similar to: match callerid against outgoing calls

Hi, Are there any resellers of phones that can take power over ethernet in Australia? All I can find for sale online is the BT-10[12], which is cheap but not featureful enough, and the Snom 190, which is about right, but neither of them support PoE. I'm particularly intereseted in the Snom 220 with the keypad expansion for our receptionist. Although, could you make a PoE split-out cable

Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: <PAGE_BEEP se.pat.ringer.13.name="Page Beep" se.pat.ringer.13.inst.1.type="chord" se.pat.ringer.13.inst.1.value="12"

I've been running into an issue where chan_agent gets stuck and all queues stop working. Here's a show channels from when it's stuck: Channel Location State Application(Data) SIP/56-be24 s@macro-stdexten:10 Ring Dial(Agent/19|50|tw) Local/*14@agentlogin *14@agentloginoff:1 Up AgentCallbackLogin() Local/*14@agentlogin *14@agentloginoff:1

Hi, Does the passdb backend = ldapsam:"ldap://ldap.daa.com.au ldap://yaminon.daa.com.au", smbpasswd syntax actually do proper failover? I have a samba 3.0.9 server on FC2 that's been overheating (our aircon failed), and the ldap server doesn't start automatically. The logs said: [2006/01/10 08:55:47, 0] lib/smbldap.c:smbldap_open_connection(678) Failed to issue the

Hi, Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone The dialplan was configured through AMP and has nothing fancy in it. As a first time user of not only Asterisk, but also a PBX, there are some operator teething problems. After much googling & searching of voip-info.org, I cannot find any answers to these

I've just noticed that if you start a call by writing a file to /var/spool/asterisk/outgoing the cdr created on termination logs the call placed to the local extension - not to the destination in the PSTN. Hence there is no record of the PSTN number dialled. I guess most people want to log the outgoing portion not the local call leg? Anyone know of a setting that changes this? Iain

If extension 101 calls 102 and user 102 hits # and then 103, the caller ID of 103's phone says 102. I've been looking for a way to have 103's Caller ID show the person that is being transferred not the person transferring. So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the

Bouncing a thread from the device drivers list: http://opensolaris.org/jive/thread.jspa?messageID=357176 Does anybody know if OpenSolaris will support this new Supermicro card, based on the Marvell 88SE6480 chipset? It''s a true PCI Express 8 port JBOD SAS/SATA controller with pricing apparently around $125. If it works with OpenSolaris it sounds pretty much perfect. --------------

I am having problems getting incoming calls from the PSTN to route to extensions, digital receptionist and even voicemail. When I call a DID number for one of the lines, it rings twice then says: "Goodbye" and hangs up. (logs to follow below configuration info). When I dial 7777 it goes to the digital receptionist without any problems. The system setup is simple; I have 8 PSTN

Here is the situation that I would like to create: Call comes in Receptionist sees that the caller ID is Jenny <8675309> Receptionist picks up phone and transfers call to Batman Batman looks at his phone and sees that the caller ID is Jenny <8675309> I can't seem to figure out how to forward the caller ID. Is this possible with Asterisk?

Hi all, Has anyone had any experience getting Polycom phones repaired? The screen on one of our IP600s got smashed, and I'm wondering if it's worth the effort to get it repaired, or if it'd just be cheaper to buy a new phone. Thanks, -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the

Hi All I have a problem with mixmonitor in 13.4.0 doing the following: 1. Caller phones in 2. Reception picks up 3. Talks to caller 4. Does attended transfer, talks to manager to screen the caller wanting to speak to him 5. Complete the transfer by putting down her handset so the caller can speak to the manager 6. Caller talks to the manager The problem is that mixmonitor only records

Is there anyone using * in AU that has successfully extracted the CLID from an incoming analogue PSTN phone call, and would like to spread the word? Note - I am only interested in analogue, not ISDN phones. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you

Hi I have an ISDN connection with 100 DIDs assigned to it... What I'm trying to achieve is set the proper outgoing callerID while showing the local caller's extension in the CDR. There is a behaviour that I just can't explain. the callerid field in sip.conf is set as : callerid="Jean-Yves/E" <300> the callerid in iax.conf is set a: callerid="Jean-Yves/E"

I've got a Polycom IP 600 that doesn't want to DHCP. It DHCPDISCOVERs, there's a DHCPOFFER, it DHCPREQUESTs and a DCHPACK is returned, but 3 seconds later it repeats the process and never boots. The phone works fine with a static IP, and the DCHP setup works ok for an IP 500. I updated to bootrom 2.6.2 and sip firmware 1.5.2 which didn't help. Any suggestions? -- James

this happened to me on a cvs update, rebuilt a clean chan capi cm and all is well. Greg ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Voicomm User Sent: Monday, September 19, 2005 3:29 AM To: Armin Schindler Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk ISDN: Problem

Is there a way for me to set my outgoing callerid string so that all callers outside of my pbx see our callerid string as company name main company number but callers inside our telephone network see extension holders name extension number? In looking at the references it looks like I can do one or the other but not both. Does anyone know how I might accomplish this? Thanks for any

Hi All, Does anyone have an example dial-plan and/or a way to use the queues to access outgoing calls on Zap channels when all the channels are congested? e.g. I have 20 users, who will be accessing Zap/g1 .. which has 3 channels I have another 5 users who will be accessing Zap/g2 .. which has only 1 channel. The outgoing numbers are dialled directly, and put through categorised extensions

Is there a way to keep the incoming CallerID from the PSTN and pass it onto the sip phone receiving the supervised call transfer? The receptionist receives the PSTN callerID, performs a supervised transfer, we get her local SIP callerID, not the original callers. The main reason we would like the true callerID is for asterisk monitor to name the file correctly for call records. Is this