[asterisk-users] Transcode Speex to G711-ulaw

2006-09-08 Thread Kokfoo Soo 
Hi All, I try to transcode Speex to G711 through Asterisk, however, the voice quality is crappy. Does anyone has a clue to fix this? The current version i running now is 1.2.4, and the Speex codec was sent from eyeBeam.The following warning below:Sep 7 13:48:55 WARNING[5297]:  codec_speex.c:278 speextolin_framein: Out of buffer  space Sep 7 13:48:55  WARNING[5297]:  codec_speex.c:278 speextolin_framein: Out of buffer  space  The configuration is  below.[EMAIL PROTECTED] ~]# cat  /etc/asterisk/codecs.conf [speex] quality = 3 complexity  = 2 enhancement = false vad =  false vbr =  false abr = 0 vbr_quality = 4 dtx =  false 
	

	
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RE: [asterisk-users] netmask

2006-09-07 Thread Kokfoo Soo 
Can we apply netmask on SIP Context instead of individual IP address?Thanks,Dean Collins [EMAIL PROTECTED] wrote: Ok, cool, thought it was probably always that, just having problem withfaktortel at the moment so must be another problem. Cheers,Dean  -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Richard Klingler Sent: Thursday, 7 September 2006 7:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] netmask  Hi Dean  Dean Collins schrieb:  I don't know if I'm mistaken or not but I noticed in a iax2 showpeers  command that it is showing my iax2
 connections as netmask255.255.255.255  /32 are hosts addresses...which is correct.   All of my lan traffic is supposed to be running on 255.255.255.0  This doesn't mean that all hosts on the internet need the same subnet as you (o;  How would you or asterisk know what netmask is used on a remote host not on the local subnet?   chers rick   ___ --Bandwidth and Colocation provided by Easynews.com --  asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  
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[asterisk-users] Capacity for transcode G711 to G729

2006-09-07 Thread Kokfoo Soo 
Does anyone know how many active channels can support for transcoding ulaw to G729 by using 4x 3.6GHz Xeon Processors?Thanks, 
	

	
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[asterisk-users] Speex Codex - Eyebean to Asterisk

2006-09-07 Thread Kokfoo Soo 
Hi Guys,I try to use Eyebean "Speex" Codec to Asterisk and transcoded to G729 outbound to Cisco. I receive very clear on Eyebeam, but transmit crappy to Cisco. Any clue?I get the following notices below:Sep 7 13:48:55 WARNING[5297]:  codec_speex.c:278 speextolin_framein: Out of buffer  space Sep 7 13:48:55
 WARNING[5297]:  codec_speex.c:278 speextolin_framein: Out of buffer  space  The configuration is  below.[EMAIL PROTECTED] ~]# cat  /etc/asterisk/codecs.conf [speex] quality = 3 complexity
 = 2 enhancement = false vad =  false vbr =  false abr = 0 vbr_quality = 4 dtx =  false 
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Re: [asterisk-users] asterisk t.38 fax failed

2006-09-06 Thread Kokfoo Soo 
Ricardo,Thanks, could you please share some of your t.38 passthrough configuration in sip.conf and also udptl.conf?Thanks,Ricardo Carvalho [EMAIL PROTECTED] wrote: No, T.38 doesn't work with Asterisk. Only works with Asterisk t38passthrough patch that you can find at URL: http://bugs.digium.com/file_download.php?file_id=9335type=bugFor me it only worked well with patch for version 1.2.4 of Asterisk.Regards,Ricardo.Kokfoo Soo wrote: Is T.38 fax work through Asterisk? I have the config below in my  sip.conf, but the fax doesn't work and give me the CLI lines below. My  current version is 1.2.10. Please help. [Inboundtopbx] type=friend context=pbx host=10.18.188.84 insecure=port
 dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 allow=ulaw t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no [OutboundfromPBX] type=peer host=10.18.161.222   canreinvite=no dtmfmode=rfc2833 disallow=all allow=g729 qualify=yes t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no -- SIP read from 10.18.188.84:50096: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP  10.18.188.84:5060 From: ;tag=19D429E8-2084 To: ;tag=as3c87a22e Date: Tue, 05 Sep 2006 19:42:28 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK --- (9 headers 0 lines)--- Sep  5 15:30:31 NOTICE[25233]: rtp.c:564
 ast_rtp_read: Unknown RTP  codec 100 received Sep  5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received Sep  5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received Sep  5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received Sep  5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown  SDP media type in offer: image 16406 udptl t38  Yahoo! Messenger with Voice. Make PC-to-Phone Callsto the US (and 30+ countries) for 2¢/min or less.  ___
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Re: [asterisk-users] asterisk t.38 fax failed

