similar to: Dealt with IAreaNet before?

Hi, I appreciate it if someone knows what is available for SIP web phones out there. I am interested in putting a soft phone on a website that registers with Asterisk using SIP. Then, when someone uses it, it directly calls into an asterisk call queue.. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I have a data set that looks like this; > b2 dato chr status PRRSvac PRRSsanVac PRRSsanDk PRRSdk 33 2007-12-03 090432 R?d SPF 34 2007-02-09 090432 R?d SPF+sanDK 35 2002-12-17 090432 R?d SPF+DK 36 2002-11-27 090432 R?d SPF+sanDK 37 2002-07-23

There are two ways to view weights. One is to treat them as case weights, i.e., a weight of 3 means that there were actually three identical observations in the primary data, which were collapsed to a single observation in the data frame to save space. This is the assumption of survfit. (Most readers of this list will be too young to remember when computer memory was so small that we had to

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

your dialplan, yes it looks like it's hanging up for a reason other than a dialplan issue. It definitely doesn't appear to be an issue in the callfile, since it never gets to the commands that interact with it.... Have you tried playing with the length of the wait? Even if you technically need 10 seconds, you could try a lower amount to see if the other priorities in that context

Hey all, I'm trying to create redial on busy for my users, but haven't the foggiest on how to make asterisk route depending on the status code returned over SIP (483, Busy Here?). . . anyone know how to do this? Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Gents, this concerns a CVS-HEAD downloaded today. I configured my system as I usually do, including using allowguest=yes to attempt to correct the following problem, but to no avail. When any call comes in from an external server I get this: Oct 1715:36:43 NOTICE[4040]: chan_sip.c:10774 handle_request_register: Failed to authenticate user "+16143691415" /(this is the number making the

Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca

Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail(${mailboxnr}@context) Thanks.

Hi, I would like to use the new interface style: rsync --rsh="ssh -l username -i key" <src> username@host::module, however I get "reset by peer" errors. Can someone give some suggestions, please? Do I need to provide a symlink to the /etc/rysncd.conf file from within the user's dir? Note, I have the groups set to same as user and the group exists. Where does

Has anyone had success sending faxes via a broadvoice byod account? Everything 'looks' to go as expected, but then my fax hangs up and I get a printout with Error 351. I am wondering if it is a codec issue or something. Any help will be great. Neri -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is

Hi group, anyone can explain me the exact difference between pri value in zapata.conf ? ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN If I use it, I also must use prilocaldialplan = local ? Thanks Giordano -------------- next

I listened to all the demos you showed. My ear discerns a little muffling and minor "slushiness" in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to "de-ess" the recording. Or, I could just be rather tired.

Hi, I've set up FreeWorldDialup on my asterisk server but when I dial the service numbers, I get message '486 Busy Here '. When I dial any other number, it says 'All Circuits are busy now'. What is the problem with my settings. I've followed all the instructions step by step. Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL:

We are using MFC/R2 driver successfully in at least three places in Brazil. I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite softfones, but SIPURAS and Linksys get a garbled audio, something like a "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and the Linksys

I am working with a small inn (under 50 rooms) that is next to a ski resort. The inn just had Cat5e Homeruns to each room installed, with a patch panel in the basement. Now it's my job to connect each of the those rooms to the Internet. I think I have a Cisco switch that I can do Private VLANS with, however I thought of another solution. Has anybody seen or does anybody know of a VoIP

I've been looking online for resources that may help me, but I haven't found anything that worked. I'm trying to run x as a non-root user. Any suggestions? I get the messages: Fatal server error: PAM authentication failed, cannot start X server. Perhaps you do not have console ownership? Please consult the The X.Org Foundation support at

Hi, I've recently reinstalled a Diva in my asterisk server (alongside a QuadBRI :-) to test the nice features Armin has been adding in chan_capi. The capi.conf format has changed, so my question is how do I define a deflect= statement for different incoming MSN's? I've tried to define a section for each (group of) MSN with a different "deflect". Is that correct? [DIVA1]

Wondering if anyone is willing to give me a little assistance with some firewall rules. I think what I'm looking for is fairly simple, and I've been trying to use webmin's firewall module without success. I have a web server that I'd like to open up port 80 and forward a specific port for a select number of allowed ips. That's it. Everything else is dropped. allow: port 80