similar to: Long pause between menus

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 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 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

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 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 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

Hi, My setup: Solaris 9, mbox format, mailboxes NFS mounted from another S9 system, imap. I got the following assert yesterday: Jan 30 19:57:15 emerald dovecot: [ID 107833 mail.error] imap(user): file index-mail-headers.c: line 258 (index_mail_parse_header): assertion failed: (part != NULL) Jan 30 19:57:16 emerald dovecot: [ID 107833 mail.error] imap(user): file index-mail-headers.c: line

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

Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should "ignore call forward requests from queue members and do nothing when they are requested." Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding

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, everyone. I''m getting a production server ready for a site I''m working on, and I''ve been a bit stumped by a difference between the development server (running under Webrick) and the production server (running under lighttpd). Here''s the scenario: I''m working on a variation of an e-commerce system. When someone finalizes an order with our

I''m trying to implement a user verification phase in my user registration controller. So I do something like this from within my controller: Notifications.deliver_verification(@user) The method in the mailer object looks like this: def verification(user,sent_at = Time.now) @subject = ''Welcome to XXX'' @body = {} @recipients = user.email @from

The card is an ASUSCOM ISDNLink PCI (passive) and the circuit is from Qwest (in the US). I will be using this circuit only for voice (I'm doing this because of the poor quality of my POTS lines). I've compiled Hisax (as a module) into my 2.4.25 kernel, and with 'modprobe hisax type=35 protocol=4 id=hisax' I get the following: Apr 7 10:34:24 dev kernel: HiSax: Linux Driver for

Hello Everyone, I am new to Ruby. My application revolves around the user having a "Customer" object and having that customer id in the session. Right now I have a customer service which creates the customer or finds one that has been created and adds the id to the session. However, I can''t guarantee that the customer service will be the first one hit when my application

Hi all, We have already bought 30 Digium G.729 licenses to install in several machines, but Digium only has provided on key to use in registration. We suspect that the registration has assigned the whole 30 licenses to the same server. Do anyone know how to distribute the licenses among several servers? Thanks in advance. -- Arturo D?az Contact me on FWD: 870436 Skype: arturo.diaz.almagro

We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James