similar to: IAX Codec

Hi, does anybody made G723 or G729 to work with a GrandStream Phone ? I've a Cisco here and it works fine with G723, but not with my asterisk. The bandwitdh is very important, since we will have our extensions at home. I know that I have what I pay, but the phone works with cisco. Trying to use G723 or G729 Asterisk says no codec available. Does anybody have it working with any compression

Hi All; I need to buy one IAXy device, but I discovered that it supports only g711 and ADPCM codec, so I was wonder that it does not support g729 or GSM?! Anyway, what is that ADPCM and how much it consumes bandwitdh? Also, asterisk support such codec? What its name in the configuration? Any advise? Regards Bilal

I have seen the list of codecs for the ATA 186's but not sure 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

Am 31.10.2017 um 14:06 schrieb Mar?a Arrea: > > ?? If you use Linux you can use netfilter (iptables) mark feature to > track individual connections from a given IP, and above a threshold you > can shape the bandwitdh based on those iptables mark. For example, if an > IMAP connection from ip X exceed 100 megabytes of data, bandwitdh for > that ip would be limited to 100 KB/sec

Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the "pass-throu" calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729

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:

On 1/27/2016 1:30 AM, Steffen Kaiser <skdovecot at smail.inf.fh-brs.de> wrote: > Sending bandwitdh can be reduced by using BCC instead of the IMAP append > to the sent mailbox. Hi Steffen, Can you elaborate on this? I would have thought that the IMAP Append command would *save* bandwidth (as opposed to having the client save a copy to the Sent folder, thereby uploading the full

Hi, I've just joined the list. Thanks Jean-Marc and other contributors for your efforts with Speex. Anyway - I've been hired by a local firm to investigate, write-code-as-required and generally help them get the maximum bang-for-the-byte for VOIP traffic on the hyperexpensive Internet bandwidth going out of South Africa. I started the process by doing some methodical measurements and

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

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

Hello, I have two asterisk boxes running and both are using DAHDI PRI Card. I wish to know if IAX is the best method to connect both the boxes? Also, need some help with the following? 1. For incoming call on server2 I wish to run an IVR to the user for which all my prompt sound files resides on server1. Is there a way I can achieve this? 2. I am also using .call file at times to make outgoing

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Hello, I''m trying to get my shaper to work, but have only a partial success. Can someone help me with that. My setup unfortunately is not so trivial, but I think some people could have similar one... 1. There is a router connected to the internet line via interface eth0 2. There are users connected to the router via two interfaces : eth1 and wlan0 3. All users are assigned private IP

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

Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050209/8cdbeeec/attachment.htm

By default, Apple Mail downloads all e-mails from server's account. Previous versions of this client allowed to opt-out. The latest two versions? however, only allow to opt-out from downloading the attachments. The stress on the server is unbearable. We cannot ask users to be considerate: this is the default behaviour of Apple Mail. We need a server-side solution to the problem. Please

?? If you use Linux you can use netfilter (iptables) mark feature to track individual connections from a given IP, and above a threshold you can shape the bandwitdh based on those iptables mark. For example, if an IMAP connection from ip X exceed 100 megabytes of data, bandwitdh for that ip would be limited to 100 KB/sec (or whatever number). First 100 megabytes, top speed, above that they

I have a domain with about 50 mailboxes, server is located here in AUS but most of the users are on a LAN is SEAsia location. They were using POP (with Thunderbird), I suggested then can use IMAP instead, so they did. now they are asking; "Looks like Imap is adding a lot to our internet bandwidth" I guess they have some bandwidth limitation on their link I think I can understand that

Hi, I've been trying to get the G.729a beta codec running with my remote Asterisk box that talks IAX2 to my local Asterisk box. Digium fixed the problem I was having in registering the beta codec, so that now works fine. I've removed the old codec_g729b.so from /usr/lib/asterisk/modules and put in place the codec_g729a.so beta from digium FTP. My CVS build of Asterisk is about a

Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa