similar to: most common VOIP echo simulaton for research purposes ?

Hi i am using pap2 phone adaptors as clients to connect to asterisk server i am able to make calls but i cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec ) before the call gets dialled i assume that the problem would be due to the dial plan in PAP2 if so please help me changing it

HI I have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ? i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3050 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/3396d198/smime.bin

Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help,

Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob.

Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

Hi, I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? Thanks in

Hi, I'm about to test VOIP connection (from my ISP provider) directly through dedicated network card instead of going through ADSL gateway with analog phone port - SPA 3000 - Asterisk. I need to have eth2 set on dhcp (to retrieve IP automatically) and then work with it under Asterisk as dedicated VOIP trunk. Anyone with more insight how to setup such situation ? Any more info anywhere

Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this

Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. But coming in via the IAX2 route leaves me with a silent phone. The prompts all work still letting me navigate the menu. But just can't hear anything. This is with A@H 2.8 (Asterisk 1.2.7.1, with FreePBX 2.1.0 also installed) Any thoughts on where to

I am seeing a problem with the Xen emulated serial console. When running a Linux 2.6.20 HVM guest that has CONFIG_HOTPLUG_CPU=n, the guest blocks on output to the console until it receives input keypresses from `xm console`. This prevents the guest from booting up without banging on some keys, and makes interactive use of the console difficult. By bisecting Linux kernel commits, I found that

> I had the idea of implementing a lot of the operations in FFTs. ( for > example, it is possible to do auto-correlation and FIR filtering using > FFTs.) There are two advantages to this. > 1. It's almost always faster > 2. By swapping fft implementations, it could be easy to recompile for > fixed or floating point versions. No. FFT's require higher precision than

Hello, I try to compile the driver zaptel and they give the following error: linux01:/usr/src/zaptel# make install gcc -Iir/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -Iir/drivers/ l -I. -Wstrict-prototypes -fomit-frame-pointer -Iir/drivers/net/wan -Iir /net -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c In file included from zaptel.c:44: /usr/include/linux/module.h:21:

<with Prof. Farnsworth voice> "Good News, everyone". I've just kicked off project "Sphinx". Which is supposed to sound like "Speex" merged with "INT". ;) Meaning I am working on an integer encoder and decoder. It looks like I will be pulling in a new "integer plumbing" into the foundation of the codec, comparing the results with the old

We're having problems with FreeBSD 5.4, 6.0, and 6.1 and an ICP Vortex GDT8546RZ 4 port SATA RAID card in a Tyan S2460 system with dual AMD Athlon MP 1600+ CPUs. We do not have any problems with this configuration under FreeBSD 4.11, and we have the same ICP cards in Tyan based Opterion system (S2882 and S4882) run with out problems under FreeBSD 5.4 and 6.1. We can reproduce the problem on

Hi, I have a 4.7-RELEASE system. I used the following cvsupfile to update my system: *default host=cvsup2.FreeBSD.org *default base=/usr *default prefix=/usr *default release=cvs *default tag=RELENG_4_8 *default delete use-rel-suffix src-all *default tag=. I then proceeded to do: cd /usr/src make buildworld I got a bunch of errors which occurred when building kdump:

Hi, Created a zpool with 64k recordsize and enabled dedupe on it. zpool create -O recordsize=64k TestPool device1 zfs set dedup=on TestPool I copied files onto this pool over nfs from a windows client. Here is the output of zpool list Prompt:~# zpool list NAME SIZE ALLOC FREE CAP DEDUP HEALTH ALTROOT TestPool 696G 19.1G 677G 2% 1.13x ONLINE - When I ran a

Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we

-----Original Message----- From: Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> To: ??????? ??????? <altersoft at mail.ru> Date: Sat, 02 Aug 2008 07:54:34 -0400 Subject: Re: [Speex-dev] fir_mem16,iir_mem16 and filter_mem16 optimisations > > ??????? ??????? a ?crit : > > I have some questions about that functions: fir_mem16, iir_mem16 and filter_mem16. > >