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Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem
by Michel Habib 10 Mar '11

10 Mar '11

Yes, I get the Audio from FS in regular calls - I already disabled all possible firewalls - all 3 machines [softphone, freeswitch, Speech Server (and mrcp connector) ] are on a switch. 192.168.5.107 is the freeswitch server 192.168.5.110 is the MRCP-Connector/Speech Server 2007 server I made too many iterations on the configuration below: <include> <profile name="mrcp-connector" version="2"> <param name="client-ip" value="192.168.5.107"/> <param name="client-port" value="5090"/> <param name="server-ip" value="192.168.5.110"/> <param name="server-port" value="5070"/>  <param name="sip-transport" value="udp"/>   <param name="rtp-ext-ip" value="192.168.5.107"/> <param name="rtp-ip" value="192.168.5.107"/> <param name="rtp-port-min" value="4000"/> <param name="rtp-port-max" value="5000"/>  <param name="rtcp" value="1"/>  <param name="rtcp-bye" value="2"/>  <param name="rtcp-tx-interval" value="5000"/>  <param name="rtcp-rx-resolution" value="1000"/>    <param name="codecs" value="PCMU PCMA L16/96/8000"/>  <synthparams> </synthparams>  <recogparams>  </recogparams> </profile> </include> ---------- Forwarded message ---------- > From: Christopher Rienzo <cmrienzo(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 7 Mar 2011 09:32:51 -0500 > Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server > [using MRCP Connector] - Audio Problem > Do you get audio between FS and your SIP client when not using ASR/TTS? > > Show me the MRCP profile configuration and your FreeSWITCH logs during the > call. > > > > On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib(a)gmail.com>wrote: > >> Hello All, >> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP >> calls and use its ASR and TTS Services successfully] >> I am also using MRCP Connector from AumTech - which allows me to use ASR >> and TTS Services through an MRCP Client . >> Now, i am using Freeswitch mod unimrcp to use ASR and TTS. >> >> for TTS, I can successfully make the call, the Audio RTP of the TTS voice >> is transferred succesfully from Speech Server [through MRCP Connector] back >> to the Freeswitch Server. >> However, Freeswitch is not sending back the Audio RTP to the SIP client. >> >> for ASR, I can successfully define the grammar and start recognition, but >> the audio RTP sent to speech server [through MRCP Connector] is silent >> [empty]. >> >> I am suspecting something is wrong with the RTP Configuration - can you >> help me? >> >> Let me now if you need any specific logs/scripts/configuration? >> >> Thank you, >> Michel. >> >> >>

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incompatible destination
by Sam 09 Mar '11

09 Mar '11

Hello, The extension 7006 was working few days ago and now it not working,rest all the phone are working properly. U 192.168.2.190:5060 -> 192.168.2.14:5060 INVITE sip:7006@192.168.2.14:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..Max-Forwards: 70..From: "" <sip:9322273640@192.168.2.190>;tag=4QggZ1XcNZD3F..To: < sip:7006@192.168.2.14:5060>..Call-ID: 7bf ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: < sip:mod_sofia@192.168.2.190:5060>..User-Agent: NOVANET..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-Dis position: session..Content-Length: 203..X-FS-Support: update_display..Remote-Party-ID: <sip:9322273640@192.168.2.190>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1299446773 1299446774 IN IP4 192.168.2.190..s=FreeSWITCH ..c=IN IP4 192.168.2.190..t=0 0..m=audio 31474 RTP/AVP 8 101 13..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20.. U 192.168.2.14:5060 -> 192.168.2.190:5060 SIP/2.0 488 Not Acceptable Here..Via: SIP/2.0/UDP 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..From: < sip:9322273640@192.168.2.190>;tag=4QggZ1XcNZD3F..To: "7006" < sip:7006@192.168.2.14:5060>;tag=983B5A59-EB953B22..CSeq: 9403059 INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: < sip:7006@192.168.2.14:5060>..User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.2.0477..Accept-Language: en..Warning: 488 SDP "NotAcceptableHere"..Content-Length: 0.... I tried all the possibilities with codec priorities on U/A . http://pastebin.freeswitch.org/15577 Any help. Regards Sam

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FreeSWITCH Conference Call Followup - Recordings Online
by Michael Collins 09 Mar '11

