All,
Does the dlink DVC-1000 work with asterisk?
On the wiki all it has is a link to ebay...
Jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
. There are no configuration parameters that can be changed...
It is all hard-coded in the firmware. Also, D-Link indicates that they have
no plans to develop open firmware for the units.
Regards,
Derek
- Original Message -
From: Jerry Geis geisj at pagestation.com
http://lists.digium.com
All,
Is it possible to have a list of IP addresses to register to?
Example: Service provider has multiple register locations, DC, Chicago,
LA etc...
If one register site is down then I should be able to register to a
different site
automatically.
Is this possible?
THanks,
Jerry
Ken,
Point to a different proxy. I had the same issue with chicago...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
All,
I am intersted in having 2 extensions for the same sip phone...
The reason is when I am using the outgoing spool file to initiate a
SendText() command
to my SIP phone I dont want voicemail to answer the call and my text
message go there.
I want my outgoing call to FAIL and then I can
When I callin to a meetme (first one) and it prompts for a PIN access if
I hangup at that point the meetme is created with 0 parties. I would think
if I hangup it should not create the meetme yet?
If I go ahead and enter the valid PIN it says I am the only in in the
meetme,
then I hangup and
I used the outgoing spool directory, added a variable like TEXT=Hello world
and going to context send_my_text. tehn send_my_text has exten =
s,1,SendText($TEXT)
Works great.
Jerry
---
Hello,
i dont get this feature, how can i send a text to a certain SIP-phone
that support this
I have had a number of occasions where asterisk stopped
working. (1.0.7)
When this occured I tried to issue an asterisk -rx "stop now"
and nothing happened.
I then killall -9 asterisk, and it stops - but mpg123 is still hung.
I then killall -9 mpg123 and it stops.
I then restart asterisk and
Is there any metric on the number of AGI's that can run
at the
same time. Shouldnt be a limit in my mind but I am thinking in
terms of system performance.
My AGI is a C program with 3 meg executable size.
Thanks,
Jerry
___
Asterisk-Users
I am connecting to a Nortel CS 1000. I can place calls out to an extension
so we are half way there. When calling into the box I get the following from
sip debug ip X.
I get dead air when calling into the box.
In my sip.conf I have a context of nortel and in extensions.conf the nortel
context
I have a SIP trunk coming into my asterisk box.
the error is Unable to create/find channel.
How do I define that a DID number is handled in a certain manner???
Asterisk cannot find how to handle an incoming call from a number.
I have in sip.conf my definition to the PBX. That seems good as I
All,
I am connecting to a CS 1000 nortel PBX. I can call out,
I have limited success with call in. I get debug traffic that a call
is coming in but I get the message Unable to create/find channel.
I was expecting that incoming calls over the trunk would
be handled from my sip definition and
I am getting the following information and asterisk
1.0.7 is not answering the call.
Any ideas?
jerry
--
Sip read: INVITE sip:2828;[EMAIL PROTECTED]:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server
SIP/2.0
From: sip:3173241052;[EMAIL
works great when it works. Seems like lately its been doing good for me.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
I am connecting to a local nortel PBX. The person setting up the PBX did
not
define a user name and password (dont ask why). So I presume my register
command
can leave off the username and password and look like:
[sip.conf]
register localpbx/5551212
I presume that is good enough then so calls
Any WORKING uses of asterisk connected to a nortel BS1000 using SIP?
I can setup asterisk fine. I have never seen a BS1000 the Nortel guy is not
being able to set it up? Anyone have setup steps?
Thanks,
Jerry
___
Asterisk-Users mailing list
Sorry wrong model CS1000 not BS1000. typo
Thanks,
Jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Was wondering if anyone has asterisk connected to a Nortel 1000
using SIP?
Did some searching on google and did not find anything.
Is it straight forward, quirks, does not work?
Thanks,
Jerry
___
Asterisk-Users mailing list
I find a need to have 2 extensions for the same phone. WIP 5000.
