similar to: Polycom IP 3000 configuration

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 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 am trying to find an ATA that will provice IAX2 and g729. I have not had much luck, I am hoping someone here might have some ideas. -- Scott Henderson ============================================================================ Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com

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

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 =>

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 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 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

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.

Message: 1 Date: Mon, 14 Feb 2005 09:53:36 +1100 From: "PHP Mechanic" <oliver.bode@phpmechanic.com> Subject: [Asterisk-Users] Who makes these phones? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <08d401c5121e$dbea4750$0200a8c0@oliver> Content-Type: text/plain; format=flowed;

I'm just wondering if anyone knows the story with these... http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5721202362&ssPageName=STRK:MEWA:IT He claims they support IAX2 and SIP... but almost no history on the account selling them. I didn't see anything in the wiki about this company either.. Does anyone have any history with these phones? Thanks, Jared

I received a sample IP/Speakerphone from my friends in China today. Asterisk setup was fairly uncomplicated and I had it running as an extension on my server within a few minutes. Sounds quality of both the receiver and the speakerphone are fine (wife's opinion). Are there any tests I should run with this phone? Following are the specs: - Single line appearance - Alpha display, 2x16 chars -

Hi all, I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028. This IP phone is registered in an Asterisk PBX and I've a problem when I dialing internal number. If I dial an internal number, like for example 102, the IP phone takes 35 seconds to send the number to Asterisk; here below the debug output 192.168.0.75: first 192.168.0.75: 1 192.168.0.75: 10 192.168.0.75: 102

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.

Hi, I've got two pa1688 phones that I want to set up to communicate between branch offices without a gatekeeper. Both phones will be behind a firewall and I want to use port forwarding so the phones can communicate. I tested the phones behind a firewall on the same network segment and there were no problems at all using sip. However, I then moved the phones into situ and port forwarded

I am about to buy AT-320EE phones for use together with asterisk. Has anyone used these phones ? Are there any known problems using them using Asterisk ?

Greetings, I'm having a job getting asterisk to register with a Huawei softX3000 softswitch via SIP. I keep getting 401 Unauthorized. Funny thing is I can successfully register SJPhone, a PA1688 IP Phone as well as a WiFi Phone against the switch without *any* problems. I think it's got to be something as simple as perhaps the register string which is currently

Who else makes Analog Telephone Adapters with IAX protocol besides Digium? I've seen Farfon is advertising their unit but I'm not sure if they shipped it or not. Why is so hard for others to support this protocol in their adapters? SIP is totally unsuitable for firewall traversal for security reason. I would like to get min. two port unit. Is it possible to use IAX protocol without