search for: voiping

I just updated one of my stable asterisk systems to head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from '<sip:3034585725@voip.livewirenet.com;user=phone>' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable..

Hi, every one I'd like to introduce some new feature of our products. mg3000-r fxo gateway provides more feature to work with asterisk. 1.play asterisk ivr with no interuption. when the mg3000-r received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.

For the last few days I can not resolve voip-info.org from many DNS servers. It does resolve with some DNS servers but I suspect it may be related more to caching. Using the host command: host -a voip-info.org 130.179.16.23 Trying "voip-info.org" Using domain server: Name: 130.179.16.23 Address: 130.179.16.23#53 Aliases: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR,

Hi everybody: I configured my Asterisk to register to my VoIP provider, and I can make outgoing calls, but I can't receive any calls with it. I used Ethereal to sniff the activity of it, and I found something that might be causing the problem: When my provider's gateway does the "Request: INVITE mynumber@my-voip-provider.tld ..." my Asterisk asks for "Status: 407 Proxy

The documentation of how to use the 79xx series' phones and features with Asterisk is really hard to find and put together. The higher end phones like 7970 are more like converged PC+phone, a thin client to telephony and network apps. But it's really hard to target it as a development and deployment platform because the docs and techniques are so obscure. There seems to be a fair amount

Hi, I am sorry if this message has been reposted, but for some reason I am having problems with posting it. I configured asterisk and zaptel modules with fedora2. I want to be able to load the zaptel wcfxo and wcfxs modules. For now I will use only the Wildcard TDM400P card. I am able to load the modules but I cant configure them using ztcfg or zttool because the tools are compiled to use the

Hi, We're pleased have a 25-year telephony veteran with us tomorrow, Aswath Rao. Aswath maintains that "Trapezoidal VoIP is Evil". Join us and ask questions, make comments, argue about geeky details... and maybe win a Gigaset S675IP SIP/DECT g722-capable phone with an additional handset. Those of us who have these phones like them a lot. All dial in info is here: http://VUC.me -

Guys, I am new to this group and asterisk. I downloaded the free asterisk software and compiled successfully. I was able to get to CLI and type 'dial'. As usual because of sound card problem i could not hear anything. I do not have any hardware T1/E1 cards or equivalent to test out asterisk. I want to test out simple VoIP to VoIP softphone type of apps. Do you know if any available for

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E. Does the same apply to other Linksys VoIP equipment? Is there some way VoIP equipment allow themselves to be identified by requesting data from some ports?

Has anyone had any good experiences with a voip to voip bridge... where you have an incoming call on a voip line which is redirected out another voip line to a regular phone line? Whenever we do this, the connected call is kinda lagged and the quality isn't always that great. It seems to me this is just a problem with the inherent delay in the voip connections. But I was wondering if

I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my

Looking for some advanced thoughts relative to exten number assignments. We're in the planning stage for rolling out asterisk at multiple small US telco/isp operations. Their typical voip customer has had their pstn line for a looooong time and wants to keep the pstn line and number, but add voip to their existing home/soho arrangement. One approach (from a planning perspective) is to

Hello - I have a TDM400P with 2 FXO and 2 FXS modules. Feeding the FXS modules are two VOIP lines which are terminated by VOIP adapters and have regular RJ11 wires connecting to the FXS ports. Since the two different VOIP lines have different phone numbers, and I know and can tell asterisk which VOIP line is connected to which FXS port, can I cause a distinctive ring on the extensions if a call

Hello Everyone, I've been running Asterisk as our PBX for several months now, and recently I've been thinking about using one of the VoIP providers to lower our phone bill. I know that VoIP providers can supply their customers with a local number and/or virtual numbers, and then that number/account can be used with Asterisk (well, it depends on the provider and whether or not their

I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN

I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the topology: VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension | FXO ___ PSTN extension When I call a VoIP extension on that box (from a VoIP extension), voice is good. But when this box tries to bridge the call with a

Hello, I'm using LDAP as my passdb backend and I am having a problem joining my machines to the domain. I am running Samba 3.0.10: When I run: [root@core1 samba]# bin/net rpc join -U Administrator%5P4nkm3 Create of workstation account failed Unable to join domain VOIP. >From the command line, this is what appears in my smbd logs: [2005/01/27 17:14:44, 0]

My apologies for the cross-posting. If you are a business or individual providing Voice over IP services in Canada then we encourage you to read this email carefully otherwise please disregard. ----- As you are most likely aware, the CRTC has undertaken the roll of regulating VoIP services in Canada and is currently conducting hearings with the goal of putting in place regulatory requirements

The DNS for www.voip-info.org seems to be non-responsive. Is there a mirror of this invaluable resource site? Tx, Steve dig www.voip-info.org ;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server ; <<>> DiG 9.4.1-P1 <<>> www.voip-info.org ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: SERVFAIL, id: 61402 ;;