The reason is one extension needs voicemail the other extension does not
as I want to SendText() to that extension and I dont want voicemail to
answer and my SendText() be lost due to voicemail answering.
How is that accomplished?
I have a normal setup of calls coming in on analog lines (4 of them)
coming into
an old KX1232 pbx. I have those lines forwarded to a T1 card in the KX1232
going over to my T1 card, it then uses IAX going over to my REAL
asterisk pbx.
(these steps are there for testing and other items)
Anyway
I happen to have a WIP-5000 phone. I can send text messages to the phone
no problem. The phone also has a feature Write Msg. I use it and enter
a valid extension for a WIP-5000 phone (so It will receive the text message)
and enter some small message and hit send...
I get a 405 Method not allowed
There are no cordless devices in this situation. :)
Jerry Geis wrote:
/ I have a normal setup of calls coming in on analog lines (4 of them)
// coming into
// an old KX1232 pbx. I have those lines forwarded to a T1 card in the
// KX1232
// going over to my T1 card, it then uses IAX going over
asterisk only allows to send messages to sip phones
using the sendtext application, which can only send text when in a
call.
On 5/18/05, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
/ I happen to have a WIP-5000 phone. I can send text messages
I have a couple WIP 5000 phones, They are working great.
I can use sendtext() and send message to the phone.
However, when I select Messages menu on the phone, then
write message, I enter a phone extension, some short message
and click send. I get FAILED on the screen.
I dont see anything on the
Jim,
My (3) WIP -5000 phones work just fine with DTMF.
I setup the user.ini file to the following and of course the same in the
sip.conf.
Hope this helps. The OpenSip is also in the browser config for the phone...
[OpenSip]
*T1 = 500
*T2 = 4000
; DTMFType - 0 RTP
; DTMFType - 1 INFO
; DTMFType - 2
I am running head from apr-6-2005.
I am using Broadvoice (may or may not be an issue)
On some calls I will have like 10 or 15 seconds when I cannot hear
the other party, they can hear me, then it comes back in again and
is fine.
Just wondering if I dont have something setup correctly (and what
I am doing a sendtext() message to a phone.
It works just fine. However I am looking at the case where the phone is
off or not available...
In that off case voicemail picks up for the phone so the call is still
answered and sendtext() is still called.
How do I tell if I am getting voicemail so I
Any one using voip.net with asterisk?
Jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Is there a way to have the IAXY dialout with a call file?
I tried:
Channel: IAX2/606/5068012
Context: smvoice-dialout
Extension: smvoice
Priority: 1
RetryTime: 2
WaitTime: 20
MaxRetries: 0
And it did not work.
Is there a way to have the iaxy call out from the outgoing spool directory?
606 is the
The question is will the hitachi WIP 5000 work when crossing subnets?
Anybody doing it?
thanks,
jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
yes
from sometime after 4 to around 5:30 pm.
jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
did you do the following
service zaptel stop
service zaptel start
then run asterisk...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
I have the WIP-5000 phone. working great.
Anyone know how to put it in 12 hour mode?
The time on hte screeen is showing int 24 hour mode.
Cant get used to it...
I've looked all over the pdf files. have not found it.
Thanks,
Jerry
___
I am having the same broadvoice issue at the moment.
jerry
Is anyone else having difficulty with their Broadvoice service? When I
dial my number right now it rings either fast busy or tells me it cannot
complete the call.
I can make outgoing calls from my system through broadvoice
If there are multiple asterisk boxes in use is there a
way to "link" them
together so when the manager api command "show channels" is executed
ALL boxes are reported?
Certainly I can connect to each box and execute the command show
channels
but was just wondering if there was something
is there an easier way to ask through the manager api
what the connected channel is for a given channel.
Example: I dont know the session number for SIP/401
but I what to know what channel SIP/401 is connected to.
