Displaying 9 results from an estimated 9 matches for "rickaster".
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recmaster
2005 Mar 21
9
why even use SIP
I am setting up a new asterisk based call center. I just read:
http://www.voip-info.org/wiki-IAX+versus+SIP
After reading this and other google results for "IAX vs SIP" is there
any reason why i should use SIP anywhere !!
t
2005 Jul 04
1
Unresolved symbols - Zaptel 1.0.9 and Linux 2.4.31
I installed a vanilla 2.4.31 kernel from kernel.org and my system was
working great.
Then I tried upgrading zaptel to 1.0.9 and now I get unresolved symbols:
# modprobe zaptel
/lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o:
unresolved symbol proc_mkdir_R8712438a
/lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o:
unresolved symbol add_wait_queue_R93ee100c
2005 Mar 04
2
IAX Codec
I have 2 Asterisk servers connected with IAX. It's working fine I can
call an extension from one phone in an office to another phone in the
other office. The only problem I have is lagging. What codec should I
use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I
configured it to disallow all and use GSM only. In my sip config of each
phone I use disallow all and allow
2005 May 16
2
Pass variable to Authenticate?
I'm trying to figure out a way to make my own agent login, because I
don't like how the default works.
I have the login and logout working fine using the dynamic add and
remove commands, but I need to be able to create a list of users and
passwords.
I thought of a way to do it using a list of passwords, but the agent
would only ever be prompted for their password. I won't want that.
2005 Jan 28
1
MusicOnHold with no sound card?
I've been reading the wiki and mailing list archives that come up in
Google searches, and so far I haven't been able to find out how to do
this.
In my Asterisk console, when I dial an extension that is supposed to
play MusicOnHold, I get this message:
-- Executing MusicOnHold("SIP/8001-e0f0", "default") in new stack
Jan 28 14:22:10 WARNING[1927]: res_musiconhold.c:354
2005 Feb 14
1
Uptime/reliability with SER, Asterisk
Could anyone shed any light on how SER and/or Asterisk (stable branch)
has held up for them in that last while?
Are you using SER and/or * in a production environment? Do you ever
restart the software or reboot the system? How many users are
utilizing the system? How many calls per day/concurrently?
I read some uptimes and such on the mailing list from long ago, so I
was wondering what some more
2005 Mar 21
1
iLBC codec and mute issues
I tried using the iLBC codec, and whlie I like it, I ran into a
strange issue. I did a few searches on Google and haven't found anyone
with the same issue as this.
Anyhow, I was using a Plantronics analog headset and box plugged into
a Digium TDM card, dialed out over my VoIP provider's IAX channel to
the PSTN.
I was in a conference call which is running on an Avaya PBX (which
2005 Feb 04
4
HP ProLiant server for Asterisk
I'm looking at ordering a server from HP. I checked around on Google
and found in the Wiki that the ProLiant DL380 is supposed to be known
to work with *.
I'm going to get a price quote on the following setup:
HP ProLiant DL380 G4 Server w/ the following options:
Intel Xeon 3.20GHz/1MB
2GB REG PC2-3200 (2 X 1GB)
HP ProLiant Battery Backed Write Cache Enabler for SA6i
RAID 1 drive set
HP
2005 Mar 04
4
Hardphone deployment recommendation
I'm looking to purchase and deploy a bunch of hardphones for agent
use. The phones will have to register with Asterisk and/or SER,
depending on where the phones go. They need only one line, G729 codec,
and no super fancy features. Preferrably something that is easy to
provision.
I would think the BudgeTone would be good, but then I've read so many
people complaining about them, and some