search for: 711u

Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100323/da2eea19/attachment.htm

What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate c...

...#39;s, and just starting recently (after the broadvoice patch I might add) after about 1-2 days these phones ring, and answer, but we get no audio on the phones. The caller can hear us, but we cannot hear the caller. Its happened 4-5 times and is only intermittent. No errors on the console, using g.711u. Any ideas? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827x101 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041122/a2db57b5/attachment.htm

Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an asterisk box providing PRI_Network signaling on a T1 interface card using a long haul point to point ESF/B8ZS T1? I do not need the techn...

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.

...buzzing, etc). However, all the network monitoring we're doing on our fully switched, underutilized 100baseT-FD network shows that we have sub-1ms ping times and no jitter to speak of. Looking at the SPA-841's main Status page, I see call status: Line State: Connected Tone: None Encoder: G711u Decoder: G711u Type: Inbound Remote Hold: No Callback: Peer Name: xxxxxxxxxx Peer Phone: xxxxxxxxxx Duration: 00:09:53 Packets Sent: 29545 Packets Recv: 29666 Bytes Sent: 4727360 Bytes Recv: 4746560 Decode Latency: 50 ms Jitter: 0 ms Round Trip Delay: 0 ms Packets Lost: 0 Packet Error: 0 M...

Hi All, Can someone explain to me the difference between g711's ulaw and alaw codecs? Is it just different header info or is the actual payload in each encoded differently? I have thus far noe been able to find any difinative information onthe matter. All I've managed to find out that they are "similar", they sound the same and that it doesn't matter which you use. Could

...T u-Law, 8 Khz, 8 bit mono. On asterisk's side, this is what I get on the console *CLI> oh323 show info Information about active OpenH323 channel(s) -------------------------------------------- 0 : token=ip$192.168.0.101:3678/24464, state=ESTABLISHED, from_remote=1, RX/TX=8/160, format=G.711U , Any hint?

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_pro...

Hi, is there a definite answer if asterisk can pass calls between SIP and h.323 protocols? Thanks, Yiannis.

...if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodec?Configure the codec ID. * G.723.1?Codec ID 0 * G.711a?Codec ID 1 * G.711u?codec ID 2 * G.729a?codec ID 3 Thanks David

...y/distorted... However, the voice call itself sounds fine. Asterisk, the Cisco phone, and the call gateway are all configured to use rfc2833. From my research, asterisk generates progress tones out-of-band (I think) unless turned on. We don't have any problems with the progress tones when G.711u is used. Any help/ideas would be greatly appreciated. Todd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050817/f42ab485/attachment.htm

...irmware). Sometimes, when i but a call on hold, the caller has got the music, but when i "resume" the call, then the caller does not hear me (and nothing at all)... I must wait for 10, 20, sometimes 60 seconds before he could hear me again. Any body already had this problem ? I use G711u codec for the cisco and i had the probem with a IAX client (using GSM) a PSTN caller (through my SPA3K using 711u) ... Thanks a lot ;) Julien.

...network are flawless with either the ArtDio or Cisco phone, and calls between ArtDio-only or Cisco-only are flawless. What gives?? I tried setting specific audio codecs in sip.conf for each phone with disallow=all and then allow=ulaw, I set the SIP<MAC>.cnf file for the Cisco phone to be 711u as the preferred codec, and through the web interface configured the ArtDio to use 711u as it's preferred codec, to no avail. I tcpdump'ed the data during the call and set the ArtDio phone to use RTP port and Control ports within the 17000 - 32000 range (SIP<mac>.cnf sets the Cisco p...

...OTICE[24269] channel.c: Dropping incompatible voice frame on Local/[removed number]@context-5c3e,2 of format ulaw since our native format has changed to slin Can anyone provide an English translation of what this means? The extension is a Polycom IP 501 The only allowed formats are g.711u MOH is MP3 files (obvious) All prompts have been re-recorded in .ul uLaw Voicemail is recorded in wav|ulaw so there should be native playback to g.711 UAs and the wav is for windows email attachments. Outbound termination is to a Sonus GSX media gateway, g.711u is the only allowed codec for tha...

...sion from the SIP provider, the phone rings and shows CID - BUT, when I answer the phone, there is no audio either way. I thought this was a firewall issue but the clients ring and I CAN leave and retrieve voicemail. My next assumption is that it is some codec issue. The Polycom defaults to G.711u. I've tried changing this to G.729AB - but there problem persisted. Any ideas? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050602/f600ef01/attachment.htm

Can Meetme be made to work with G.729? (I gather not) If a call comes in (internally or externally), the call comes in as a G.729 call, which then re-negotiates to a G.711u call when if gets transferred to a MeetMe room. Is there a way to set up asterisk that will allow me to have internal phones renegotiate to G.711, with the external lines instead transcoding within asterisk. (runtime is more available than bandwidth). Also, does anyone know if there's a way t...

I have a SIP connection to Broadvoice and sometimes when I make outgoing calls from a SIP ATA-188 (could be the same number) (the ATA-188, is currently the only extension), there is no audio passed for 5-10 secs. I have set all the codec the same to 711u and also ensured canreinvite is set to no. Any suggestions? Places to look for? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040623/06403d74/attachment.htm

...nly problem I have is lagging. What codec should I use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I configured it to disallow all and use GSM only. In my sip config of each phone I use disallow all and allow ulaw and alaw only. I see that the Cisco phone support G.729a, G.711u and G.711a codecs. I know that Digium is selling G.729 codec on there website. Should I get that to fix my problem? The less bandwitdh the codec takes over the Internet the better chance I have that it will work fine I presume. The maximum upload speed I can reach is 64k/sec so 512kbits. But t...

...f NAT gateway port=5060 fromuser=PHONENUMBER fromdomain=216.229.127.40 username=USERNAME type=friend secret=PASSWORD auth=md5 host=216.229.127.40 defaultip=216.229.127.40 reinvite=no canreinvite=no dtmfmode=rfc2833 context=intern qualify=1000 disallow=all allow=gsm allow=ulaw They support g.729, g.711u, GSM and probably others too but I didn;t try them. In /etc/asterisk/extensions.conf place the following statements; [galaxyvoice] exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@galaxyvoice) exten => _1NXXNXXXXXX,2,Congestion That's all folks!! I still can't make the DTMF work right bu...