similar to: Correct way to update Asterisk

I'm sure that this question gets asked frequently, but a quick perusal of the list archives shows that it hasn't been asked in a least a month or so, so pardon any repetition. What is the current state of asterisk on linux 2.6? I ask, because I spent yesterday giving it a whirl, and everything seems to go just fine till the very last minute. Zaptel, libpri and asterisk compile just

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety."

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hmm, My music on hold has always worked fine. But I discovered that under incoming IAX2 calls they don't get any MOH! All I could find was a comment saying let me know if you find a solution... Nor does the debugger does say: Started music on hold So it's not starting the MOH, why? I do have it configured and it does play under other

I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need to be fully indexed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of steve szmidt Sent: Monday, June 27, 2005

I have not been able to find anything that relates to this problem. The agents are using Cisco phones. Calls goes into a queue. but once an agent picks it up it cannot be transferred. However if they call directly to the agents extension it's not a problem transferring calls. It sounds like a misconfiguration but I cannot see what's wrong. Any takers? -- Steve Szmidt "They

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin

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"

Hi, I'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same with the Asterisk box, is it a good idea, is there any other solution for the SIP emote Clients. Regards. __________________________________

Great points Steve. I think the best we can do is all throw the newbies a bone ounce in a while. Redirection to the content that is relevant is enough to get most people on the path. Like you said, the hardest part is not seeing the trees for the forest. This is the whole "teach a man to fish" parable. It is pretty easy to tell someone A) How to search and where to look B) The

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I know I'm missing something obvious, but I cannot wrap my wits around this one. I've been staring at it for too long I think. Maybe it's the three am syndrom! : ) So a call comes in and my snom ends up with this entry: CALLER NAME <sip:1231231234@server.ip> under missed calls, or whatever. Now I want to just click OK and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as the

I have two problems with Asterisk and my Budgetone 100s. 1. Sometimes when outside callers dial my Budgetone it will ring twice, then the caller will receive a busy signal. This happens intermittently and I cannot pin down the circumstances of when it works and when it doesn't. Asterisk logging doesn't give me too much info when this happens: *CLI> -- Executing

Whicch version of zaptel and Zapata should I use with 1.0?

For bridging VOIP with PSTN Lines Which one is giving better performance SIP or IAX ? I am looking at a result without NAT in picture ? Can some body give details from experiance ? Also with single SIP/IAX channel can I use more than one call at a time ? Thanks Sandeep

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

Hi, We?re going to build an IVR system with a TE405P and 4 E1. We?re sure that the 120 channels will be filled by 120 simultaneous calls during peak, so we want to have the good server to manage this. We wonder a lot of things and maybe you could help us. - Are you ever build a similar system ? - What hardware is the most adapted for this type of use ? - What is the most important ?

current setup SIP phone 192.168.1.30 --> linksys wrt54g sveasoft -- INTERNET -- (xl0) Firewall (xl2:172.16.0.50)--> (em1:172.16.0.101) Asterisk problem is RTP stream not oging trouhg from * to sip and vice versa. #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as return address.... or #2 asterisk trying to get back to me as 192.168 on public internet.. got

One would assume they have better things to do as they are quite busy. I think this is just a proactive measure meaning they say you cannot do it upfront so that in the event of a problem, it was predeclared. As to the rest of the TOS, I could be wrong but I got the impression that the owner of VoipJet speaks English as a second language due to some email exchanges. If that is the case, the TOS