similar to: SIP dtmf=rfc2833 not working

We have an application which is primarily DTMF driven (automated on both sides), which we are trying to deploy over VOIP and Asterisk (using some Sipuras and some IAXY's). We are finding that in around half the cases, the Asterisk server can't decode the DTMF digits from the field office (or at least some of them). Though, when we place voice calls for testing, we can hear eachother quite

I am not sure if this is the place for Digium user-to-user discussion, but... We have deployed many (20+) IAXy's in the field. At a couple of locations, the IAXy's have just stopped working after 1 or 2 days use. No lights go on, no DHCP lease is renewed as far as we can tell, and of course no dialtone and no registration with the server. Has anyone else experienced high failure rate

I'm having an issue sending DTMF to cisco dialing this extension I should hear the dtmf tone RTP playload 101 has been sent to the cisco phone, but no audio. in the dialplan exten => 8603,1,Answer(1) exten => 8603,n,sipdtmfmode(rfc2833) exten => 8603,n,SendDTMF(1|100) exten => 8603,n,hangup() sip.conf dtmfmode=rfc2833 SIPDefault.conf I did play with all possible settings for

Sorry if this was posted yesterday, I was having issues with being auto-unsubscribed because of my spam filter. Not sure if my post made it through. Hi everyone, I'm wondering if I'm missing something obvious here, or if Asterisk just doesn't support what I'm trying to do. It seems like it should be simple, but appearances can be deceiving. I've got an Asterisk box

I am having trouble with a DTMF-based application on Asterisk 1.0.3. Specifically, when two IAX2-based devices are talking, when they send DTMF to eachother, the other side only hears clicks, and maybe a millisecond of DTMF tone, but not any real duration. Furthermore, when one IAXy device calls the Echo test program, we can hear our echo, but when we punch DTMF in, we get the same effect

I have only been able to confirm this issue with my IAXy as it is the only ATA I have. I am running 1.2.5 stable. Example: User @ exten 100 (iaxy) receives call from PSTN (call 1). While on the call, another user from PSTN (call 2) calls 100 (iaxy) sending a call waiting beep to the iaxy. User 100 answers the call from PSTN call 2 putting PSTN call 1 on hold with music. While call 1 is on

I've heard comments about the instability of the IAXy (loosing IP, locking up, etc) . I want to deploy some, in a remote site. The comments on the instability suggest that the simplest approach is to power-cycle them when you have problems. The IAXy would only be in use from 3PM to 8 PM. My potential simple solution is to have a timed power brick (like what you would use for a block heater)

I'm having problems with a new asterisk PBX install. the phones/ATAs are all linksys/cisco. They all worked before with a commercial softswitch. Most of the linksys devices offer auto, inband, INFO and AVT. I'm looking for suggestions. Thanks in advance -- One day at a time, one second if that's what it takes

Hi, I've got an IAXy, and provided it with connection data - unfortunately didn't provide a fixed ip address, but left with dhcp. Everthing went fine, and I could work a while with it. After a restart of IAXy, it didn't negotiate a new IP address, nor used the old one. I even restarted the DHCP server and cleared its cache. No success. I went to another office with a different

Greetings all- Long story short - I find myself suddenly running a Asterisk PBX after old PBX suddenly died. Fortunately, I had been "playing" with Asterisk (via Trixbox) on a server in consideration of replacing our aged Merlin Legend - so over the course of last weekend, I brought my testbed PBX up to full operation and now supports about 30 users. All in all, went smoother than

Does anybody know what DTMF coding does S101i adapter using? I've been testing one for over a week and here are my observations: - DTMF signaling is working perfectly with Asterisk, much better than Sipura 3K Though, I think the Asterisk "iaxy" firmware is buggy, the unit is using auto-update feature; so I have Asterisk 1.2.13 and iaxy firmware version is: 23 When enable in

Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can

I have a lot of experience, all of it pretty good, with various Sipura products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into Asterisk as clients. Good sound quality, great reliability. I've tried two of the units named in the subject line, and frankly I'm frustrated. Calls usually start out OK, but within a brief period the sound goes totally to

Hi Jean-Marc, thank you for that suggestion! It seems that the file "fixed_c6x.h" is not part of the Opus sources, so the compiler cannot find it after enabling the TI_C6X_ASM config option. Maybe it was only part of an early version of the Opus sources? I looked for the file in versions V1.1, V1.1.1, V1.2alpha and V1.3 but did not found it. Do you have an idea, where I can get the

Hi Robert, The file is not distributed in the official releases, but I can find it in the git repository. Cheers, Jean-Marc On 10/22/2018 03:53 AM, Robert Madinger wrote: > Hi Jean-Marc, > > thank you for that suggestion! > It seems that the file "fixed_c6x.h" is not part of the Opus sources, so the compiler cannot find it after enabling the TI_C6X_ASM config option.

ala cisco 7960 -----Original Message----- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck.

There is no DNS functionality built into the IAXY, I am assuming you want to be able to home onto a SOHO site with a dynamic IPaddress. I have not been able to figure out a clean way to do theis (make it easy for the roaming sales-folk etc,) who will not have a high tolerance for running something like an IAXYPROV. You would need to somehow have an external fixed address which would

I've noticed that Asterisk only supports the following DTMF modes: inband, rfc2833, info, auto. Auto mode is broken. Though, Sipura unit have the following options: inband AVT info auto Inband + info AVT + info What is AVT? How does for example "inband + info" works, it either/or? -- #Joseph

Hi all, getting a server going wiht a few TDM400's and some phones, and some IAXys too I haven't heard any issues about AU phones being able to RING in Australia, like the problem in the UK with ring capacitors on the BT system. Are there any problems like that? Also, with the iaxy's -- they should work (and ring) in Australia right? The only hint I'm seeing around is the use

Dear Opus family, we have implemented the Opus codec at a Texas Instruments DSP C6418. It is working fine! Does anyone has experience with the configuration of the codec for a speed optimized implementation on that DSP? At the moment, we use the following settings: #define NONTHREADSAFE_PSEUDOSTACK 1 #define FIXED_POINT