similar to: RE: Call being answered, but no audio on either end

I think I found the source of this. Been tracing it for a week. Look in sip.conf. It appears the definition of localnet has a bearing on how some sip devices handle invites and NAT. I had changed the localnet to 192.168.3.0, but did not change the netmask. localnet=192.168.3.0/255.255.0.0; All RFC 1918 addresses are local networks When I changed the netmask to 255.255.255.0 the problem

The best type of error possible, intermittent. We have PSTN numbers being switched to SIP then forwarded to our Asterisk server which sits inside our LAN Every once and a while (maybe 1 out of every 20 calls) goes like this: -- Executing Answer("SIP/213.199.36.50-0818e3e8", "") in new stack -- Executing Ringing("SIP/213.199.36.50-0818e3e8", "") in

http://www.zoomtel.com/products/voip_products.html anyone using these? they look very interesting in that they support ilbc, and they offer a separate cheaper model without g729 license. i'm wondering if their EC is better than the spa3000, and what the echo tail length is. -Dan

Correction: The hardware is a Wildcard T100P (not a TE110P) Thanks! > -----Original Message----- > From: Geoff Manning [mailto:gmanning@zoom.com] > Sent: Wednesday, May 18, 2005 9:07 AM > To: Asterisk Users (E-mail) > Subject: [Asterisk-Users] Integrating Asterisk into our Legacy PBX > <--Newb > > > I have been successful in setting up asterisk and making >

I received this before, and it is because you are using the wrong context in the iax.conf. For example: The context must match the username in the register statement. Iax.conf... register => username:secret@iax.host.net [username] type=friend context=iax-in user=username secret=secret auth=plaintext host=iax.hust.net ----------------------------------------------------------------

Hi List, Have you got experience with this product? http://www.voipsupply.com/product_info.php?products_id=885 From its description, it looks like the ideal appliance to set up some "double play" ISP data / telephony offer and I was wondering if anybody was using it and what it was worth. Cheers, Jean-Michel.

Hi, I've been using CentOS & Xen on a server that has 2 VM's configured. The default configuration includes one physical iface that is propagated (by a default bridge) to the VM's. Since I wanted to configure additional physical iface, define a new bridge and propagate it to the viface-s of the VM's, i configured the bridge/phys. iface and brought it up (here are

Dear all, I am getting the below errors when using asterisk. I am using sjphone for testing purpose. Below are the setting for sip.conf and extension.conf When i dial the number it rings on the remote telephone. but after ringing 1 time it will disconnect. Can anybody tell me what does this error means and the how to solve this issue. Thanking You, Joel sip.conf [general] context=default

Hi, I'm trying to get the latest version of darwine working with slickedit on my mac os x box. Everything is using the latest version. When I try to run my application I get this error: > wine: created the configuration directory '/Users/GMan/.wine' > err:module:DelayLoadFailureHook failed to delay load setupapi.dll.InstallHinfSectionW > wine: Call from 0x7b830f70 to

How would I use validations to ensure I either received a blank value or a ten digit number? If I set validates_length_of :field, :in => 0..10 then that would allow any value in-between. The allow_nil => true option does not seem to work. I''ve also tried using this with some combination of validates_numericality_of but cannot get it to work. Any help would be greatly

Im having sound problems with Ventrilo. All was working well before when i my headset mic were working because then the mic were plugged into my sound card but now im using an usb mic and now i can't get ventrilo to usb my usb mic with alsa drivers in wine but i have output sound. If i switch to OSS drivers in winecfg my mic starts working but no more sound :S. If i have both enabled i only

hello, in trying to build xen-unstable (up to changeset 9238) under x86_64 sles10 beta 7, gcc-4.1.0-3, python-2.4.2-9, i''m getting the following error during the build: building ''xs'' extension creating build/temp.linux-x86_64-2.4/xen/lowlevel/xs gcc -pthread -fno-strict-aliasing -DNDEBUG -O2 -fmessage-length=0 -Wall -D_FORTIFY_SOURCE=2 -g -O2 -f omit-frame-pointer

WUA_table<-WUA.df[,2:dim(WUA.df)[2]] WUA_discharge<-WUA.df[,1] colour_scheme<-palette(rainbow(dim(WUA_table)[2])) # Main scatterplot p1 <- ggplot(NULL, aes(WUA_discharge,WUA_table)) + geom_line() + scale_color_manual(values=colour_scheme)+ scale_x_continuous(expand = c(0, 0)) + scale_y_continuous(expand = c(0, 0)) + expand_limits(y = c(min(WUA_table) -

Hi, Samba 3.0.2a, Debian linux, 2.6.5 kernel, PDC server, WinXP clients. I'm getting the following error when I try to add an XP machine to our domain. "The specified domain either does not exist or could not be contacted." I've applied the SIGN-OR-SEAL patch. The computer is connected through a wireless NIC to the network. I am able to ping the server. One other XP

My ISP currently runs version 1.2 and I'm looking to make use of UIDPLUS because I want to append and then move the message I've just appended to another folder so then I need the UID in order to address the new message. My ISP says they have plans to upgrade to version 1.4 soon. Do either of these versions support UIDPLUS? ie. give me the new UID as the response to an append?

tags 614101 +moreinfo thanks I'm afraid there is not much which can be done with this otherwise. Hints on setting up serial console can be found on the Xen wiki: http://wiki.xen.org/wiki/XenSerialConsole If you can please also try the Wheezy packages. Thanks, Ian.

Hi everyone, I am trying to create a image using this command and I get this message. Does anyone know which configuration I did wrong? # xen-create-image --hostname=image01 --ip=192.168.188.15 --dir=/home/cloud15/xen --dist=sarge Please choose either DHCP or static usage, not both! Thanks in advance! Felipe -- *-- -- Felipe Oliveira Gutierrez -- Felipe.o.Gutierrez@gmail.com --

> > I am looking for "how to" information / references on use of either Audiocodes MP104 or 108, or Vega 50 Gateways for interconecting Asterisk to the PSTN via FX0 interfaces. > > Any info of references / personnal experiences would be appreciated > > Stratus? THE WORLD'S MOST RELIABLE SERVERS * > Richard C. Sparacino > Telecom Technology Manager

Hey, Im running Asterisk 1.2.2 and im having problems with the audio when bridging calls between the zap interfaces and sip. zap to zap work fine, as do sip to sip (but asterisk isnt in the media stream, as it doesnt need to be) and terminating the call and playing a test message via either sip or zap work fine. Basically, the only time I see this problem is trying to bridge between sip and

On inbound calls from my SIP provider I get multiple warnings as follows: WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid host Everything else works but these warnings are a pain and I don't know what they are about.... Nothing on previos lists or Google explains... -------------- next part -------------- An HTML attachment was scrubbed... URL: