similar to: Asterisk Conferencing using g729

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

I have come to realize that I don't have to have a g729a license in order to make use of an ATA-186 or 7460 connecting to another 7460. I just need to allow the codec in sip.conf. Now what ramification does that have when I dial out over one of my analog line (connected to * by a channelbank and a T100P) using my 7460 or ATA-186. The only benefit I am looking for is reduced bandwidth

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

I wonder if anyone else's Asterisk box drops the connection to voicemail after 30 secs even when the maxmessage parameter is set to 180 (3 mins). Here is the general section of my voicemail: [general] ; format=wav49 maxmessage=180 attach=yes Even if it only gave the caller 30 sec to leave a message it would be nice to tell the caller that they have run out of time before ending the

Hi, I set a preference order of the codecs to my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls of not registered phones disallow = all allow = g729 allow = g723 allow = alaw allow = ulaw Connected a 'Sipura SPA' sip phone to asterisk with g729 as its preferred codec. Problem: asterisk cannot make

Just come across a problem - we have sent out heaps of PAP-2 ATA's and just discovered that when joined in a conference they are choppy on the up leg (so the other users in the conference will hear them with a choppy sound) but the down leg is perfectly fine (so the end user can hear the conference participants perfectly). I have tested the same setup with different brands of ATA's

Hi, I am running asterisk 1.6.1.6 with a howler screamer card. I have g729 and alaw trunks from a pbx /sip providers. The howler screamer will only transcode from g729 to alaw / ulaw but my voice prompts are in SLIN and throws errors when i try and access these applications. Is it simply a case of converting the prompts into other codecs and asterisk will pick these up? ? Thanks

Can anyone tell me how I can implement the features added in the following link for call transfer? The authors seem to feel they are finished but it doesn't appear to have been integrated into what everyone can download. It is referred to as a patch but I don't understand how it could be applied. Here is the link: http://bugs.digium.com/bug_view_page.php?bug_id=0002010 I guess I just

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) --

Hello all, I am running Asterisk 1.4.17. I have 2 Linksys SPA3102's and one PAP2-NA (I have a second on order). They have G729a built into them. This is supposed to be compatable with G729. I was trying to have them use that codec when they talk to each other, but it seems they always switch to alaw or ulaw (depending on my sip.conf file). Shouldn't they be able to use G729a in

Hello, I am very new to asterisk and voip in general and so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e FXO)? Or it can be used as both? My understanding is that its just like another ATA186. Is that true? I guess what I

Hello, How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error: Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No compatible codecs! My sipura adapter

Hi, I have not implemented any of the spa-2000's yet. Do they work ok with asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is it two fxs ports with the same extension?

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

Hi, I've Sipura 1001 phone adapter. In the settings it has separate Line 1 and Line 2 tabs, which apparently means it can control two separate phone lines. I've Asterisk@Home server and want to setup two different extensions for two phones, i.e. 201 and 202. After doing all this, I can see in Info tab that both lines are registered but only one phone gets the dials tone. Am I doing