similar to: Out Of Band DTMF and SIP

Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which

Andrew, I don't think it's a Polycom issue. We took Asterisk out of the picture and had our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session Progress, and the polycom's play the correct tones in this case. We WANT Asterisk to send progress tones in band. In our case it IS needed.

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

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

I've MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension. Anything longer than 3-digits is cut off, example I dial extension 1000: [Apr 29 10:03:30] NOTICE[3817][C-000000e9]: chan_sip.c:25902 handle_request_invite: Call from '54' (10.0.0.115:5060) to extension

Is anyone on this list using and audiocodes FXO gateway? I have Asterisk(1.07 on OS X) setup and working fine, including SIP phones and IAX2 phones - I can make outbound calls just fine and receive inbound calls just fine. However, I can't seem to find the right series of DTMF settings on the AudioCodes to allow DTMF tones to be sent after an outbound call is connected(phone banking,

Hello, Anyone here have experience with Audiocodes MediaPack MP-108 Gateways? I would be willing to pay someone for advice and support with configuring my gateways for a telemarketing project I am starting. My experience is somewhat limited but all I want to do is make outbound calls just like I would on normal pots lines. (That's the best way to explain it) I do not need any special

Hello, I am fairly new to asterisk. I am currently using asterisk as a more convenient sip side voicemail system. My problem: I have cisco 7960 phones whose out of band dtmf tones are recognized properly(when dtmfmode=rfc2833) by asterisk but whose in-band dtmf tones are recognised poorly(when dtmfmode=inband) . For example 7999 comes out as 799999, 4242 comes out as 442422 ... etc I

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

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 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! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers

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 currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

Does anyone have any information for me? Welinghton. Citando Welinghton Magno Guimaraes <welinghton.guimaraes at ufvjm.edu.br>: > Hello! > ? > I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) > to make external links. Does anyone have any manual or instructions on > how to proceed? > ? > Asterisk ?=>? Mediant 1000 (AudioCodes) ?=>?

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

Topology: PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server When I make a call to a VoIP user from the PSTN, the call gets routed through the PBX, and Cisco. Because of that the DTMF tones are passed inband, which I can hear on the VoIP end of the call. However, I have one extension on asterisk set up so that I can check voice mail when away from my

On 02/15/2018 03:44 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >> >> IN audocodes setting I have: >> "EndPoint Phone Number" >> >> Channel: 3 phone number: pstn-4444 >> Channel: 4 phone number: pstn-9998