Displaying 20 results from an estimated 51 matches for "callon".
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2006 Dec 05
2
Realtime question
Hello all,
I was wondering if anyone has had much experience with Realtime
Asterisk. I like the ability to setup my extensions and voicemail boxes
in MySQL, but I have a huge worry. What if MySQL crashes. I played with
rtcachefriends, but can't seem to find a way to have asterisk store the
extension information to ensure the phones will continue to work even if
MySQL has a hiccup.
Any
2006 Dec 15
2
Fast Busy Followup
So I might have a bit of a more narrow question from my earlier one.
Previous, I had been wondering what would cause a phone dialing into a
DID that connects to the asterisk box to get a fast busy.
I've noticed the following message:
chan_zap.c: Ring requested on unconfigured channel 0/1 span 2
Any idea what would give me this error? And would this cause a fast busy?
Thanks again everyone
2008 Nov 20
4
SIP to IAX2 with delayed echo
A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
having a conversation. Call quality is reported as good except for an
echo with a 3 second delay.
Most of my searches are saying echo happens only on the PSTN piece, but
there isn't one here.
Can someone point me in the right direction?
Asterisk 1.4.21.2
Under 40 users
Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM
2008 Mar 27
3
Star Wars Echo Sound
We have a location that is having a really odd issue. We have a sangoma
POTs card. We are running software echo cancellation with the card
(through asterisk) to try to eliminate some major echoing problems. I've
turned on both EC and echotrain, which seemed to have gotten rid of the
echo for the most part. However, we are now running into an issue where
the outside caller hears a star wars
2007 Apr 05
2
PRI DCHAN Errors
Hey all,
I had a user complaining of calls which were dropping mid-conversation.
I looked into the time of one of the calls, and saw the following:
Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available!
Using Primary channel 28 as D-channel anyway!
Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for
'0x82b8430', 10 retries!
Apr 4 12:13:05 WARNING[6660]
2007 Feb 08
1
Auto Answer (Paging)
I'm trying to duplicate a behavior we had with our old avaya system, and
I've come across Auto Answer (Ring Answer). However, its not quite the
same yet.
Right now, when I dial **5053, it will add the SIP header for Ring
Answer and it will call 5053. The phone auto pickups just fine. However,
we need that call to be muted. If you were to call into a meeting, we
wouldn't want them to
2008 Jan 09
2
Intercom & Paging with Polycoms
I've been able to page to a specific phone (intercom type of thing), but
I'd like to have a macro or agi that pages all phones but first checks
if their on the phone. It looked like there used to be a pageall.agi
type of script on the wiki, but that link isn't valid anymore. Does
anyone have that script, or something else that would work? I would just
do SIP/1000&SIP/1001, but
2008 Apr 04
1
rxfax issue
...eve is a
"Normal Hangup", but could be wrong. Any thoughts as to what could cause
this?
-- Accepting call from '3126290600' to '3125727758' on channel 0/1,
span 2
-- Executing [3125727758 at from-pstn:1] Macro("Zap/25-1",
"faxreceive|7758|rschall at callone.net") in new stack
-- Executing [s at macro-faxreceive:1] Answer("Zap/25-1", "") in new stack
-- Executing [s at macro-faxreceive:2] Set("Zap/25-1",
"FAXFILE=/var/spool/asterisk/fax/7758_3126290600_1207329420.10.tif") in
new stack
-- Executi...
2008 Apr 09
1
Queues +Exiting
I'm having a problem getting my queue to function as it should.
After 20 seconds or so, it should prompt the user with a message "thanks
for holding..... press # to leave a message or stay on the line to
continue holding". I set up the "context" in the queues.conf file, so if
a user presses a digit, they should be able to leave. But I get a SIP
BUSY message.
Here are my
2007 Feb 13
1
Paging Followup
Hello All,
Hoping all of you might have an additional option for me to try at this
point. :)
My Goal:
To have a paging option that does the following.... When I press **_XXXX
it will send a ring-answer page to that person. The person on the other
end should be muted, so if they are in a conference, you can't hear what
is going on in the meeting. If that person hears me and decides they
want
2007 Feb 19
2
Transfer Caller ID
I'm sure this was asked before, but I can't seem to make this work...
