similar to: Asterisk G722

Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the

Hi! When I'm mounting a client with fstab and a server-supplied client-spec file I get the following in the server-log: 2008-09-15 15:04:42 D [tcp-server.c:145:tcp_server_notify] server: Registering socket (63) for new transport object of 192.168.10.12 2008-09-15 15:04:42 E [server-protocol.c:5212:mop_getspec] server: Unable to open /etc/glusterfs/glusterfs-client.vol.192.168.10.12 (No such

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

Hello All When sending the Opus Packet in RTP Payload, the compressed frame is the output of the encoder? Also the config value as given in the RFC6716, 16...19 | CELT-only | NB | 2.5, 5, 10, 20 ms 16 corresponds to 2.5 ms 17 corresponds to 5 ms 18 corresponds to 10 ms 19 corresponds to 20 ms Is this correct representation of the data? Also in the RFC3551 the payload

I recently switched from Mandrake to Redhat and I noticed that vmail.cgi does not work with the default apache installation that comes with Redhat. Here is what I get in my error logs: [Thu Dec 04 11:59:57 2003] [notice] suEXEC mechanism enabled (wrapper: /usr/sbin/suexec) [Thu Dec 04 11:59:58 2003] [notice] Digest: generating secret for digest authentication ... [Thu Dec 04 11:59:58 2003]

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Dear, When I start asterisk -vvvvvvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14)

Hello, I try to move our asterisk installation (3 Asterisk servers in different offices connected using IAX and a lot of SIP phones, as well as ISDN connections using CAPI) to use G.722 instead of G.711. Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves the gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for

Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I migrated our DCs from 4.5/internal dns to 4.7.1/bind9_dlz. Short summary of the steps taken: - added a new temp dc, - removed the old DCs - cleaned sam database - installed new DCs, with their old dns/ip - removed the temp dc again - synced sysvol and all is looking well: no db errors, no replication issues, ldapcmp matches across DCs, etc. So, I took things to production today, and

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Hi list! Again Wengo has made changes to their servers that require modifications to * configs. Is there anyone that has the 'new' wengo working with asterisk that could post their configs? Also they switched codecs, now G720a is required to connect. I can only find an (open) G729 codec, is this the same as G729a? Thanks!

Hello list, I am trying to solve a problem and after unsucessfully chasing forums and google for some hours, I turn to you in hope of a solution. I feel it's just a configuration issue but I just can't get my head wrapped around it. The situation is basically this: I have an Asterisk connected to an Alcatel OmniPCX via SIP. Asterisk only ever does SIP and has no dedicated hardware

Hi, We are having major problems with a remote site that links to the head office via a VPN tunnel. The phones will register fine and work for a few minutes to hours but then will drop their connection and will no register to asterisk even with a restart of the phone. We have 2 other remote sites that work exactly same and they are not having any issues so i believe it has to be be something

Hi list! I have a strange echo problem. Two days ago I setup * 1.0.6. at a friend of mine. Just an * server and for outbound calls wengo.fr was used to place calls via sip. He had a strange echo on the line I didn't experience on my setup. Today I upgraded my asterisk 1.0.5 to 1.0.6 and suddenly I have an echo too on sip calls thru wengo!! I already verified wengo was not the source of

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

Would just like to warn everybody for Wengo.fr Once you sign up there is no possibility to remove your credit card and even though you send them resignation letters they keep charging your credit card. Now I understand what they mean when they say `unlimited subscription'.

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this

Hi, I wanted to follow up on this thread about WB support on the MeetMe bridge that is in 1.6.2. Does it only work for G722 or any WB codec ? I am working with another 16k WB codec that I can transcode to 722 and vice versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722 with any other WB codec natively(without downscaling). Thanks, Serhad Doken ------------------> Razza wrote: