipcodecs
>>
>> codec,G.726 32k,mod_voipcodecs
>>
>> codec,G.726 32k (AAL2),mod_voipcodecs
>>
>> codec,G.726 40k,mod_voipcodecs
>>
>> codec,G.726 40k (AAL2),mod_voipcodecs
>>
>> codec,G.729,mod_com_g729
>>
>> codec,GSM,mod_voipcodecs
>>
>> codec,H.261 Video (passthru),mod_h26x
>>
>> codec,H.263 Video (passthru),mod_h26x
>>
>> codec,H.263+ Video (passthru),mod_h26x
>>
>> codec,H.263++ Video (passthru),mod_h26x
>>
>> codec,H.264 Video (passthru),mod_h26x
>>
>> codec,LPC-10,mod_voipcodecs
>>
>> codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
>>
>> codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
>>
>> codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
>>
>> codec,Speex,mod_speex
>>
>> 25 total.
>>
>>
>> On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone <david.ponzone at gmail.com
>> >
>> wrote:
>>
>> Mark,
>>
>> I confirm that, as I wrote that wiki page (the early negotiation
>> part) :)
>>
>> Can you really confirm your FS version ?
>>
>> The parameter you showed is old.
>>
>> codec-prefs has been replaced in SIP profiles by:
>>
>> <param name="inbound-codec-prefs" value="$$
>> {global_codec_prefs}"/>
>>
>> <param name="outbound-codec-prefs" value="$$
>> {outbound_codec_prefs}"/>
>>
>> David Ponzone Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel: 01 74 03 18 97
>>
>> gsm: 06 66 98 76 34
>>
>> Service Client IPeva
>>
>> tel: 0811 46 26 26
>>
>> www.ipeva.fr - www.ipeva-studio.com
>>
>> Ce message et toutes les pièces jointes sont confidentiels et
>> établis à
>>
>> l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion
>>
>> non autorisée est interdite. Tout message électronique est
>> susceptible
>>
>> d'altération. IPeva décline toute responsabilité au titre de ce
>> message s'il
>>
>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire
>> de ce
>>
>> message, merci de le détruire immédiatement et d'avertir
>> l'expéditeur.
>>
>>
>>
>> Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :
>>
>> Check this good wiki page for how FS negotiates codecs (early
>>
>> negotiation default):
>>
>> http://wiki.freeswitch.org/wiki/Codec_Negotiation
>>
>> I have this set in my internal profile:
>>
>> <param name="codec-prefs" value="$${global_codec_prefs}"/>
>>
>> and as stated before vars.xml:
>>
>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
>>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
>>
>> Setting late negotiation works (thanks Sergey), but reading the wiki
>>
>> page, I see the following sentence, which I interpret that GSM should
>>
>> still be sent:
>>
>> When FS calls leg B, the list of codecs in outbound-codec-prefs for
>>
>> the SIP profile is reorganized by pushing the codec negotiated above
>>
>> for leg A at the top . If B does not accept any of the codecs, the
>>
>> calls fails, obviously.
>>
>>
>>
>> On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano
>>
>> <tgraziano at myitdepartment.net> wrote:
>>
>> I'm a newb to fs, but doesn't codec get neogtiated by the endpoints?
>>
>> Wouldn't fs only get involved when its media server is referred to?
>>
>> If the "other endpoint" will only accept G729, doesn't that mean you
>>
>> need to change that endpoint to also accept G711 or also license G729
>>
>> in FS?
>>
>> On 6/23/10, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>>
>> Test Setup:
>>
>> vars.xml:
>>
>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
>>
>> <X-PRE-PROCESS cmd="set"
>> data="outbound_codec_prefs=G729,PCMU,GSM"/>
>>
>> The call setup is extension 1000 calls extension 1020
>>
>> 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
>>
>> FS as the A-leg codec
>>
>> 2. Extension 1020 only supports GSM codec. The call fails with Not
>>
>> Acceptable Here.
>>
>> FS only offers G729 and PCMU to 1020. How do I change the number of
>>
>> codecs that are offered to an extension? I know I can change the
>>
>> order in the codec_prefs, but would prefer FS to offer all three
>>
>> codecs to an extension.
>>
>> m=audio 23662 RTP/AVP 0 18 101 13
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:18 G729/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=rtpmap:13 CN/8000
>>
>> a=ptime:30
>>
>> Thanks
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>> --
>>
>> Sent from my mobile device
>>
>> ======================
>>
>> Tony Graziano, Manager
>>
>> Telephone: 434.984.8430
>>
>> sip: tgraziano at voice.myitdepartment.net
>>
>> Fax: 434.984.8431
>>
>> Email: tgraziano at myitdepartment.net
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>>
>> Telephone: 434.984.8426
>>
>> sip: helpdesk at voice.myitdepartment.net
>>
>> Fax: 434.984.8427
>>
>> Helpdesk Contract Customers:
>>
>> http://www.myitdepartment.net/gethelp/
>>
>> Why do mathematicians always confuse Halloween and Christmas?
>>
>> Because 31 Oct = 25 Dec.
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
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>>
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>>
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
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>>
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>>
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>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>>
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>>
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>>
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>>
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>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
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>>
>
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