2006-09-06 Thread Kokfoo Soo 
Ricardo,I found compilation error below, any thought?chan_sip.c:3895: error: `UDPTL_ERROR_CORRECTION_REDUNDANCY' undeclared (first use in this function)chan_sip.c:3898: error: `UDPTL_ERROR_CORRECTION_FEC' undeclared (first use in this function)chan_sip.c:3901: error: `UDPTL_ERROR_CORRECTION_NONE' undeclared (first use in this function)chan_sip.c: In function `add_t38_sdp':chan_sip.c:4728: warning: implicit declaration of function `ast_udptl_get_us'chan_sip.c:4772: warning: implicit declaration of function `ast_udptl_get_local_max_datagram'chan_sip.c: In function `transmit_response_with_t38_sdp':chan_sip.c:5044: warning: implicit declaration of function `ast_udptl_offered_from_local'chan_sip.c: In function `handle_response':chan_sip.c:10516: warning: implicit declaration of function `ast_udptl_stop'chan_sip.c: In function `sip_set_udptl_peer':chan_sip.c:13488: warning: implicit declaration of function
 `ast_udptl_get_peer'chan_sip.c: At top level:chan_sip.c:13821: error: variable `sip_udptl' has initializer but incomplete typechan_sip.c:13822: error: unknown field `type' specified in initializerchan_sip.c:13822: warning: excess elements in struct initializerchan_sip.c:13822: warning: (near initialization for `sip_udptl')chan_sip.c:13823: error: unknown field `get_udptl_info' specified in initializerchan_sip.c:13823: warning: excess elements in struct initializerchan_sip.c:13823: warning: (near initialization for `sip_udptl')chan_sip.c:13824: error: unknown field `set_udptl_peer' specified in initializerchan_sip.c:13824: warning: excess elements in struct initializerchan_sip.c:13824: warning: (near initialization for `sip_udptl')chan_sip.c: In function `load_module':chan_sip.c:13986: warning: implicit declaration of function `ast_udptl_proto_register'chan_sip.c: In function `unload_module':chan_sip.c:14038:
 warning: implicit declaration of function `ast_udptl_proto_unregister'chan_sip.c: At top level:chan_sip.c:13821: error: storage size of `sip_udptl' isn't knownmake[1]: *** [chan_sip.o] Error 1make[1]: Leaving directory `/usr/src/asterisk-1.2.4/channels'make: *** [subdirs] Error 1Ricardo Carvalho [EMAIL PROTECTED] wrote: In sip.conf add to [general] context and to every peer context that you want to register in Asterisk to use T.38 the following lines:t38pt_udptl=yest38pt_rtp=not38pt_tcp=noIn udptl.conf file I have the following configurations:[general]udptlstart=4000udptlend=4999T38FaxUdpEC = t38UDPRedundancyT38FaxMaxDatagram = 400udptlfecentries = 3udptlfecspan = 3Good luck,Ricardo.Kokfoo Soo
 wrote: Ricardo, Thanks, could you please share some of your t.38 passthrough  configuration in sip.conf and also udptl.conf? Thanks, */Ricardo Carvalho <[EMAIL PROTECTED]>/* wrote: No, T.38 doesn't work with Asterisk. Only works with Asterisk t38passthrough patch that you can find at URL: http://bugs.digium.com/file_download.php?file_id=9335type=bug For me it only worked well with patch for version 1.2.4 of Asterisk. Regards, Ricardo. Kokfoo Soo wrote:  Is T.38 fax work through Asterisk? I have the config below in my  sip.conf, but the fax doesn't work and give me the CLI lines below. My  current version is 1.2.10. Please help.   [Inboundtopbx] 
 type=friend  context=pbx  host=10.18.188.84  insecure=port  dtmfmode=rfc2833  canreinvite=no  disallow=all  allow=g729  allow=ulaw  t38pt_udptl=yes  t38pt_rtp=no  t38pt_tcp=no   [OutboundfromPBX]  type=peer  host=10.18.161.222  canreinvite=no  dtmfmode=rfc2833  disallow=all  allow=g729  qualify=yes  t38pt_udptl=yes  t38pt_rtp=no  t38pt_tcp=no   -- SIP read from 10.18.188.84:50096:  ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0  Via: SIP/2.0/UDP 10.18.188.84:5060  From: ;tag=19D429E8-2084  To:
 ;tag=as3c87a22e  Date: Tue, 05 Sep 2006 19:42:28 GMT  Call-ID: [EMAIL PROTECTED]  Max-Forwards: 6  Content-Length: 0  CSeq: 101 ACK--- (9 headers 0 lines)---  Sep 5 15:30:31 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received  Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received  Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received  Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP  codec 100 received  Sep 5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown  SDP media type in offer: image 16406 udptl t38   
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[asterisk-users] asterisk t.38 fax failed

2006-09-05 Thread Kokfoo Soo 
Is T.38 fax work through Asterisk? I have the config below in my sip.conf, but the fax doesn't work and give me the CLI lines below. My current version is 1.2.10. Please help.[Inboundtopbx]type=friendcontext=pbxhost=10.18.188.84insecure=portdtmfmode=rfc2833canreinvite=nodisallow=allallow=g729allow=ulawt38pt_udptl=yest38pt_rtp=not38pt_tcp=no[OutboundfromPBX]type=peerhost=10.18.161.222 canreinvite=nodtmfmode=rfc2833disallow=allallow=g729qualify=yest38pt_udptl=yest38pt_rtp=not38pt_tcp=no-- SIP read from 10.18.188.84:50096: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0Via: SIP/2.0/UDP 10.18.188.84:5060From: sip:[EMAIL PROTECTED];tag=19D429E8-2084To: sip:[EMAIL PROTECTED];tag=as3c87a22eDate: Tue, 05 Sep 2006 19:42:28 GMTCall-ID:
 [EMAIL PROTECTED]Max-Forwards: 6Content-Length: 0CSeq: 101 ACK--- (9 headers 0 lines)---Sep 5 15:30:31 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 receivedSep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 receivedSep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 receivedSep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 receivedSep 5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown SDP media type in offer: image 16406 udptl t38 
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