similar to: Sound quality issue

Anyone know if it is possible to control how aggressively the "Aggressive" mode behaves. Meaning, is it possible to dial back the aggressive mode to have a happy medium between Regular and the Aggressive defaults. I have a situation where Normal echo cancellation is not quite enough, however when I turn on aggressive mode We are attacking it to hard and I am unhappy with the walkie

Hey sorry to hijack this thread, but I just remembered a request I wanted to make to the speex devs. I tried using the activity detector, but I just couldn't get it working well. I ended up using my own, where I think it just considered voice on if it passed a certain threshold (I know, pretty primitive). I also tried one that checked for a signal, like if the strongest frequency

> Anyway, my request is, can you build in a pre and post buffer into the > VAD? In mine, if I detect voice any time between now and say a quarter > second later, I start sending, and then I wait a half second or whatever > after I stop detecting. You pretty much have to have this, or people > start getting anxious talking over an internet stream. They have to > enunciate

Hello everyone, I'm having a hard time configuring my router to forward asterisk traffic correctly. I have the following ports being forwarded to asterisk: 5060, 10000-20000 Now, I can register the accounts when outside the network and I can call every extension that is inside the network. The problem is that I can't ear anything nor can the phones inside the network phone the

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on

Well, to do an upgrade on a traditional system you have the same issues, perhaps even worse as everything is physically wired to one system. To develop for production you must have a dev environment, a beta test and a scheduled release right? Todd Jonathan Moore <moorejon@usd465.com> wrote: __________ >These are good issues, but I am even thinking of something simpler and more

Hello, After I have re-read the "PJSIP Advanced Codec negotiation" document, it occurred to me that the desired behavior should actually happen automatically, just due to the codec negotiation logic, but it looks like asterisk doesn't actually follow the described logic which is likely a bug. Can you please follow with me through a simple sip call and see if I'm missing

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,

Hello people, I've ran into two problem that I can't seem to be able to solve on my own. Here's my scenario (running Asterisk 13.28.1): In short: - Asterisk behaves unexpectedly (at least to me) when negotiating between endpoints             that have a different but intersecting set of codecs (preventing direct media flow).           - Also, when an endpoint sends RTP with an

have problems with outgoing calls. When I tried this, the same way you did, I could make calles externally but had no audio each way reguardless of what I tried to pass to the sip provider. Best bet is to use what your sip provider can use or find another provider that that can do g722. That's what I did when I wanted to use g726. my2cents On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys

Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722&ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 -> ulaw to ulaw is chosen 100 dials 101 -> g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format.

Hi! I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it

Hello! On 20.10.20 at 14:00 Asterisk Development Team wrote: > The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. > This release is available for immediate download at > https://downloads.asterisk.org/pub/telephony/asterisk I just tested the new codec negotiation feature and unfortunately wasn't able to get it working as expected. I tried several

I have Polycom phones that support the g722 codec. Adding allow=g722 to the [general] section of sip.conf works great and I can make calls between the phones using g722. However Asterisk is negotiating g722 for calls going out my voip provider and transcoding these to ulaw. In sip.conf for the provider I have deny=all and allow=ulaw. This can cause potential audio degrading and wastes cpu cycles.

For about 10 months I have been running a succession of Asterisk SVN trunk versions on an Athlon 64 X2 4400+ based machine with OpenSuSE 10.2 at my home. I have a variety of SIP phones (mostly Polycom) internally; my external connections are two POTS lines on a TDM400P (with HPEC) and an IAX2 link to a VoIP provider. I had Asterisk configured to allow G.722 and u-law on the Polycom phones,

Hi! Maybe you can help me. I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722, i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it possible that

Hello list, I'm having some issues with some phones that don't stop ringing after the call is answered somewhere else. Basically, a call comes, enters a queue and all the phones in the queue ring. The problem is that when the call is answered, some phones don't stop ringing. I don't know if it is a configuration file, but I don't think so. queues.conf, sip.conf and

Well, I'm trying to migrate to chan_pjsip so that I don't have to do that. It's so surprising that the issue so seemingly obvious and trivial hasn't been addressed yet that I wanted to query the collective wisdom of this list to verify my observations. Thanks for github pointer. Michael On 7/5/23 16:46, asterisk at phreaknet.org wrote: > On 7/5/2023 4:19 PM, Michael

Hi there, I've been playing with a SNOM 360 and 190 trying to get them talk to each other using g722 with 16 kHz. However all I see in the SIP log codec negotiation is "g722/8000" which makes me believe that this is only a 8 kHz link (and that's what it sounds like). Anyone every managed to establish a 16 kHz wideband call between SNOM phones? Cheers, Philipp

Greetings everyone, I'm trying to compile asterisk for an AVR32 (Atmel NGW100). Buildroot for AVR32 already has the asterisk package, though it has bugs. Firstly it tries to apply a patch for 1.2 on a 1.6, but deleting the contents of the patch file did the trick. Now, the problem is making asterisk. The first error is because asterisk needed to be ./configure:ed. Trying to just do