SIP/401 is presently something like SIP/401- type session number
and the
I have a box. 3GIG P4, 1 TDM02B card , 1 GIG ram only running asterisk
at this time.
it has SATA drives. the wctdm is on its own interrupt and yet sometimes it
misses DTMF keys. What else might be the situation? It is not running X.
It is conneted to broadvoice. However, I have another box that
All,
I added to my dialplan something like
exten = 777,1,SMS(0,s,530,Hello)
thinking this would send the Hello message to my SIP/530 phone
(Hitachi WIP 5000).
The phone is working just fine just not receiving SMS yet. I enabled SMS
on the phone.
I was just wanting to play with the SMS and just
ALL,
I am trying to call DUNDiLookup from the manager api.
Action: Command
Command: DUNDiLookup 401
I get back No such command.
Is this possible or what dont I have right? I just checked out CVS as of
3/31/05.
THanks,
Jerry
___
Asterisk-Users mailing
I am attempting to call DundiLookup from the manager API.
Action: Command
Command: dundilookup 401
I get back no such command.
I have CVS as of 3/31/05.
What am I missing so I can have the manager API return to me the
dundi lookup information?
Thanks,
jerry
I am using the manager API for "show channels".
If I have a multi line phone extenstions 510 - 515
and 510 has a call on hold and 511 has a call on hold
and I am answering 512 the manager API show channels
doesnt seem to tell me that 510 and 511 are on hold?
They are reported as Up.
How do I
I have connected the KX-1232 to asterisk with the T1 card.
Is it dissappointing though as I have not gotten any Caller id
information running over the T1.
But it does function.
Jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
CHris,
I had the exact same problem with the exact same error.
My password was entered incorrectly in context section.
The register line had the correct password. That is why you get
incoming calls. and not outgoing.
Jerry
___
Asterisk-Users mailing list
is not grabbing it.
Jerry
On Tue, 08 Mar 2005 17:34:31 -0500, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
/ SEPDEFAULT.cnf is not required nor recommended. The 7912 only uses the gkMAC file
// and the software version CP7912XXX file
//
// The gk
I have added the three lines to the sip.conf file based on the latest
changes
from broadvoice. I can receive incoming calls but cannot place any
outgoing calls.
The error I get is:
*CLI -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569
-- Attempting call on
Broadvoice/5068012
It should be "Attempting call on SIP/Broadvoice/1(area code)5068012"
Try it and see if you can place outgoing.
-Original Message-
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jerry Geis
Sen
Here is my configs. from a previous post...
Jerry
--
; Broadvoice
register = PHONE at sip.broadvoice.com
http://lists.digium.com/mailman/listinfo/asterisk-users:SECRET:PHONE at
sip.broadvoice.com http://lists.digium.com/mailman/listinfo/asterisk-users/PHONE
] On Behalf Of James Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis
-0500, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users
wrote:
/ Here is my configs. from a previous post...
//
// Jerry
//
// --
/
___
Asterisk-Users mailing list
Asterisk-Users
http://lists.digium.com/mailman/listinfo/asterisk-users
[mailto:asterisk-users-bounces at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 2:45 PM
To: asterisk-users at lists.digium.com
http://lists.digium.com/mailman
Looks like I had mistyped that long password.
so the register statement was correct but the context was NOT correct
off by 1 character in the middle.
I never susspected the password as it worked before the weekend changes.
Thought it was OK.
Thanks to everyone..
Jerry
I have 8 cisco 7912 phones. 5 are working just fine.
They boot grab the gkMAC file from the tftp server and
off to the races.
The other three phones are a no-go. I can view the tftp log and
see they are asking for the file. The file is there but these 3 phones
do not grab it.
As you can see below
SEPDEFAULT.cnf is not required nor recommended. The 7912 only uses the gkMAC file
and the software version CP7912XXX file
The gk file must be lower case..
jerry
-
Where is SEPDEFAULT.cnf ?
Caps?
/ I have 8 cisco 7912 phones. 5 are working just
I am getting a log message of
Failed to authenticate on INVITE to ...
after months of a system working. I have changed nothing...