09 Mar '11

FYI, We had a nice conference call with Jim Gettys and Dave Täht from the Bufferbloat project. I've made the recordings available in their usual spot: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls Use the links next to "FS_weekly_2011_03_09" to download the audio format of your choice. Thanks to Jim, Dave, and all those who participated today! -MC

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To Proxy Media or Not
by Max Clark 09 Mar '11

09 Mar '11

Hello, I have a couple of FreeSWITCH boxes functioning as SBCs on our network. We need them to: - Route Calls - Function as B2BUAs for call logging - Hide topology from the endpoints They do not need to: - Intercept/play audio - Manipulate the SDP - Transcode - Process anything other than SIP So these machines have been running with Proxy Media enabled in the dialplan. Is the the proper/recommended way of configuring for our environment/needs? Is the minimal gain of resources by not processing the RTP worth it? Thanks, Max

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Duplicate UUIDs in voicemail_msgs
by Troy Anderson 09 Mar '11

09 Mar '11

Hi All, I am regularly getting multiple copies of voicemails when checking voicemail on one of the extensions of a system, and, trying to diagnose the situation, seeing duplicate UUIDs in voicemail_msgs database. I suspect that's why I get multiple "copies" of the message. I'm trying to figure out what would cause multiple entries in the database for the same message. I checked the version of mod_voicemail.c I'm using on this machine (from Nov 18, 2010) against the latest version and none of the few changes seem remotely related. I would love to update to the latest and see if the issue is resolved, but this is on a client's production system, so that's not possible ATM. Any ideas on what may be causing this, or how to diagnose further? Thanks! Troy

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sofia.c:5172 Codec Error! v=0
by Max Clark 09 Mar '11

09 Mar '11

How do I identify the source of this error and fix it? 2011-03-09 16:02:42.845215 [ERR] sofia.c:5172 Codec Error! v=0 o=MxSIP 0 28211 IN IP4 x.x.x.x s=SIP Call c=IN IP4 x.x.x.x t=0 0 m=audio 5238 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 I'm actually surprised that I'm even seeing this with proxy_media=true enabled. The codec list is correct and what are supported by both endpoints. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/> Thanks, Max

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Pre-Answer / Sending early media Command not found
by Max Clark 09 Mar '11

09 Mar '11

I upgraded to current last night, since then I've noticed these errors on the console: -ERR 2011-03-09 15:49:15.629671 [NOTICE] sofia.c:4794 Pre-Answer sofia/external/xxxx@y.y.y.y:5060! Command not found! freeswitch@internal> 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365 Sending early media -ERR 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365 Sending early media Command not found! Do I need to add/adjust configuration on my side? Thanks, Max

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FreeSWITCH Conference Call Today: Special Presentation
by Michael Collins 09 Mar '11

09 Mar '11

Hello all! Today we are having a special presentation by Jim Gettys and Dave Taht of the Bufferbloat project. Please join us and learn more about this subject - you'll be glad you did. The official agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_09 Talk to you soon! -Michael

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Re: [Freeswitch-users] tls/srtp goes straight to voicemail
by Mitch Johnson 09 Mar '11

09 Mar '11

It turns out that the Eyebeam app was the cause of the issue. Using the Acrobits client showed no issues whatsoever. This was discovered with a lot of help from another Mitch. > From: Michael Collins <msc(a)freeswitch.org> > Date: March 8, 2011 4:49:26 PM EST > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Subject: Re: [Freeswitch-users] tls/srtp goes straight to voicemail > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Are you sure that the Eyebeam clients support SRTP? It looks like you're getting a 406: > > 2011-03-05 23:21:08.650486 [NOTICE] sofia.c:5323 Hangup sofia/internal/sip:1005@192.168.60.135:53664 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] > > You probably need to look directly at the SIP traces and see if there is any more information as to what's going on. > > -MC > > > >

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send and receive call for non registered endpoint.
by Gary Chen 09 Mar '11

09 Mar '11

I have a IAD with PRI interface and multiple SIP users. I do not want to register each user on Freeswitch. I do still want each user to be able to send call and receive call. For sending call from IAD to FS, since all the sip calls have to go through SIP-router (Kamailio) and then routed to FS, I am thinking using X-AUTH-IP in ACL so no auth required. For receiveing call from FS, I am thinking either to setup each user on FS and manually create contact information and make them never expire or add cidr to each user and modify dial-string to use cidr instead of sip contact. Are these approaches workable? What is the best way to do this? Gary

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