Re: [Asterisk-Users] Still can't get asterisk to play voicemail files occasionally

[EMAIL PROTECTED] wrote on 05/30/2006 02:31:45 PM: Hi all, posted this previously, but didn't get a resolution... Maybe a rewording would help. A few times a week I will get a call from a user who has a new voicemail, but they cannot play it. They go through the menus, hit 1 to play the

Re: [Asterisk-Users] Zaptel stop hangs server

[EMAIL PROTECTED] wrote on 10/26/2005 06:53:51 PM: I have two TE110P cards. If I stop the Zaptel service, the whole server hangs. I have had this issue with 1.0.7, 1.0.8 ,1.0.9 and 1.0.9.2. The server is a Dell 1750 with all unnecessary BIOS options off (USB, Serial, Second NIC, etc) It

Re: [Asterisk-Users] Re: Zaptel stop hangs server

[EMAIL PROTECTED] wrote on 10/27/2005 10:20:05 AM: I'll give it a shot. Do you compile it with Zaptel running or diasable it and reboot first? Either way should work fine... Of course you will hang one more time trying to unload the current zaptel drivers. Bill

RE: [Asterisk-Users] Dell Hardware

Mmhh nice !! So, why did Digium forbid it :)? If Dell is so bad... why is a Dell 2850 server one of the two listed on the compatibility list for ABE? http://www.digium.com/index.php?menu=product_detailcategory=softwareproduct=ABEtab=compatibility

Re: [Asterisk-Users] voicemail function

Below is my mailbox config I have tried using 9105551212 = 1234,Gary Carr,[EMAIL PROTECTED],attach=yes Should'nt that work? Shouldn't that be... 9105551212 = 1234,Gary Carr,[EMAIL PROTECTED],,attach=yes ...works for me. Bill___

[Asterisk-Users] spandsp-0.0.2 configure problem

Hi, I'm having trouble running ./configure when trying to build spandsp-0.0.2. I was able to configure and build spandsp-0.0.1 just fine. Obviously, I'm missing something simple, however, I just can't figure out what it is. Maybe I'm missing some package I'm not aware of? Here is the error

Re: [Asterisk-Users] Unattended call transfer with IAX softphone or IAXy?

Are you saying that the IAX protocol has no built-in message that initiates an attended call transfer? IAX definitely supports attended transfers... Here is the diagram your looking for: http://lists.digium.com/pipermail/asterisk-dev/2004-January/002874.html BD

Re: [Asterisk-Users] PRI messages while running

Every so often I get a raft of messages like this: -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 As far as I know, this is expected behavior. There is code to reset inactive B-channels periodically. I think the default is once an hour. I

Re: R: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?

Chris, Isn't that what fromuser and fromdomain are for? These effect the Digest line. I use these in my friend entry to authenticate with FWD as well as other providers. Bill Doll Jr Chris A. Icide [EMAIL PROTECTED] Chris A. Icide [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED]

Re: [Asterisk-Users] AArgh, * and the 7960

How does this have anything to do with this thread: Re: [Asterisk-Users] AArgh, * and the 7960. Bill Doll Jr Chris Clifton [EMAIL PROTECTED] Chris Clifton [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/19/2004 09:42 AM Please respond to [EMAIL PROTECTED] To [EMAIL PROTECTED]

Re: [Asterisk-Users] iconnect register problem

Scott, Try this... sip.conf register = username:password@sipauth.deltathree.com/ext for incoming calls to ring [sipauth.deltathree.com] type=friend secret=password username=username host=sipauth.deltathree.com context=Incoming-iConnect canreinvite=no dtmfmode=inband callerid=asreceived

Re: [Asterisk-Users] total newbie sanity check

DID is inbound only. DID will not work if you plan to use the same trunks for outbound calls. Do the Digium cards support DID? Normally DID lines require an external power supply which connects to the card. I don't remember seeing anything like that on the Digium analog cards. Distinctive

Re: [Asterisk-Users] Re: say.c compilation error

I believe that the Janus patch 2 compatible Makefile was rolled back from CVS. The current Readme int he CVS (last time I checked) is wrong. I think the Readme never got rolled back. I've been anxiously awaiting the official Janus patch 2 release of chan_h323, but have not seen it yet. Stick

Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

Do you have a jitterbuffer enabled on your inter-asterisk IAX trunks? If so, try disabling it cleared everything up for me. With jitter buffer enabled using the default settings my audio across the IAX trunk was terrible. BTW, my 7960's are running 5.3 firmware so I probably don't see the

RE: [Asterisk-Users] iconnecthere behind NAT, strange deal

Hi, I had the same issue with iConnectHere as well as FWD (authenticated). If you are running CVS -head (4/26 - 5/1) i would suggest getting the latest CVS. I worked with Mark over the weekend to resolve this bug. http://bugs.digium.com/bug_view_page.php?bug_id=0001533 Hopefully, this will

Re: [Asterisk-Users] Talking SIP to Vocal

Hi, Try latest CVS. There was an auth sip bug fixed on Saturday. http://bugs.digium.com/bug_view_page.php?bug_id=0001533 Hope this helps. Bill Doll Jr Mark Turner [EMAIL PROTECTED] Mark Turner [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/02/2004 05:24 AM Please respond to

Re: [Asterisk-Users] iconnecthere behind NAT, strange deal

Check 0001436 in the bugtracker. This was the original bug fix which broke outbound calls. Additional work was done on this bug to fix a problem with incoming calls (see marks comments at the end). Maybe you got a CVS while this was being worked on? Maybe there is still a problem? Bill Doll