similar to: IAX2 with g729 ATA Device

I am using Cisco 7960 phones and have had a request to have the receptionist phone ring on multiple phones just in case she is not around. Call pickup is the theory here but the issue is that not all the people that need to hear the ring would here the receptionist phone ring so I think I need to have a second line appearance on the phones in question so that line will ring. Can this be done

I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid="Conference Room Polycom" extensions.conf exten => 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten => 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper

I have encountered a frustrating problem with the meetme rooms and calls entering the system on the Digium analog cards. The typical scenario is: Callers on SIP phones, X-lite, Eyebeam, Cisco 7960, IAXy Callers entering the system from the PSTN via the digium Analog card (TDM400P) In the meetme room the SIP connections can all hear each other loud and clear. The PSTN people can hear

When using Xten's Eyebeam software I am noticing that I get a blocked port 5067 on my firewall. The source port obviously varies but the 6057 seems to be consistent. I have done some looking and can find any reference to what may be happening here. I am guessing I need to modify some packet filters but I would like to make sure I understand this so I can open the right port ranges. --

I have just finished setting up a new asterisk system which is basically the same as our first system. We are using 7960 phones and I used the phone config files the first installation with appropriate changes. The problem is that on the new system I get no message lights, I can't figure this out. One thing I do notice is that when I monitor the sip debug on the second system the sip

I have been trying to get multiple lines on the 7960 to work for several days. i have read all the posts I can find and have run multiple "sip debug" and have gotten no place on this. Here are the relevant section of the config files: sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson [scott1] type=friend

I have set up an auto attendant and all is working but I am bothered by a long pause when switching between menus. This pause is between 5 and 7 seconds and is quite annoying. Is there anyway to address this. One other thing I find interesting is that when I move from the main menu to the sub menu the delay is there but when I move from the sub menu to the main menu the delay is not there.

I need to be able to dial a long distance provider that uses an access number and an auth code. I would like to be able to program this so that the user can dial 8 and then the long distance number, asterisk will hopefully do everything in the middle. The sequence to accessing the provider is on my traditional phone speed dial as: * Dial local access number * Wait 5 seconds * Dial the auth

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Does anyone have any recommendations on an IAX Desktop Telephone or ATA Device. I currently have 2 of the SIPURA-841's on my local network and now I am wanting to try an IAX Device at my remote office since I think that it would be easier to configure through various routers than a SIP Device. I just started to look at the Digium IAXy Single FXS

I have Asterisk version 1.0-RC1 running on Debian Woody. I have 1 analog phone working, 2 inbound lines working, X-Lite is working. The problem that I am having is with Cisco 7960 with SIP version 7.2 software. I can make outbound calls and they work fine, I even get a notice that I have voice mail on the phone and it seems to register properly but I can seem to dial to the phone.

So far I have been used to using linux to provide simple routing from my network to others using commands such as ip route add 192.168.1.0/24 via 192.168.0.4 etc and it has all worked perfectly. I also use smoothwall GPL to provice vpn services, however I have hit on a problem and am not at all clear on the way in which to proceed. I now need to provide a route to services, the access to these

Hi, Well, I'll try to start at what I think the root of my problems are. When I do a getent group, I only get a list of the BUILTIN groups. BUILTIN+administrators BUILTIN+users But if I do a wbinfo -g, all the AD groups show up. This alone is not the overall problem, but it is creating a problem because I need getent to return the groups for logging different AD groups to different log

Here is the discovery that I have made. * I can Ping my phone w/o any problems. * Near as I can tell, I have set up the phone correctly in sip.conf * When I set up sip debug and/or sip debug ip and I reboot my phone, I do not see any activity from the server from my phone. Username Secret Accountcode Def.Context ACL NAT 1202 <password>

I am having a major dillema here, I have been trying to get my sip phone (hard phone) to communicate with the asterisk server. Below is my configuration: sip.conf [1201] type=friend username=1201 secret=<password> mailbox=1201 host=dynamic [1202] type=friend username=1202 secret=<password> mailbox=1202 host=dynamic extensions.conf exten => 1201,1,Dial(SIP/1201,9,rt) exten =>

Hi, I have used the Budgetone 102's extensively on Asterisk and found then quite reliable as long as you update the firmware. The GXP2000 is quite a mess at the moment as the current firmware does not support 3 quarters of the advertised functions and codec support is extremely limited. I have tested the unreleased latest firmware update for the GXP2000 and it's an incredible difference

Hello All, I am looking for a IAX2 analog telephone adapter, just want to ask your views on which ones are bad, good and the best. Thanks in advance, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh@imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

I am running Ubuntu 11.04 with wine 1.3.20. I am trying to install the Phantom Menace. The menu pops up to install, and takes me to the setup. When the setup loads, an error pops up: Severe: get prof failed! The setup continues like normal, but then when actually installing the files, another error pops up: Severe: file set execute failed. The progress bar jumps straight to 100%, then about 8

Any reviews/comments out there on this server? Looks solid.. But would like to know if anyone has purchased one of these before. Any other companies out there offer pre-built * servers that someone would like to comment on? http://www.thevoipconnection.com/store/catalog/product_16214_VS1trade_As terisk_Voice_Server.html -------------- next part -------------- An HTML attachment was scrubbed...

I purchased an IAX2 hardphone, X100 otherwise known as a Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, from a US retailer. I was able to configure the phone to work with my Asterisk box, except the hold and transfer buttons do not work. When you press the hold button, it rings endlessly, the transfer button, displays "transferring" but it does nothing.