similar to: Priority Voicemail

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

Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP

Hello all, I'm working on a little project right now and have ran into a snag. Was hoping someone would be kind enough to give me a few pointers to help me get past the current issue... I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...) that I'm trying to get to play nice with Asterisk 1.4. I've got it to the point where the AudioCodes box picks up

Hi, I have a spiffy new gateway which seems quite promising. It's the Audiocodes MP114 FXS_FXO (2 of each). I have got it configured and working reasonably well, but have a couple of issues. 1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway after 10 seconds. This isn't asterisk saying it's quiet for 10 seconds, it's the gateway deciding it's time to go

I was looking at some fixes in the replies to the chan_sip.c problems and I am wondering if I am seeing the same thing in the earlier file version. I just checked to see that my chan_sip.c is version 1.179 when I did my checkout so I never had the later versions. The problem that I am seeing is that DTMF is not going from 1 SIP device to another and sometimes voice is not going from 1 SIP device

Is anyone on the list using an Audiocodes gateway with asterisk and SIP? I'm looking at that platform, but I have a couple of issues: 1) Echo cancellation. The echo that I'm hearing with an X100P is unacceptable. Does the Audiocodes do better? 2) Line signalling. I'm using Kewlstart with the X100P, but it looks like the audiocodes uses loopstart only. How does this work with

How to test 911 call? I'm using Audiocodes and it setup to strip the first number but I've never tested the 911 call. I don't want to go live as they might charge me. -- Joseph

I have some Audiocodes units which appear to be running Linux, according to the unit's own "System Log" kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 However my contact at Audiocodes claims otherwise On 12/4/06, Yaniv Nizan <Yaniv.Nizan@audiocodes.com> wrote: > > > > I doubt that we are running Linux on the MP-202. Perhaps there is a

I have an Audiocodes MP-118FXO in production. When an outbound call is made and the remote party hangs up, the Audiocodes hangs up the call immediately. But if an incoming call is received and the remote party hangs up, the Audiocodes does not hang up immediately. I have tinkered with Current Disconnect and Polarity Reversal settings, to no avail. Anyone experienced this issue with Audiocodes or

Thanks for all the comments on the 3Com phones. Thankfully, there is a large number of phones out there to dig through looking for the right solution. What I have not been able to find, after spending all weekend looking, is a good solution for an attendant console. We have 2 receptionists that need to be able to view all 60+ phones (we could probably weed it down a bit if we had to,

I am currently using Asterisk with G.711 codecs and in-band DTMF for several Cisco 7960's and an Audiocodes GW. When allowing out-of-band DTMF, I could use voicemail menus and anything else on Asterisk that required DTMF but I could not get the DTMF relayed out of the GW. Has anyone verified that this works between 2 SIP devices? If so, I would be interested in your settings. Also, I would

Hello, I'm helping a colleague (*) which has the following setup: ITSP --- <T.38 capable PJSIP trunk> --- Asterisk 13 --- <PJSIP>-- Audiocodes MP-112 --- <FXO/FXS> --- Fax machine My issue is the following : Audiocodes gateway reject INVITEs with 488 Not Acceptable Here It seems this gateway requires t38 settings to be present in SDP body in the very first INVITE. My

Sean, > I am thinking of proposing this system to my partner corp which would > entail around 13 extensions and 6 lines... How would I give someone > upstairs the ability to view if each user was on the phone or not? <-- > should probably be a new thread.... Currently they have 18 button > phones that are programmed with the incomming lines, then the users > (LED's glow

Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make calls on Audiocodes gateway . I got the folloowing messages on asterisk: [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80

I have AudioCodes MP-114 and I'm trying to configure SAS (Stand Alone Survivability); when Asterisk is down the MediaPack gateway should forward the call IN/OUT through the gateway (without asterisk in the middle), but it is not working. I'm working with tech. support from the source I purchase the unit from they we are just emailing back and forth and the unit is still not working. Can

Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061022/6ca85b8c/attachment.htm

I just updated one of my stable asterisk systems to head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from '<sip:3034585725@voip.livewirenet.com;user=phone>' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable..

I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO) AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to interpret it as authentication: [Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch, have <pstn-5665>, digest has <pstn-1270> [Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite:

Dear R_users, I have some troubles with a visual basic application I have recently created. This application automatically creates and executes an R script based on a dataset and settings defined by the user. The .r file (Create_Diagnostic_plots.r) is saved first in a given folder and then call using a CMD BATCH instruction in the shell. The problem is that the script is not executed on all

Has anyone experienced any date/time problems on Asterisk? I have noticed it on earlier builds and I believe on this build: CVS-02/16/04-23:57:06 Specifically, I have seen the following problems: 1. Change system time using the "date" command while Asterisk is running. Asterisk does not synchronize with new system time. 2. Daylight savings time goes on/off at the designated