similar to: IAX Experiences [WAS: Question about DSP in Digium card]

Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX <-> ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- -------------- next part

Hi all I have a server that has two TE412P (T1/E1+DSP) cards installed. One of them configured as an E1 PRI connected to PSTN and another one configured as a T1 E&M connected to Avaya PBX. Each card only uses two ports, so there are 2 E1 lines and 2 T1 lines connecting to this server. The purpose of this server is as a TDM trunk gateway that gets call from E1/T1 and then forward to an IP-PBX

Hello, I would like to double check what Echo Cancellation my Digium Card uses. I thought I bought the little more expensive card that integrates EchoCancellation. How can I check? root at sn1:~# zaptel_hardware pci:0000:0b:08.0 wcte12xp+ d161:8000 Wildcard TE121 root at sn1:~# ztcfg -v Zaptel Version: SVN-branch-1.4-r4309 Echo Canceller: MG2 Is MG2 the correct one that I am supposed

Bourvine, > > So, why won't we save the big bucks we pay them, hire two professionals > (who cost less) and support an open source code by ourselves? This way > we depend on ourselves only. > > > > Thanks, __Yehavi: I remember hearing University of Pennsylvania have been using Asterisk for sometime. I am not certain where I came across that

How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/f9612f97/attachment.htm

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

Hello everyone, I'm streaming audio on CentOS 5.8 with no problem, even on a cheap sound card using DarkIce as the input tool. For the input under CentOS5, I use: device = /dev/dsp # OSS DSP soundcard device for the audio input But under CentOS 6.2, there is no such device. I see /dev/snd, and it has: controlC0 hwC0D2 midiC0D1 pcmC0D0p pcmC0D2p pcmC1D0p seq controlC1 hwC1D0

Hi, Anyone used this service, any comments on reliability/support? Thanks John

Just noticed this today: Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based Echo Cancellation Module<http://www.voipsupply.com/product_info.php?products_id=3352> It's about time Digium got on the ball and made PCI-e cards. What are people's experiences with this card? Anyone know if there are plans for a PCI-e analog card for FXO use? -------------- next part

We have a customer who is connected to our PSTN gateway using IAX and noticing that even when the traffic from their site is modest their outbound audio has short dropouts. Inbound audio is fine. (They have ADSL so it is expected that outbound audio would be the first to experience problems.) We have several questions to pose to the collective wisdom of this list. Q1: Are there any statistics

hello, is there any equivalent, that is currently usefull, if I have some iax connections with jitter spikes and another with minimal jitter? for my jittery connections, I don't like to shrink jitter buffer too fast, because another jitter spike can occur again and small jb can't cover it. as I read, in older iax jb implementation, this can be solved using "jittershrinkrate="

Hi there I have a question regarding IAX jitter. I have 3 users on a LAN dialing into a Meetme conference on an Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the users the audio is fine, but for the 3rd user there is intermittent break up in the audio when they are receiving. I have had a look at "iax2 show channels" and

I wanted to know if anyone had any good experiences with particular payment gateways and Rails? and what did it take to integrate? My current client uses PayPal for the credit card payment processing, and would like to stay with them, if possible. So has anyone had any experiences with integrating with PayPal? I''d love to hear who provides your payment services? -- Posted via

There is a serious bug in Sangoma transcoding cards. The card has an internal, small jitter buffer and it drops samples from the audio stream when there is high jitter in the network. The bandwidth is cheap now so for me the only reason to use transcoding is where I have low-bandwidth-high-jitter links. Sangoma said they will not fix it and we had to go back to software transconding. Do you have

Hi - Does anybody know if an FXS generates line voltage when Dahdi/Zaptel is disabled? Thanks, Noah

Hi All - I've been trying to compile Zaptel w/ HPEC, but I've been unsuccessful. The system is CentOS 4.4, zaptel version 1.2.15. I believe I've got all the requisite files, and they're in the right locations in the zaptel tree. When I compile, I get the following warning from make: Warning: could not find /usr/src/zaptel-1.2.15/hpec/.hpec_x86_32.o.cmd for

Hi - Just wondering if anyone has gotten a Polycom VVX phone to successfully do an Auto Answer with asterisk. I have an older generation of Polycom phones that do this just fine, but I can't seem to make the VVX phones work. I tried the guide here: http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167 And I have this in my diaplan:

Hi all, For the record this approach isnt 100% backwards compatible, because names(mergeddf) will e incompatibly different. Thatx why i claimed bakcwards compatable-ish That said its still worth considering imho because of the reasons stated (and honestly one particular simple reading of the docs might suggest that this was thr intended behavior all along). Im not a member of Rcore through so i

Thanks Martin! Can you clarify the functionality of the 'no.dups' argument so I can change my patch to `data.table:::merge.data.table` accordingly? - When `no.dups=TRUE` will the suffix to the by.x column name? Or will it take the functionality of the second functionality where only the column in y has the suffix added? - When `no.dups=FALSE` will the output be the same as it currently

We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the