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Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion

Is anyone on the list using an Audiocodes gateway with asterisk and SIP? I'm looking at that platform, but I have a couple of issues: 1) Echo cancellation. The echo that I'm hearing with an X100P is unacceptable. Does the Audiocodes do better? 2) Line signalling. I'm using Kewlstart with the X100P, but it looks like the audiocodes uses loopstart only. How does this work with

I'm using the SNOM 200 latest SIP beta (so that I can have the GSM codec, etc). Everything seems to be working fine, but the music on hold doesn't play when I use the HOLD button on the snom. Any suggestions? Thanks, --Ernest

I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a

Is it safe to assume that a fresh CVS build will not have the SIP translation problem described? Regards, Christopher --__--__-- Message: 11 Date: Mon, 29 Sep 2003 12:45:40 -0700 To: asterisk-users@lists.digium.com From: "Ernest W. Lessenger" <ernest@oacys.com> Subject: Re: [Asterisk-Users] Is somthing broken? Reply-To: asterisk-users@lists.digium.com At 12:33 PM 9/29/2003, you

In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger: Andrew, I modified the exten line in extensions.conf as you suggested. Unfortunately, It still does not work... Ernest, I spent approx. 4 hours reading list archives (and anything else Google served up) on how to configure iax.conf and extensions.conf to work with Voicepulse. Then, I sent an email to voicepulse

Obviously, there are no DS3 TDM cards that are currently compatible with Zap channels. (or are there?) Does anyone know of an inexpensive DS3 card that could perhaps be used with Asterisk if one were to try to port the Zap drivers to such a card? PCI, of course, would be the bus of choice. I think there are quite a few discouraging comments to be made on that question. Firstly, most

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

Hello list. I'm trying to rebuild the 2.6.9.67.0.20.EL kernel, but it fails even without modifications. How did I try it? Created a (non-root) build environment (not a mock ) Installed the kernel.scr.rpm and did a rpmbuild -ba --target=`uname -m` kernel-2.6.spec 2> prep-err.log | tee prep-out.log The build failed at the end: Processing files: kernel-xenU-devel-2.6.9-67.0.20.EL Checking

I have a few questions regarding SIP and NAT that you may be able to answer. In both cases, I'm "assuming" that the customer will use SNOM phones and/or xten soft-phones. Q1: I know that it is possible to use a STUN server to handle SIP over NAT. Does this require any special configuration of the NAT router? For example, will I need to configure port forwarding? Q2: If I know

Is it possible to limit the number of "lines" provided by a given SIP/IAX connection? For example: I want to limit SIP extensions to only a single incoming line, even the phone itself can handle three. Or, I might want to prevent extensions from making more than one outgoing call at a time. Or, I might want to protect my bandwidth/call quality by limiting outgoing calls through

We are trying to implement "area-code dialing" in our asterisk PBX. Basically: we will have a number of customers, who may be in different area codes, that want to direct-dial each other's extensions. We want this to work like a "real" centrex, in that seven-digit numbers should try (1) "local" VoIP extensions, and then (2) "local" PSTN numbers.

Does anybody have a reasonable solution for an Outlook MAPI plugin that works with asterisk? At very least, I would like Asterisk to push incoming call information to the computer, which should then open an Outlook form, launch a web browser, etc. Beyond that, it would be cool to have Outlook initiate outgoing calls. Shouldn't be too difficult, and I know some of you are working along

Does anyone know of a command-line tool that I can use to mix my own MOH tracks? Specifically, I want to be able to do this: 1) Record a "Your call is valuable to us..." advertisement 2) Specify a number of song files to be played randomly/in sequence/whatever 3) Insert or overlay the advertisement every n seconds I would like this to be done live (via a configuration file) if

Is anyone using Asterisk as the gatekeeper/proxy for videophone calls? Thanks, --Ernest

Hey all, For those of you who are really worried about asterisk performance, I thought I might alert you to a "toy" you might play around with. The Intel Performance Primitives contain a number of optimized functions for use in digital signal processing that could help with echo cancellation, codec transformations, etc. I don't have any idea how useful this would be in Real

I am having some oddness with the 11/11/2003 CVS of *. Specifically, outgoing audio to NuFone doesn't seem to be transmitted (I can hear the other side just fine). My firewall is set to allow all outgoing traffic, and the IAX2 connection is definitely established correctly. Also, I can watch UDP traffic going by on the firewall so I know that * is transmitting. This happens with X-Ten on

Does anyone have the physical spec sheet for the T100P from Digium? The one on the website doesn't have what I need. Things like 3.3 or 5v operation, uses n IRQ channels, requires 32-bit PCI, must be installed while standing on one foot and reciting the GPL, etc. Also, if anyone is selling a used T100P or TE4xxP I'd like to talk... Thanks, --Ernest