With Polycom 501s, creating custom ringtones isn't hard at all.
First, grab your favorite mp3 or wav file and create a file that is
about 10 seconds long (max). If its an mp3, convert it to a wav file.
Next, use this command to ensure the wav file is properly formatted for
a Polycom phone:
sox
Here's what I'm looking to do
exten = 10,1,Dial(SIP/1000SIP/1001,15,wW)
exten = 10,2,Voicemail(u1000)
This should ring both phones and they should keep ringing for the
alloted time before moving on. However, it appears that if one of the
phones is Busy, it returns with a busy and moves on
We've experienced the same problem twice now in the past month. The
asterisk pid stops responding. We aren't able to connect to the CLI and
all calls are dropped. The lots are pretty bare as well.
This is the message log:
Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for
I'm running Asterisk 1.4.18 and having a problem with the
clearglobalvars option.
I have a NIGHT_SERVICE variable which I initially set equal to off. I
then have an extension they can dial which will toggle that variable. My
problem is when you enter the CLI and type reload, it resets to off
Running Asterisk 1.4...
We have a customer service queue which works great. The members are hard
coded (member = SIP/1000), etc. However, we have a special need. If the
queue becomes busy, we would like to be able to dial an extension and
grab only the next caller in the queue. We don't want to
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
that occurs, the EC is
kicked in and everything is fine?
--
Chris Earle
System Solutions Specialist,
Network Technologies Division
CBL Data Recovery
w: http://www.cbltech.com
Rob Schall [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
We have a location that is having
They are all connected directly to the same switch which asterisk also
connects into. Its a small office (6 people).
Rob
Anthony Francis wrote:
If the problem is specific to certian inspections I would verify the LAN
segments involved in connecting those devices.
Rob Schall wrote
Hi all,
Here's our setup:
Asterisk 1.4.18
Agx-ast-addons 1.4.5
Problem:
When accepting a fax, the fax itself comes through just fine, and it
does successfully create a tiff file. However, the dialplan should be
executing a system command right after that completes, but isn't due to
hanging up
)} ${FAXFILE} ${ARG2})
exten = s,n,Hangup()
exten = h,1,System(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE} ${ARG2})
exten = h,n,Hangup()
Doug Lytle wrote:
Rob Schall wrote:
Hi all,
Here's our setup:
Asterisk 1.4.18
Agx-ast-addons 1.4.5
Problem:
When accepting
,Hangup()
exten = h,1,System(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE} ${ARG2})
exten = h,n,NoOP(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE})
exten = h,n,Hangup()
Doug Lytle wrote:
Rob Schall wrote:
exten = s,n,Hangup()
exten = h,1,System
Doug Lytle wrote:
Rob Schall wrote:
exten = h,1,System(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE}
Take the h extension out of the macro.
Doug
The problem is that I need the macro so I can make it mobile. I reuse
the macro about 20 times
Nevermind. I'm a dummy. That method does work, but I didn't have a
${email} blah. Long week...
Thanks again for the help,
Rob
Rob Schall wrote:
Doug Lytle wrote:
Rob Schall wrote:
exten = h,1,System(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE
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,
That fixed it. I always thought the s would be the fall back from all
extensions that didn't match. I guess that doesn't work in this case.
Thanks!
Rob
Guido Hecken wrote:
-Ursprüngliche Nachricht-
Von: Rob Schall [mailto:[EMAIL PROTECTED]
Gesendet: Mittwoch, 9. April 2008 15:50
We've just upgraded to asterisk 1.4 and we have changed the way we
handle our calls a bit. This seems to be giving us a bit of an issue.
We now allow the phones to reinvite. In the rtp.conf file, i've set the
range from 1-2. However, when the phones begin talking to one
another, they
If the voicemail portion is reached, but hungup on, the extapp portion
of the config file is still executed. So you could have an external
app which does any number of things (IM, etc).
Rob
Jean-Marc Salsa wrote:
Hi,
First, sorry to repost, As I didn't get any replies, maybe this time,
I
is the number of message to trigger such a
notification ... but 0 is the number of message as well when you erase
all your messages, so you shouldn't send a notification in that case.
Any idea please ?
