similar to: Flash hook not going through SPA-2002

Greetings, I have one PSTN line connected to my Asterisk@ Home box with call waiting. I also have an SPA-2002 connected to an analog phone. When I am calling on the PSTN and a call waiting beep comes through, I can hear it, but when I press the flash key, nothing happens. It is as if the Sipura is not passing the flash through. I monitor the asterisk box with the verbosity turned up, but

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

Hi there everyone! I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of

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 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.

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

All: The latest draft RTP Payload Format for Speex is available via the IETF. See below for details. Greg ---------- Forwarded message ---------- Date: Tue, 09 Mar 2004 15:56:23 -0500 From: Internet-Drafts@ietf.org To: IETF-Announce: ; Subject: I-D ACTION:draft-herlein-avt-rtp-speex-00.txt A New Internet-Draft is available from the on-line Internet-Drafts directories. <p>

Hi Robert, There's also a TI_C6X_ASM config option, that causes the fixed_c6x.h header to be used, but I think it hasn't been tested in years. I don't know if it still works, but if not it's probably not too hard to fix (patch welcome). The fixed_c6x.h file can also probably be extended to cover more of the C6x arithmetic operators. Beyond that, you'd have to go to

interesting for anyone testing out speex :) kfish. ----- Forwarded message from Alan Clark <alan.d.clark@telchemy.com> ----- From: Alan Clark <alan.d.clark@telchemy.com> To: avt@ietf.org Date: Mon, 13 Sep 2004 12:57:01 -0400 X-Mailer: Microsoft Outlook IMO, Build 9.0.6604 (9.0.2911.0) Subject: [AVT] Open Speech Repository We've started to build a database of speech samples in

does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term "remote provisioning". they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i

Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001>

Hi all, Below is the Vorbis RTP Internet Draft as sent to the AVT working group of the IETF. Comments and feedback is still welcomed from the Vorbis community. Cheers Phil ---------------------------8<-----------------8<------------------------ Network Working Group Phil Kerr Internet-Draft The Ogg Vorbis January 07, 2003 Community / OpenDrama

Hi All, Please find below an updated Vorbis-RTP Internet Draft document for review and discussion at the Xiph IRC meeting on Saturday. The changes in this version have been: Codebook caching mechanism Expanded SDP parameters Expanded MIME section Expanded introduction Packet loss section Minor tweaks and clarity changes to text There are probably some minor tweaks to the formatting needed

Hello, We have problems with Asterisk and Sipura SPA-2002. SPA is behind the NAT. Asterisk has nat=yes. Sometimes call doesn't hangup when user finish the call and hangup the headset. In this case during all conversation SIP packets contains Call-ID: bee522ee-8efa7d25@123.123.123.123 but the final BYE packet from adapter contains Call-ID: bee522ee-8efa7d25@10.0.0.2 Is such

We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones). The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that

On Fri, 27 Feb 2009, Jean-Marc Valin wrote: > Hi Aymeric, > > Yes, I'm receiving the emails but haven't had enough time to look into > the details yet. I've seen you responded to many comments, so what are > the ones for which we still need to respond? Summary is there: https://datatracker.ietf.org/idtracker/ballot/2837/ As I understand: we need to change

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.

Hi Jean-Marc, Alfred and Greg, Are you receiving the mails from IETF about draft-ietf-avt-rtp-speex The mails are not coming from AVT mailing list, but I think we are all 3 part of a minimal list (draft-ietf-avt-rtp-speex at tools.ietf.org) dedicated to latest discussion about the draft. I have answered some questions, but there are small changes and adaptation still required to the ietf

Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly & also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to