What can cause this. I did some searching and tried setting
in sip.conf (canreinvite to both yes and no - made no difference)
by default I had no entry at all when this
Looks like this is a broadvoice issue... I have been bitten by the
changes at broadvoice.
I have searched the list and added the three magic lines:
/ username=phonenumber
// authuser=phonenumber
// secret=registration password/
But still I cannot place calls out. Calls in are working.
Any body
I am running into a problem where I have a menu and I want the
user to enter # when they are done. However doing so then asks
to transfer. How do I disable that.
Thanks,
Jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I am using the outgoing spool directory and place two files there to
call two extensions.
I have FC3 and T100P card.
example call files:
Channel: ZAP/g1/207
Context: testing-multicall
Extension: 55
Priority: 1
RetryTime: 2
WaitTime: 20
MaxRetries: 0
Channel: ZAP/g1/216
Context: testing-multicall
I am placing multiple calls in outgoing spool. (2 to be exact)
One calls my extension on a T100P.
The other calls my cell. This goes from Box A using IAX over to Box B
then connects SIP to my provider and calls my cell phone.
By now I have answered my extension - it begins speaking with automated
Here is the console screen.
Starting simple switch on Zap/1-1
Executing Dial(Zap/1-1, SIP/403) in new stack
Called 403
SIP/403-9c60 is ringing
SIP/403-9c60 answered Zap/1
Spawn extension (smvoice-incoming, 403, 1) exited nonzero on Zap/1-1
Hangup Zap/1
I have a grandstream 101
I have a grandstream 101 that is calling an extension on Zap/1 of a TDM20B.
The grandstream 101 can call another grandstream 101 at a different
extension- that works fine.
The two phones on TDM 20B can call each other.- no problem.When I call
the TDM20B Zap/1
from the grandstream phone it rings
Is the broadvoice patch part of asterisk 1.0.4?
The changelog does not mention it.
Thanks,
Jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
I am looking at interface the T100P with a NEC C2400 IPX.
Has this been done before by anyone???
quick search did not bring anything up.
Thanks,
Jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi all,
I have setup my Cisco 79XX phone. Did the tftp, put the config files in the
right location with the right names. Booted my phone, it does the tftp
things,
the screen shows my extensions everything seems fine. However, when I
come offhook and try to dial 11 which is just a playback of
All,
I have a machine with a TDM04B and a TDM13B card in it.
For over a week it was working find placing a call out every 10
minutes to the intercom system. Today that stopped working.
The call was still getting placed just as it was in the outgoing spool
and looked like everyting was working just
All,
I have FC3 fedora core 3 and just installed and compiled 2.6.10.
after rebooting I attempted to recompile zaptel-1.0.3. I did a make clean
then make. I got the following errors.
Any suggestions?
---
/usr/src/digium/zaptel-1.0.3/torisa.c:1139: warning:
Did you do a make config in the zaptel source directory?
THat works for me.
Jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
All,
I am running FC3 with asterisk. I want to use the CONSOLE/dsp in
alsa mode. This is working just fine.
I also want to be able to click on an audio file from the browser and
play the wave file. This works if I noload chan_alsa in asterisks.
However, I can only do 1 of my 2 items at a time and
I found this information elsewhere
making it /root/.asoundrc helped.
# This is .asoundrc
#
# this makes legacy OSS apps use alsa software mixing dmix
pcm.dsp0 {
type plug
slave.pcm dmix
}
# mixer0 can stay unchanged, because it isn't used anyway, I guess ? ;)
ctl.mixer0 {
type hw
If I start up 2 agi applications (by using the outgoing spool directory)
doing one at a 1 time is no problem. If I start both of them so they
are active at the same time and there is some message that plays
when the first AGI is done it executes a hangup.
It appears the second agi receives that
Additionally.
This seems to be related to the first AGI channel being CONSOLE/dsp.
The Hangup that my AGI sends at the end is sent to the wrong channel
so I get 200 result=-1 on the wrong channel.
-
If I start up 2 agi applications (by using the outgoing spool directory)
doing one at a 1 time
I have a T1 line connected. It is working great.
How do I tweek the digit timeout???
What I mean is when I manually Dial 9d5d0d68012 (from an analog
extension routed to the T1)
it does not work (where d is slight delays to look at my number to dial
and also then hit the key).
If I dial real
I place a call from an IAXY to an IAXY device. INitially the calls show
in the output of show channels. Then after a few seconds the show
channels
command shows 0 active channels even though I am still talking on the
channels.
Any ideas on this?
THanks,
Jerry
I have a couple Iaxy's and when calling out on them I dont hear ringing.
Everything else is working fine. Any ideas?
THanks,
Jerry
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
I have broadvoice working with ulaw like the example shows.
I was wondering if another codec like gsm can be used.
THanks,
Jerry
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
Is there a way to have asterisk setup in ALSA mode (which is working)
but also be able to play file.wav and hear that also.
CONSOLE/dsp is working.
If I stop asterisk and play file.wav that works.
But if asterisk is running I cannot also play a file.wav
I am not actually doing them at the same
All,
I was wondering if it is possible to use the manager api to
stop a "stream file" agi command for a channel.
Either through posting a DTMF digit to the channel or
something like that - or a cleaner way also.
My AGI cannot cancel the playing of a "Stream file" command
unless the user
I have the Cisco 7960 SIP version 7.3 phone.
When someone calls in I cannot always hear that person.
They can hear me though. (The ear piece is DEAD quite like it
is muted or something - no noise at all).
This never happens with the other 4 grandstream SIP phones I have.
Is there a problem
Sir,
I am using FC3 with no problem. I have the T1 card.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Does anyone have an example of what the setup paramters
are
connecting asterisk with cisco call manager using a quad T1 card?
I see the other examples using SIP and CCM 4.0 but we dont have 4.0 yet.
I see other examples using h323 but we are not using that.
I dont see any configuration
I am getting the following message.
Nov 16 13:38:48 NOTICE[000640]: indications.c:397
ast_unregister_indication_country: Removed default indication country
'us'
Nov 16 13:38:48 WARNING[1087277760]: acl.c:148 ast_append_ha:
sip.broadvoice.com not a valid IP
Nov 16 13:38:48
I am doing a record in my AGI. When I play it back at
that moment
on the phone it sounds just fine.
When I do a "play file.gsm" it is REAL LOW on the soundcard
When I send dial using IAX2 to another asterisk box and play it on
Zap/2 it is low also.
When I "play demo-congrats.gsm" on the
My sox version is 12.17.5 from FC3.
Jerry
Jerry Geis wrote:
I am doing a record in my AGI. When I play it back at that moment
on the phone it sounds just fine.
When I do a "play file.gsm" it is REAL LOW on the soundcard
When I send dial using IAX2 to another asterisk bo
I am needing to have multiple options to the Dial()
command.
I tried:
exten 1212,1,Dial(SIP/333,20,D(123)A(beep))
but it did not work. What format do I use for having
multiple options for the Dial command.
THanks,
jerry
___
Asterisk-Users mailing list
I am still getting a Busy message when I call in to my broadvoice
number.
Is anyone else still getting that or found a fix to it?
I can call out all I want no problem.
This seemed to start happening after the patch was applied.
Jerry Geis
I does show registered.
Host Username Refresh State
sip.broadvoice.com:5060 XXX 1184 Registered
On Mon, 2004-11-15 at 15:01, Jerry Geis wrote:
I am still getting a Busy message when I call in to my broadvoice
number.
Is anyone else still getting that or found a fix to it?
I can call
After the broadvoice patch I am getting busy messages
also on call in.
Is anyone else experiencing a lot of busy signals after this patch? ie
Broadvoice becomes disassociated with asterisk..
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Goto the cisco.com web site, register as a user then you can search
for the 7912 sip software. It is downloadable with instructions on their site.
I just did the 7960 phone yesterday this way.
Jerry
---
Hello
I have a Cisco 7912g for testing purposes. But it seems that it needs a special
Did you search for "7912 sip software" in the search
tab?
That is where I found mine.
Jerry
Hello
I did register, but I only find manuals and guides. But no software.
And when I go to Downloads-Voice Software, I only have
- Gatekeeper Transaction Message Protocol
- Cisco Voice Call Manager
Found the setup docs to convert cisco to SIP phone.
setup tftp
downloaded version 7.3 from cisco, put in /tftpboot directory.
reset the phone.
looked at the /var/log/messages and found this:
Nov 11 16:35:21 snorkel in.tftpd[4465]: RRQ from 192.168.1.85 filename
OS79XX.TXT
Nov 11 16:35:21
the file responsible for telling the
phone that a new firmware file is available.
-Chris
On Thu, 11 Nov 2004 11:42:30 -0500, Jerry Geis geisj at pagestation.com wrote:
Found the setup docs to convert cisco to SIP phone.
setup tftp
downloaded version 7.3 from cisco, put in /tftpboot directory
I just tried the tftp localhost and "get OS79XX.TXT"
it says access violation.
Here are the permissions of the files. any idea on why I'm getting
access violation?
drw-r--r-- 2 nobody nobody 4096 Nov 11 11:35 tftpboot
[EMAIL PROTECTED] tftpboot]#
[EMAIL PROTECTED] tftpboot]# ls -l
All,
I got my cisco 7960 phone to work. Two issues I had was correctly setting up
the tftp server for access and the fact that I had to FIRST use a
version 5.1.
I could not go directly to verison 7.3. So the phone is working.
I can make calls out on the phone - but I cannot call into it. I get
Somewhere on the download pages it talks about compiling the apps like
below!
tar -zxvf zaptel-1.0.0.tar.gz
cd zaptel-1.0.0; make; make install; make config; cd ..
tar -zxvf zapata-1.0.0.tar.gz
cd zapata-1.0.0; make; make install; cd ..
tar -zxvf asterisk-1.0.0.tar.gz
cd asterisk-1.0.0; make;
at this time.
--
Jerry Geis
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677
(240)282-0319 Fax
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
I have my cisco 7960 phone. It gets a DHCP address from server A.
It connects to SIP on server B. server B does NOT know the DHCP IP address
of the cisco phone.
I can call out from the cisco phone but I cannot call into it.
If I add in the /etc/hosts file on server B and entry that matches the
On my present phone system I can pickup a call that is ringing on another
phone.
How do I do this with asterisk? I searched on the wiki for pickup
and did not find anything.
Jerry
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
All,
I just put FC3 (fedora core 3) on my machine. It uses udev.
I looked at the README.udev in zaptel and put those lines
in the /etc/udev/rules.d/50-udev.rules file.
I rebooted just for grins and I still am getting the message:
Notice: Configuration file is /etc/zaptel.conf
line 147: Unable to
All,
What is a good multiline sip phone for an operator? Model and and
manufacturer.
I presume the multiline phone looks like 4 or 8 independent SIP
phones and asterisk would handle that by a call queue.
Then the operator just does her normal routine answering calls etc...
Thanks for the
ALL,
is it possible to plug a standard analog 56K modem into my
iaxy device and make a modem call out? 9600 baud call would
be fine actually. I just want to make a call out with my iAXy
device and eliminate my PSTN line.
THanks,
Jerry
___
Asterisk-Users
steve,
Thanks, do you recall what config commands you gave the modem
to drop it down and only connect at lower speeds? I'm not a modem guru.
Thanks,
Jerry
On Thu, 2004-11-04 at 13:50 -0500, Jerry Geis wrote:
/ ALL,
//
// is it possible to plug a standard analog 56K modem into my
// iaxy device
901 - 1000 of 1015 matches
Mail list logo