Thanks,
JM
On 4/16/07, *Rob Schall* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote
That defiantly makes sense. And it probably could be one less step than
mine as well. :)
Enrico Pasqualotto wrote:
Rob Schall wrote:
One easy way to get close to this affect:
Create a group dialDial(SIP/1000SIP/1001)
then have a dynamic meetme room generating extension. This way
One easy way to get close to this affect:
Create a group dialDial(SIP/1000SIP/1001)
then have a dynamic meetme room generating extension. This way, you can
put them on hold for a brief second, dial that extension, create a room,
then transfer them into it. This keeps the number of conference
the cdr analyzer should work for most of what you need. The call costs
will be the hard part. If you know how much each type of call should
cost (based on destination number, location, etc), then you could do the
math on your own. But if you don't, then you'll have to wait for your
provider to
Hello all,
I currently have all outgoing calls set to mask the caller id so it will
always appear to be coming from our main number. The problem I'm having
though, is with both the call detail in mysql and with the automon
(recording) feature. It shows the originating number as the number I
that would even work. This seems like something most companies
would want to have working. How would your average company handle this?
Rob
Alex Balashov wrote:
On Thu, 19 Apr 2007, Rob Schall said something to this effect:
Hello all,
I currently have all outgoing calls set to mask the caller id so
My problem is this
We have a location outside of our network which is done over vpn.
Everything works except for the voice quality to that location isn't
very good. To try to resolve this, I wanted to try to make all calls go
over gsm. Right now, when i say show sip channels, they all show
I was in the asterisk console and I typed reload. Is this not enough
to reload the sip.conf file?
Rob
Andreas Sikkema wrote:
However, even once I reloaded the extensions, its still only
using ulaw.
You didn't reload the sip config? Maybe that's your problem?
]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Schall
*Sent:* Tuesday, May 01, 2007 2:06 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Calls in ulaw, not gsm as desired
I was in the asterisk console and I typed reload. Is this not enough
situation?
Rob
Ed Nuñez wrote:
Reload will reload your sip.conf file! As well as iax.conf,
extensions.conf, queues.conf, voicemail.conf, users.conf
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Schall
*Sent:* Tuesday, May 01, 2007 2:06 PM
. .
On 02/05/07, Rob Schall [EMAIL PROTECTED] wrote:
So I reloaded things and had just gsm set for 2 of my polycom 501
phones. However, the logs say No codec found, which leads me to
believe that polycom 501 phones can't use gsm. Does anyone have
something like this working? If not gsm
I'm not sure if this is a problem with our polycom 501 phones or with a
setting in asterisk.
When you set the forward option on the phone and have it point to an
outside number (a cell phone) we see the following problem... The call
does forward, but while its doing so and while its ringing,
With our current setup, we have an older avaya system which is linked
with our asterisk system via a em wink connection. When you press 2 on
the avaya network, it will jump to our asterisk box and then sends DTMF
digits. Asterisk listens for those numbers and then responses as soon as
it has a
Hey all,
We're starting to see all circuits are busy and a few dropped calls.
When these happen, in the messages log, I see the following error.
May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or
TOK_LP or
Hello all,
Normally I just use pri's with our asterisk systems, but a request came
in to add some normal pots lines to the setup. We have 3 lines, and they
run into the fxs ports. They hit the dialplan just fine, and they always
dial the s extension. However, my question would be... Is there a
lines. Then the line fees balance out.
I was hoping for a solution more along the lines of Use this x
variable that contains what ZAP channel it came in on, then I can
program that one to point to a particular person.
Thanks,
Rob
Sean M. Pappalardo wrote:
Rob Schall wrote:
Normally I just
you don't care which number was dialed, you just want
failover at the telco.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: Wednesday, May 23, 2007 8:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Well, asterisk was working without flaws until just a few minutes ago.
Asterisk stayed running, but we were just getting dead air when you'd
pick up the phone and you defiantly couldn't send/receive calls. I
looked at the call detail, and there isn't anything abnormal there. I
then looked at the
I too have this problem. I have two queues set up, and one is in use. I
didn't realize thats what caused those errors. I am also using sip.
Here are my setups if it helps anyone find a bug:
Queues.conf
[billing]
music=default
strategy=ringall
reportholdtime = no
timeout=8
retry=10
wrapuptime=10
If all you need is a soft reboot to load config files and want to do it
remotely, there is no need to cut power to the phones. I know this works
on polycom 501 and 601s. I assume it would work on other polycoms as well.
asterisk -rx 'sip notify polycom-check-cfg 192.168.1.XXX '
There is a
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent: Thursday, May 31, 2007 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] reset Polycom phones remotely
Rob Schall wrote:
If all you need
A simple question but one I can't seem to find easily...
