similar to: Asterisk/Polycom Dialed Party Name

Hi, I am trying to use an Avaya 4602 phone, which I just updated from a very old SIP software to the latest I could find on avaya's site (032207). The upgrade went fine and it gets registered on the Asterisk server. Now, a couple of glitches, though. - The phone's web server is not working ... so I have no easy way to configure it. It used to work with the old release of the software. I

I recently sent this email to a user in response to a problem report of phone calls going to voicemail without the phone ringing. I'm wondering if I've covered all bases, or whether there is some logical explanation I haven't considered, and generally what others' opinions/experiences are that relate. This is an Asterisk system, of course. ------- I looked at the server logs

Hi, I just ran into what seems to be an issue on re-invites. I'm not sure if it's a bug or as designed, so I thought I'd ask the question. Here's my setup: - Asterisk 1.8.13.0 - Phone A: Polycom ip331, only allowed to use ulaw, canreinvite=yes - Phone B: Polycom ip330, only allowed to use alaw, canreinvite=yes Phone A calls the extension of phone B. After the normal call setup

Hello all, I hope someone can help me with this old Asterisk version. I have to run this version because of a custom IVR written on it. Porting it would take much too long and we'd have to hire a consultant because of all the hooks it has into Oracle databases and real-time information. We have a brand-new Avaya phone system in place and we will be cutting over to it in late March 2019.

Thanks for the reply John. About 85-90% of what this box has to do is just handle calls, but it also has options to transfer calls to the main phone system, which up to now has been another asterisk box. For example, you can hit 6 to be transferred to the Lost & Found Department. I do have allowguest set to “yes” already, but of course I also have type=peer and the other stuff for a sip

Hello, I've got a FreeBSD 5.5 box running Samba 3.0.21 and every-time I try to do a "mount -t smbfs -o username=username,password=password // server/share /mnt/folder" I get the error "smbfs: -o username=: option not supported". I've google'd this with no luck... any help in the right direction is appreciated. --- Jason Zondor

Does anyone know if avaya voip product is running linux under the hood? Thanks, /glen -- Glen Ford gford@idiom.com

Hello, I'm looking for any information on setting up VLANs to seperate the telephony network from the ordinary network. I have google'd around but haven't found a lot of information in the best way to go about this. Has anyone managed to do this successfully in conjunction with asterisks? If so, could they provide an overview of what they did and how they find it? Did the performance

I have a connection of Asterisk with Avaya by H.323 and so far everything worked well because only sent to Avaya. Now, the matter is that from Avaya will send me an IVR calls to capture credit card information, the link is active on Avaya 23 channels which is not how to configure Asterisk for those 23 simultaneous channels of Avaya's collect asterisk. Do not know if I can be with a group or

Dear sir I have setup Avaya with mediant with asterisk [sip_phone]---[ * ]---[mediant]---E1-trunk--[Avaya]---[analog_phone] This is my configuration when i call from SIP phone i got ringing sound of phone but whn i call from analog_phone behind avaya i didn't get ring sound of ring but SIP phone speaker ring why i am not able to hear ring sound from analog phone Regards

I need to add a few phones to an existing installation. They have a dozen IP430 at the moment. Does anyone feel that there are advantages to the IP330? Cost is not the major consideration as long as they're in the same range. (under $175) Michael -- Michael Graves mgraves<at>mstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:mjgraves at pixelpower.onsip.com skype mjgraves fwd

Let me add another variable into the mix...what about the Linksys SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a deal at $80 street price. Michael On Fri, 22 Feb 2008 08:28:37 -0500, Matthew Brothers wrote: > >Michael Graves wrote: >> I need to add a few phones to an existing installation. They have a >> dozen IP430 at the moment. Does anyone feel that

We are having an issue with Asterisk 1.6.1 and the MWI turning on when a user doesn't have voicemail. We see random MWI lights come on and the phone indicates a random number of messages (its been anywhere from 1-14) when a server reload is done. I just checked one user, they have no messages old or new and the phone (Polycom IP330) indicates that they have 2 messages. The user will check for

All, Since I have now (at least partially) got my 7961G phones working with Asterisk, I have temporarily moved on to try to get the expansion modules working. There doesn't seem to be much in the way of documentation here either. Does anyone have this combination working (or any 79X1) here? My goal is ultimately to do the monitoring approach. I have Google'd around, but come up

Question was ask of me today and I haven't yet google'd it... Any one know of a VB6 opensource voip project (in particular SIP) ?? Gary .

Hi, I'm looking for IBM hardware to support: 100 SIP hard phone users 10 fax machines on SIP ata's maybe later an additional 100 sip soft phones. Initially, all calls will be through PRI. Some conferencing. Don't know yet if this will even get used. Using 1.4 + ( probably business edition ) I'm looking for anyone who some experience / gotchas. I've google'd and

All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am

Hello, I have recently compiled Samba 3.0.0 Beta 3 on Debian Woody. I compiled it with LDAPSAM support. After reading and working through the PDF file that comes with Samba 3 to configure the LDAP server I get the following error messages when I do smbpasswd -a <username> sparc10:~# /usr/local/samba/bin/smbpasswd -a ******** New SMB password: Retype new SMB password: failed to add

Hello, I'm running samba 3.0.10 as a PDC with users stored in MySQL on FreeBSD 5.3. And it works great. But i have one litte hickup. My samba logs are getting flooded with entries like this on: "Connecting to database server, host: localhost, user: XXX, password: XXX, database: auth, port: 3306" Containing the actual mysql username/password, it logs this everytime samba does a

Hello R-Users! Does anyone know of a package to draw/analyze silicon wafer maps? Here are some examples http://www.java2s.com/Code/Java/Chart/JFreeChartWaferMapChartDemo.htm http://dp.pdf.com/site/products/wafermap/binmap.html I've google'd and searched CRAN with no luck. It seems possible for R, given the hexbin and hist2d graphs I saw at the Graph gallery.