If a customer dials one of our DIDs, and the operator transfers that
call to another employee, the Caller ID doesn't seem to do what I would
expect it to. I would expect it to show the original caller's ID.
Example:
John calls in from the outside using (213-555-1234) and he calls into
the asterisk system
2007 Jun 12
2
Softphone behind NAT issues
We are trying to use a softphone from a location that is behind a
firewall. We are using x-lite as the softphone.
So far, we've been able to get the phone to register with the asterisk
server, and it can receive voice from the asterisk server (IE,
voicemail, etc).
However, asterisk can't hear anything from the softphone. We have used 2
different machines to test this on. We are watching
2006 Dec 13
2
Realtime +Mysql +Failover
Hoping someone out there has run into this or has some ideas for us.
We currently have asterisk set up with Realtime (using mysql) for its
extensions,sip and voicemail files.
The problem we are trying to solve, is one of a failover mechanism. What
if our mysql server went down.
Can Realtime be set up with a secondary mysql server to get its data
from. We can set up mysql to sync with its fellow
2007 Mar 09
0
(no subject)
...onToMarker
(addInfoWindowTabsToMarker(new GMarker(new GLatLng
(47.6666508055,-2.9833339433),{title : "gere668",icon :
icon_contact}),[new GInfoWindowTab("Membre","pas de nom<br /
>gere668<br />Membre depuis : Ven, 03 Mar 2007 15:57:27 +0000<br /
>m.callon at recolocal.com")],{}),"gere668"),addDescriptionToMarker
(addInfoWindowTabsToMarker(new GMarker(new GLatLng
(47.6666508055,-2.9833339433),{title : "glefe726",icon :
icon_contact}),[new GInfoWindowTab("Membre","pas de nom<br /
>glefe726<br />...
2007 Mar 09
0
Clusterer
...onToMarker
(addInfoWindowTabsToMarker(new GMarker(new GLatLng
(47.6666508055,-2.9833339433),{title : "gere668",icon :
icon_contact}),[new GInfoWindowTab("Membre","pas de nom<br /
>gere668<br />Membre depuis : Ven, 03 Mar 2007 15:57:27 +0000<br /
>m.callon at recolocal.com")],{}),"gere668"),addDescriptionToMarker
(addInfoWindowTabsToMarker(new GMarker(new GLatLng
(47.6666508055,-2.9833339433),{title : "glefe726",icon :
icon_contact}),[new GInfoWindowTab("Membre","pas de nom<br /
>glefe726<br />...
2007 Dec 11
1
Asterisk not sending 200 OK
We're trying to get a SIP peer going between our asterisk box and our
provider. It should then ring our phone.
The call does come in and it does execute the extension in the dial
plan. But the provider says they never get a 200 OK back and therefore
they send another INVITE and then after a few seconds drop the call.
Here's our setup:
sip.conf
[ngt-trunk]
type=peer
qualify=yes
port=5060
2010 Aug 02
5
Asterisk and TV media server
Hello,
I would like to know whether there is a way to associate a TV media server
with Asterisk. Is it possible to access TV Chanels in the Telephone Sets.
Anybody have any tips or documents related to this please let me know.
Thanks
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2007 Jan 04
2
[Fwd: PRI Problems]
<Correction in my zapata.conf file I used>
Hey Everyone,
So this is a problem I've been having for sometime now. I sent a few
messages to the list with no luck.
The problem is that when people dial into the Asterisk system using DID
numbers, it works the first time or 2, then I get busy signals.
A friend recommended I clear out the zapata and zaptel, start over, and
recreate my
2007 Jan 18
1
Sip Phone CID
This might sound like an odd question.... but here it is anyways...
We currently have Polycom 501 phones. We have Asterisk with
Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone
dials another, the receiving end does in fact see the callers ID. But...
our old phone system set the caller id on the senders phone to show who
they called.
Example...
If Sally calls Jim, then
2007 Jan 30
1
Queue Dial Plan
A question about Queues and Dial Plans....
We are trying to set up a customer service queue. I've set up the queue
and agents who will participate. However, there's still one area I'm not
sure how to make it work. After 60 seconds, I need it to decide that no
one is available, and forward it to an email box of my choosing. Is this
possible?
Rob