I have 90 or so DIDs. For all outbound calls, I edit the callerid so
that it will always read out main line's number. This poses a problem
though, because the CDR detail isn't written until after everything is
done. So when you look at the
Are you able to access the phone via a web browser? And did asterisk
register the phone? If both are true and you set the always reboot flag
to 1, then rebooted the phone by hand, there shouldn't be anything
standing in the way.
Rob
Stephen Bosch wrote:
Rob Schall wrote:
Correct. Once
We have the same problem with our system. Unless you have a solid (not
just high speed) connection between the 2 parties, you're going to get
silence a few times during the call. We had set up a user on a business
comcast high-speed, thinking that would be more than enough. Turned out
though, with
on their local network, and then calls go through the
PRI. There is a VoIP trunk too only for long distance, and same
problem happens there. So I was thinking its the network issue.
On 6/1/07, *Rob Schall* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
We have the same problem with our system
file and the mysql database, the src and clid
are set to our main line's number. I need to to read the correct src. I
don't care what the clid says.
Rob
Steve Murphy wrote:
On Thu, 2007-05-31 at 15:44 -0500, Rob Schall wrote:
A simple question but one I can't seem to find easily...
I have 90
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple asterisk
setup with one FXO/FXS card and basic static extensions file, etc. When
we dial out over the pots line, 4 out of 5 times, it will work. However,
every 4 or 5
wrote:
Rob Schall wrote:
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple
asterisk setup with one FXO/FXS card and basic static extensions
file, etc. When we dial out over the pots line, 4 out of 5 times
Also, in the dial command the w says its for the *1 recording. Not
waiting. Is the documentation wrong? What is the correct way to wait in
the dial command?
Rob
John Novack wrote:
Rob Schall wrote:
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card
= _91NXXNXX,n,Dial(${TRUNK}/w${EXTEN:${TRUNKMSD}},,wW)
Rob Schall wrote:
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple asterisk
setup with one FXO/FXS card and basic static extensions file, etc. When
we dial out
In the logs, does that phone try to re-register itself, or does it just
give up?
If its not trying to re-register, you might want to look at the
Expires, Register and Retry settings in the phone.
Rob
Laurent CARON wrote:
Hi,
One of my users is in trouble with his polycom phone hooked to an
CARON wrote:
Rob Schall wrote:
In the logs, does that phone try to re-register itself, or does it just
give up?
The phone is giving up.
Jun 7 14:29:36 NOTICE[22015] chan_sip.c: Auto-congesting
SIP/XXYYZZAA24-08553940
Laurent
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
No luck. Still no outbound sound.
Leonardo Kamache (Gmail) wrote:
In [general] section:
externip=your_extern_ip_address
localnet=your_local_net/bits i.e. 192.168.0.0/24
Try this...
On 6/12/07, Rob Schall [EMAIL PROTECTED] wrote:
We are trying to use a softphone from a location
This is probably what we'll have to do. We wanted to try to use all SIP
though. As I read through the documentation, it seems possible though.
Not sure where i'm off.
Rob
Tim Panton wrote:
On 12 Jun 2007, at 17:53, Rob Schall wrote:
We are trying to use a softphone from a location
behavor of where to send its packets back to.
Any thoughts?
Rob
Rob Schall wrote:
This is probably what we'll have to do. We wanted to try to use all SIP
though. As I read through the documentation, it seems possible though.
Not sure where i'm off.
Rob
Tim Panton wrote:
On 12 Jun 2007
If they're polycom 501s or higher, you could have each phone use a
different homepage. Those pages could be loaded dynamically (say in php)
and then you could just store the message to display in a database.
Probably easiest to maintain and create an interface for.
Rob
Gordon Henderson wrote:
501s don't have the
microbrowser.
Rob Schall wrote:
If they're polycom 501s or higher, you could have each phone use a
different homepage. Those pages could be loaded dynamically (say in php)
and then you could just store the message to display in a database.
Probably easiest to maintain
Voip-info has some different links to packages out there for a gui based
asterisk. In my experience, I've found it much easier to tweak a
dialplan and user accounts by hand. We are using realtime/mysql for all
our voicemail/sip/extensions, and I have a small gui I made that creates
those initial
We were able to buy a linksys router and install openwrt on it with
openvpn. That router was a client to our openvpn servers at the main
office (where the asterisk box is) and it was able to route all the
traffic that way with a few extra ip table routes. A possible cheaper
solution (since a
We were having phantom calls as well. In our case, we had 2 pots line
running in our sangoma card, and when you dial out, would would wait for
whomever to pickup. If you gave up waiting an hung the phone up (we also
had 2 normal phones plugged into fxs ports), it wouldn't immediately
receive the
An alternative to this method might be to create a call file and place
it in the spool. Have it either dial and connect the caller to an
extension that plays that sound, or just execute that sound itself.
