similar to: I need use sip

I am new to the mailing list, but I am very interested in running my small home business office phone system using Asterisk. However, Broadvoice, a VoIP provider of choice based on my research, is not available in my area. I currently use Vonage VoIP. Their website mentions nothing about being able to link to Asterisk. I was wondering if any US subscribers have been able to configure Vonage

We've just started testing with Asterisk (CVS HEAD) and a pair of Cisco 7940G phones running the SIP 6.3 firmware. One issue that we've run into is the ability to have multiple calls ring to the phone. Our scenario is that the user is using an extension and another call comes in for that extension. We'd like to have that second call ring the second line -- the same extension is

Hello. I am trying to write an R function to plot the survival function (and associated hazard and density) for a Siler competing hazards model. This model is similar to the Gompertz-Makeham, with the addition of a juvenile component that includes two parameters---one that describes the initial infant mortality rate, and a negative exponential that describes typical mortality decline over the

Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:wsiler@education2020.com <mailto:wsiler@education2020.com>

Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone's post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler -------------- next

Hello all, For some reason I am not showing registration in SIP. Can anyone give me an idea what can cause this? asterisk1*CLI> sip show registry Host Username Refresh State -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/bd3a7577/attachment.htm

Hello. This is a follow-up to a question I posted last week. With some previous suggestions from the R-help community, I have been able to plot survival (, hazard, and density) curves using published data for Siler hazard parameters from a number of ethnographic populations. Can the function below be modified, perhaps with a "for" statement, so that multiple curves (different line

Hi everyone, I'm wondering I would accomplish the following: I want to dial several SIP extensions simultaneously, HOWEVER, for different times (say ext 10 for 15 sec and ext 11 for 30 sec), and potentially with different headers (such as ALERT_INFO) and codecs for each extension. Obviously whoever picks up first gets the call. After the longest timeout expires (30 sec in this example) I want

Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley

>From atacomm.... ________________________________ From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Thursday, May 12, 2005 2:24 PM To: Wiley Siler Subject: Re: Got a date yet? No specific release date as of yet; but, we're hoping to have a physical date soon. So far planned release is either in June or July. Right now they developers are cleaning up the echo cancellation

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

Guys. Anybody know why sometimes on some servers Asterisk shows more than once while doing a ps? [root@server2 akrall]# ps -ax|grep asterisk 20555 ? S 0:00 /bin/sh /usr/sbin/safe_asterisk 20557 ? S 0:00 asterisk -vvvg -c 20558 ? S 0:00 asterisk -vvvg -c 20560 ? S 0:00 asterisk -vvvg -c 20561 ? S 0:00 asterisk -vvvg -c 20562 ? S

A BETA firmware upgrade toasted my ATA286. It now has limited operations. It will get an IP address via DHCP and register to the last configured SIP server, but the web interface is gone as is the voice config menu. Apart from registration, there doesn't appear to be any other SIP functionality. An Ethereal dump does not show the device trying to grab a new firmware via tftp on bootup, so

Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050812/1bff12e4/attachment.htm

I have many remote locations that dial into a central server to retrieve voicemail via IAX. Outbound calls are handled as SIP calls from a Snom to a local (to them) Asterisk server that dials the main server thru IAX. I have trained them to check their voicemail via the emailed WAV file, however some of them are, how shall I put it, idiots*, and insist that they *have* to have the MWI indicator

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

Hello all, I'm having problems to make Dovecot proxy work, I configured it following dovecot's site. See my test below: It accepts login and password and then closes the connection. bastion01:~/build# telnet localhost 110 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. +OK Dovecot ready. user raphael.costa at xxxx.com.br +OK pass xxxxxx +OK Logged in.

Hi All, I have two SIP softphones(Windows Messenger) running on different subnet (Phone-1 on IP XXX.XXX.25.ABC & Phone-2 on IP XXX.XXX.15.XYZ) and my Asterisk Server is running on IP XXX.XXX.25.PQR.Because of some security issues both the subnets are completely isolated ( U cant even PING from one to other) and I want to connect Phone-1 & Phone-2 to the *. How can I proceed? Please

Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using Asterisk@Home 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. >From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley