[Asterisk-Users] wctdm module goes missing after a reboot - Gentoo?
Hey all I have a Gentoo system with Asterisk 1.2 installed. Its been working great, for some reason the Zaptel module for my Wildcard TDM (wctdm) seems to go missing anytime the server reboots, causing me to have to go to the Zaptel source directory and do a quick make install. This is the first Linux box Ive administered in a number of years Gentoos module stuff is a bit unfamiliar to me. Any idea what file is getting read at boot thats taking wctdm out of the modprobe path? Any help on how to solve the problem would be much appreciated. As it stands now, anytime I have to reboot the server, I have to manually login, install the module and then start Asterisk. Thanks! Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Rolling dialplan... best practice?
Erik,
This looks like a great
option! Thanks. Im wondering about two things (a bit of a
newbie I guess) and am hoping for a bit more clarity.
1.
In your example is Local
the context? Ive seen Local/ referenced in the
documentation online, but dont understand what it is. Im
sure this is a basic question, but I appreciate the help.
2.
My global variables (${E25},
etc.) are set to SIP extensions SIP/25. That being said, can
I use pattern matching to pass variables? Or, would I simply change your
Dial to something like:
exten = s,2,Dial(${E25}Local/*18*24Local/*30*28Local/*42*22)
and
then have the dial in the waiting extension be:
exten
= _*XX*XX,1,dial(SIP/${EXTEN:3})
Again, any help in
clarifying this would be awesome!
Thanks again!
Ryan
-
Date: Wed, 21 Dec 2005
09:04:26 +0100
From: Erik
[EMAIL PROTECTED]
Subject: Re:
[Asterisk-Users] Rolling dialplan... best practice?
To: Asterisk
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain;
charset=windows-1252
Ryan Booz wrote:
I have an Asterisk
system for a small office with 12 extensions. For
parts of the incoming
dialplan that go to _support_/_sales_ we have
phones ring various
people in an _additive_ fashion. Example:
- snip --
exten =
s,2,Dial(${E25}|18)
exten =
s,3,Dial(${E25}${E24}|12)
exten =
s,4,Dial(${E25}${E24}${E28}|12)
exten =
s,5,Dial(${E25}${E24}${E28}${E22}|12)
Create an waiting extension:
exten =
_*XX*XX,1,wait(${EXTEN{1:2})
exten =
_*XX*XX,1,dial($EXTEN{3:2})
Then dial using that waiting
extension:
exten =
s,2,Dial(${E25}Local/*18*${E24}Local/*30*${E28}Local/*42*${E28}Local/*56*${E22})
This wil dial all the
numbers at the same time, however eacht local number waits a bit longer before
executing the dial, hence it hunts :)
So ${E25} will ring instant,
${E24} starts ringing 18 seconds later, ${E28} starts 12 seconds after ${E24}
(timing is related to the 1st phone ringing)
Kind regards,
Erik
- snip --
This works, but I_ve
just realized that it has the unfortunate side
effect of making each
extension appearing to get one call for each
_Dial_ command.
So, ${E25} appears to get four calls if nobody
answers it and it goes
to voicemail. ${E24} three calls, etc.
Is there a better way
to do this kind of extension plan?
Thanks for any
suggestions!
Ryan Booz
Director of IT
Good Steward Software,
LLC
111 Sowers Street, Suite 400
State College, PA 16801
Phone: 877-327-3702
x.26 (814-237-3744 x.26)
Fax: 719-623-0577
Visit us at www.energycap.com
Ryan Booz
Director of IT
Good Steward Software, LLC
111 Sowers Street, Suite 400
State College, PA 16801
Phone: 877-327-3702 x.26 (814-237-3744 x.26)
Fax: 719-623-0577
Visit us at www.energycap.com
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[Asterisk-Users] Rolling dialplan... best practice?
I have an Asterisk system for a small office with 12
extensions. For parts of the incoming dialplan that go to support/sales
we have phones ring various people in an additive fashion. Example:
- snip --
exten = s,2,Dial(${E25}|18)
exten = s,3,Dial(${E25}${E24}|12)
exten = s,4,Dial(${E25}${E24}${E28}|12)
exten =
s,5,Dial(${E25}${E24}${E28}${E22}|12)
- snip --
This works, but Ive just realized that it has the
unfortunate side effect of making each extension appearing to get one call for
each Dial command. So, ${E25} appears to get four calls if
nobody answers it and it goes to voicemail. ${E24} three calls, etc.
Is there a better way to do this kind of extension plan?
Thanks for any suggestions!
Ryan Booz
Director of IT
Good Steward Software, LLC
111 Sowers Street, Suite 400
State College, PA 16801
Phone: 877-327-3702 x.26 (814-237-3744 x.26)
Fax: 719-623-0577
Visit us at www.energycap.com
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RE: [Asterisk-Users] Re: Meetme and Sipura SPA-941-badjitter/distortion
Dan, thank you for the pointer. I read through the whole thing and will potentially try this next week. I'll post back with any thoughts. Thanks! Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: Thursday, December 08, 2005 6:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: Meetme and Sipura SPA-941-badjitter/distortion It might be. I'm going to work with one of the remote users again tomorrow to see if we can get it working better. You're also right that the PSTN calls don't hear the echo, INSTEAD I hear a faint static/waves on a beach sound whenever I talk though a PSTN set through the system to this user. Pushing the packet size back to .03 makes direct calls better, but then MeetMe goes screwy again. ARG! :-) Anyone have experience with the mentioned fix at: http://bugs.digium.com/view.php?id=5374 and Asterisk 1.2? Does it make call quality difference with SIP? I read the whole thing thinking it was going to end up saying this was a 1.2 feature, but looks like it got pushed to 1.3. Thoughts? That patch and bug does help quite a few scenarios, but they won't help with this problem. MeetMe strictly assumes 20ms audio in 1.2.0. Earlier releases would and could process larger payloads, but the method used was identified as a source of increasing delay. The buffering used in 1.2.0 to send and receive audio packets from the zaptel mixing engine now drops anything past the initial 20ms. Check out http://bugs.digium.com/view.php?id=5697 for one possible fix. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion
I have a new * 1.2 server running on a dual-processor machine, 1GB of RAM, Gentoo with Linux 2.6 and a Digium TDM400 (four fxo boards) installed. Everything has been working great until we tried our first Meetme conference call yesterday. I have a total of 12 extensions. 9 of them are in the office with a direct connection to the server, all of the phones are Polycom 501s. The three remote users have the new Sipura SPA-941. I decided on this phone because of the features and it was easy to setup behind NAT (which all of these users have). Regular calls to these users work great with no issues at all. Its been wonderful. However, we had our first company conference via Meetme yesterday, and the SPA-941s sounded horrible in the conference. Very distorted, jittery sound. It was surprising and we ended up having them call in on the POTS line and come in that way and it sounded fine. So, I thought maybe it was a connection issue, but tested with one of our remote uses and have narrowed it down to the phone. If the user connects with X-lite to the conference room the sounds is great. If he then calls back with the SPA-941, the sound is horrible. Hanging up and dialing the extension directly to the SPA-941 sounds good as well. Any ideas what could be going on and how to fix it. I thought it could be a timing thing. The documentation on the Sipura phones is non-existent at the moment, so I have no idea what might be able to be changed. Id greatly appreciate any help or thoughts! Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion
The RTP packet size defaults to .03 (packet size in seconds). Changing this to .02 or .01 fixed the issue with Meetme. Anything .03 or above introduces the doppler effect in a Meetme conference. Thanks. Codec is uLaw and silence suppression was off already. Now, however, there is a (very) slight echo introduced into any calls made to this extension. So obviously the way that the phone sends packets is causing some issues. Anyone have a resource or guide to point me to on best way to debug packet transmission for good calls? Thanks so much for the quick help! Most Excellent! Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Thursday, December 08, 2005 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion I'd greatly appreciate any help or thoughts! try: RTP Packet size on SIP tab ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Meetme and Sipura SPA-941 - badjitter/distortion
It might be. I'm going to work with one of the remote users again tomorrow to see if we can get it working better. You're also right that the PSTN calls don't hear the echo, INSTEAD I hear a faint static/waves on a beach sound whenever I talk though a PSTN set through the system to this user. Pushing the packet size back to .03 makes direct calls better, but then MeetMe goes screwy again. ARG! :-) Anyone have experience with the mentioned fix at: http://bugs.digium.com/view.php?id=5374 and Asterisk 1.2? Does it make call quality difference with SIP? I read the whole thing thinking it was going to end up saying this was a 1.2 feature, but looks like it got pushed to 1.3. Thoughts? Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang S. Rupprecht Sent: Thursday, December 08, 2005 4:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Meetme and Sipura SPA-941 - badjitter/distortion Ryan Booz [EMAIL PROTECTED] writes: Now, however, there is a (very) slight echo introduced into any calls made to this extension. So obviously the way that the phone sends packets is causing some issues. Anyone have a resource or guide to point me to on best way to debug packet transmission for good calls? Are you sure the echo isn't acoustic echo from the handset itself? Its older sibling, the SPA-841 was really bad in this regard. On a purely sip call between two SPA-841's, if you bumped the earphone gain past halfway on the display the other side would invariably complain about the echo. I always wanted to fill the Sipura handset with modeling clay and see if that helped things any. (The echo was only a problem on direct sip-to-sip calls. Any calls going into the PSTN seemed to always be processed by an echo-can, so it wasn't noticed there.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