Right now, we use this functionality for a server scanner. When it
detects a particular port
me an example of a call file?
thanks
On 6/19/07, Rob Schall [EMAIL PROTECTED] wrote:
An alternative to this method might be to create a call file and place
it in the spool. Have it either dial and connect the caller to an
extension that plays that sound, or just execute that sound itself
Since its all part of a program I would do it using the AGI like
Christopher was talking about. However, I think this would be a 2
program issue. First, you would have a program that would check a
database or whatever to see who is late and make the call to the
supervisior. That call I would
In the features.conf file, under featuremap, add automon = *1
Then in extensions.conf...
[general]
DYNAMIC_FEATURES=automon ; Auto Monitor Calls by pressing *1
now if you press *1 while on a call, it will begin recording. Press *1
again and it will complete the recording.
Rob
Drew Gibson
We use broadwing and paetec for most of our pri stuff. Paetec is a bit
better with their call detail, but both seem to provide steady service.
It depends on your location, pricing, etc though.
Rob
Duracom Lists wrote:
We have an Asterisk box setup and are ready to start offering VOIP to
our
Tom,
I disagree with your argument for a number of reasons. Each of these
reasons should be more than enough to convince you I'm correct and you
should do it my way and only my way.
And for the record, VI and CLI.
Rob
Tom Rymes wrote:
On Jun 19, 2007, at 12:37 PM, Senad Jordanovic wrote:
I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql (realtime), and all of the extensions
for the phones route through the same macro (stdexten). This all works
fine until I tried to set up notify status.
On voip-info, they say do something
it .there is a fix on bugs.digium.com http://bugs.digium.com
or you can wait till next release or use asterisk 1.4.4
On 28/06/07, *Rob Schall* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work
Eric ManxPower Wieling wrote:
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql (realtime), and all of the extensions
for the phones route through the same macro (stdexten). This all works
fine until I tried to set
Anthony Francis wrote:
Eric ManxPower Wieling wrote:
Rob Schall wrote:
Eric ManxPower Wieling wrote:
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup.
The dialplan is set to work with mysql (realtime), and all
Dovid B wrote:
You are right but my concerns is the ITSP's may stop allowing it
because they don't want to get in to trouble. They may request a list
of all the DID's that I have and limit me setting my CID to the list
that I gave them.
I doubt this will ever be an issue. The telco companies
Noah Miller wrote:
Hi Eve -
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
Or, if you can have a trigger of some type. If you have say, a database,
that stores the current night service status, then you can query that
to determine if you should send the call to the after hours steps, or to
dial into the phone. Then set up another extension that the internal
people
[EMAIL PROTECTED] wrote:
I'm looking for an easy way to make asterisk perform as a basic
(broadcast)autodialer.
Basically all I want to do is give it a list of phone #'s and a
pre-recorded message and have it call each one and play the message or
leave it on the person's answering machine.
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
I set both the trustrpid and sendrpid to yes, but the calling phone
still doesn't show the caller id of the person they are calling.
Jason Fuermann wrote:
check out rpid
Mark Johnson wrote:
Rob Schall wrote:
This might sound like an odd question but here it is anyways...
We
Here is what I have in my extensions.conf file now. Trustrcid and
sendrcid are turned to yes in the conf file.
I'm not fully sure how the SIPCalledRPID works though. The example I
found seems to try and provide the stuff automatically (id and name),
but does the SIPPEER stuff even exist?
I think
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
, they'll all get the email (its a group email), but it would
be nice to have the light as well.
Rob
Lee Jenkins wrote:
Rob Schall wrote:
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
Is your agent logged into that queue to receive the calls? You can
typically say show queues to list all queues and see who is not in
use vs unavailable. If they are all unavailable, are you getting a
successful Agent Logged In message when you log that guy in?
Rob
Manish Gupta02 wrote:
Hi
Hello all,
I think Lee has given me a head start, but I'm still running in a circle
(at least i'm in the lead).
The problem is with my queues. The phones go to their own voicemail
after 5 rings.
That's about the same time I allow the phone to ring before trying
another phone in the queue. Is
voicemail).
