search for: g711a

...an dial *98 and PSTN calls, and yes they are all in the same context since April 2006! SIP to PSTN - OK SIP to IAX - OK This is a graph from ethereal: Dialing 4214, my own SIP extension! |Time | 192.168.34.26 | XXX.XXX.XX.XX | |11,219 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:4214@194.117.36.75:5060 To:sip:4214@XXX.XXX.XX.XX:5060 | |(2752) ------------------> (5060) | |11,721 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:4214@194.117.36.75:5060 To:sip:4214@XXX...

...a way how i can solve the following problem. we want to offer our customers in the country side also isdn over voip but we have to use internet connections from another company for this. This company offers a QoS on this connections but only with 192kbit bandwith and with the ATM headers a normal g711a call has exactly 103,5 kbit/s so we can only use 1 channel but for isdn we need 2 :( my idea was if i can find a way that the first call of a peer has g711a codec (like normally) and if a second call comes in, or has to be placed for this peer i only offer g726 (40kbit) so i dont have a bandwith i...

I've got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP? thanks, -reed

...onse="94f0139b69bb01ddc4aa362ab3edc130" User-Agent: Linksys/SPA2100-3.3.6 I'm using the following features: - Network jitter buffer: very high - Jitter buffer adjustment: disable - Call Waiting: no - 3 Way Calling: no - Echo Canceller: no - Silence suppression: no - Preferred Codec: G711a - Use pref. codec only: yes - Silence Threshold = medium - Echo Canc Enable = no - Echo Canc Adapt Enable = no - Echo Supp Enable = no - FAX CED Detect Enable = no - FAX CNG Detect Enable = no - FAX Passthru Codec = G711a - FAX Passthru Method = NSE - FAX Process NSE = yes - FAX Disable ECAN = no -...

...d here is my scenario: fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 -> SIP ATA (T38 enabled) -> fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys SPA2102, Asterisk 1.4.22 (configured with t38pt_udptl = yes) and I have a pretty good link so faxes are going through even if T38 is switched off. Interesting thing is that faxes are going through even when one ATA is T38 enabled and the other isn't... Thanks for...

...nation Address","Destination Port","SSRC","Payload","Packets","Lost","Max Delta (ms)","Max Jitter","Mean Jitter","Status" "6X.XXX.XXX.XXX",34170,"1XX.XXX.XXX.XXX",10602,277011456,"g711A",7289,0,21. 303999999996449,21.265543809819981,0.073286945955809715,"" "1XX.XXX.XXX.XXX",10602,"6X.XXX.XXX.XXX",34170,2020146713,"g711A",2099,0,36. 296999999998661,2.9025967411766738,0.97877393850963945,"" "1XX.XXX.XXX.XXX",10602,&quo...

...ANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve these values in my MySQL CDR table in order to calculate a MOS value: "ssrc=592614191;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=20734;rlp=0;rtt=0.094000" codec used: g711a -- -- -- Marc LEURENT lftsy at leurent.eu

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

...gin successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX. >From the Asterisk CLI I see the following errors; i) Unknown RTP codec 72 received ii) RFC3389 support incomplete Anyone got ideas on how I can go about this? Thanks in advance. Julius Kidubuka "When you do the common things...

...related ; options ;----------------------------------------- [codecs] ; ; Define the codec list of the channel driver. ; Every "codec" option may have a "frames" option ; associated with it. ; Valid values for the "codec" option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K...

...no h245inSetup=no inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout userInputMode=TONE amaFlags=default accountCode=H323 language=en context=voip-h323 [register] alias=ASTERISK [codecs] codec=G711A frames=20 [22xx2912] type=friend ip=AVS@210.21.118.XXX port=1720 alias=HMA0200.10szxn-xxxx e164=22xx2912 context=default disallow=all allow=ulaw dtmfmode=rfc2833 [22xx2913] type=friend ip=AVS@210.21.118.XXX port=1720 alias=HMA0200.10szxn-xxxx e164=22xx2913 context=default disallow=all allow=ulaw...

...f file generated as expected) I get the warning 'FAX CNG detected but no fax extension' on the consol. If the fax is received ok then what 'fax extension' does it expect and what should I do there? My Setup: Sender -> Public PSTN -> provider -> SIP trunk (configured with G711a) -> Asterisk (13.6.0) My extension.conf on relevant section is this (obviously this is not production code): exten => s,1,Answer() same => n,Verbose(0, Attempt to Receive FAX) same => n,Set(FAXOPT(gateway)=no) same => n,ReceiveFax(/var/workspace/testfax.tiff,d) same =...

Hello, SIPp's PCAP play feature can replay pre-recorded audio stream towards destination (see [1]). Doc mentions tcpdump and Wireshark as tools to record such RTP streams without further details. Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/ directory. Sample pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to 10.1.6.18:2006 1. How can you "forge" IPs and/or ports of a pcap file ? 2. When generating simultaneous calls from one source device to a single target device, do you need to have specific PCAP files (one specific for each call) with specific...

...related ; options ;----------------------------------------- [codecs] ; ; Define the codec list of the channel driver. ; Every "codec" option may have a "frames" option ; associated with it. ; Valid values for the "codec" option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K...

Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 -> asterisk -> DIAX :: sound is ok. When I perform an outbound call: DIAX -> Asterisk -> h323 :: sound is terrible and CPU load is 80% When I perform an asteisk internal call with DIAX: DIAX -> asterisk IVR :: sound i...

...10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 amaFlags=default accountCode=H323 [223];OpenPhone type=friend defaultip=193.25.30.223 username=223 context=default [register] ..... [codecs] codec=G711A frames=20 codec=G711U frames=20 codec=GSM0610 frames=4 /////////////////////////////////////////////// /////////////////////////////////////////////// Extension.conf [general] static=yes writeprotect=no [default] ;Xlite exten => 224,1,SetLanguage(de) exten => 224,2,Dial(SIP/xlite1,10) exte...

Hello all, I've got an AVM Fritz!Card PCI that I'm using with Asterisk under chan_capi (0.3.5) Phones on our internal network all use g711u. I'm aware the chan_capi uses g711a by default. To reduce the need for transcoding, I decided to make everything use the same codec. First I changed all the phones (Cisco models 7905G, 7940G and ATA 186 all running SIP) to g711a, but it seemed to break the echo suppressor in the ATA 186, so that wasn't a viable option. I then...

I use codec g711u or g711a but comuncation between two sip client (XTen lite) have bastard dalay of 0,5 - 1 second Is it normal ? Are there any configuration to solve problem ? Thanks all

I have two phones, one does not need stun, the other one needs. All settings are identically, except the number/password and said above stun - not stun I use codec in the order: g729 g711u g711a Any ideas, why the user can hear me, but I cannot hear him (stun) while the other user without stun has no problem. bye Ronald

...=100 outboundMax=100 inboundMax=100 simultaneousMax=200 wrapLibTraceLevel=9 libTraceLevel=9 libTraceFile=/tmp/oh323_debug.log gatekeeper=DISABLE gatekeeperTTL=300 userInputMode=TONE amaFlags=default accountCode=H323 context=sip [register] alias=ASTERIX [codecs] ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B...