Hope that helps.
Rob Schall wrote:
Hello all,
I think Lee has given me a head start, but I'm still running in a circle
(at least i'm in the lead).
The problem is with my queues. The phones go to their own voicemail
after 5 rings.
That's about the same time I allow the phone
We just had the oddest thing happen which worries us as new users.
We had 3 calls running on asterisk (one from sip to sip and the other to
sip to zap). It seemed for no reason, the connections just dropped and
the lines went dead. You couldn't call a phone (not even yourself). Once
I restarted
Here's an interesting issue we're facing...
We would like users to be able to use softphones from home/work and to
use their same extensions they do at work.
The first step of getting the phones to log in as their same extensions
as work is easy and works. However, on the database side, once the
That's what I would have thought. I set the timeout to be 300 secs, but
the phone never seems to re-register. We could do a group dial, but like
you said, there would be a lot of errors in the log, which we are trying
to avoid. Has anyone been able to get a polycom 501 to re-register
itself
(asterisk sets it to 0.0.0.0).
Thoughts?
Rob
Jason Fuermann wrote:
our Polycoms reregister almost immediately. I think the problem your
running into is that when the softphone is registered the polycom gets
some kind of error from asterisk which prevents it from reregistering
Rob Schall wrote
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
From what I read on the voip-info page, they did not define how to have
the phone muted when a paging call would go out. I already have the
paging (without mute) working using those same headers from that site.
But they don't cover the issue I'm trying to solve with my Polycom
Soundpoint 501s and
if you are the callee?
Rob
Steve Davies wrote:
On 2/12/07, Rob Schall [EMAIL PROTECTED] wrote:
From what I read on the voip-info page, they did not define how to have
the phone muted when a paging call would go out. I already have the
paging (without mute) working using those same headers from
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 **_
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
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
sends operators callerid
On Feb 19, 2007, at 8:55 AM, Rob Schall wrote:
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 seems like an interesting idea, but if this would be a public user
updated 411, who would ensure that was more up to date than 411. If the
numbers are off from 411, then the phone provider isn't keeping the
records properly. A customer should be notifying the phone company when
they are moving,
If you're getting a 404, I would assume it is reacting like any other
non-connection would (http, etc). Do you know if the packets are
reaching the phone, or if the phone is registering its correct IP
Address? If it is registering, but no packets are reaching it, could it
be a routing issue?
Rob
Hey all,
This should be an easy one. I have a few different queues and wanted to
set up a standard macro to handle them, so I can shrink the dial plan
down and stop having so much redundancy. But when I try to use it, i get
a no answer.
Here's what does work (non macro):
exten = 5054,1,Answer()
That was it. :) Thanks much!
A followup... well, kinda related... And not really a asterisk q.
On the polycom 501 phones... There's those 3 lines that you can setup.
Is it possible to make one of them a shortcut to the queue login/logout
extension?
Rob
Philipp Kempgen wrote:
Rob Schall wrote
An odd question...
I have asterisk running just basic sip phones and sending/receiving
calls using ZAP. The phones are polycom 501s.
When a user presses the Forward soft key and puts an external number
(a cell phone), and then someone from the inside (another extension) to
the phone which has
I wanted to try and see if I could get my Hawkings Net-Talk USB phone to
work with our asterisk setup via yakaphone. Has anyone ever tried this?
It sees the mic and speakers, but if we could get the keypad to talk
with yaka and in turn with asterisk, that would be really nice.
If there are any
This should be easy, but I can't find the right wildcard.
Right now I have
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW)
for international and for local
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:1},,wW)
The problem is if the call isn't typed in, then you press dial, we have
Yeh, I'm sure. If I watch the debug logs in *, I see each digit running
checks to see if it matches a dialplan yet. :)
Rob
McGhee, Stefano wrote:
I pick up the handset and get a dialtone. I press 9011331234567 or
something international. Before I can finish, the local
option kicks in
Not sure if this is what you mean But we have includes in our
sip,extensions and voicemail files.
;#include sip.inc
We keep them commented out only because they are a copy of what is
running in realtime. Every night the include files are generated and put
in /etc/asterisk. If MySQL were to
I also have this problem. Unsure how to fix it though.
Rob
Drew Gibson wrote:
Hi,
We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on
Debian Sarge) and the behaviour of our Call Centre queues has changed
slightly.
Before the upgrade, when a caller was waiting in the queue,
1 - 100 of 152 matches
Mail list logo
