On Fri, Jul 15, 2005 at 01:28:45AM -0400, Jose Raborg wrote:
Do you want to route the calls depending on the caller id? Or Do you
want to assign a DID to a SIP?
The remote SIP device will route the calls appropriately based on the
information sent to them (the *ANI*DNIS* sent as an extension),
Hi
I want to use a H264 as video conference codec, bu as I know Asterisk
does not support it, however there must be a way to include some
special codecs to be implemented in asterisk for just passthru. Does
anyone have idea about how to do that?
Regards.
Hi,
-Original Message-
So far I've gotten Asterisk to say:
-- Extension 'XX' in context 'pstn' from '' does not
exist. Rejecting call on channel 0/23, span 1
(where XX is the phone number I dialed)
So, that's a start, I guess ;)
Turns out my VoIP provider made a booh-booh... ;-)
Evert Meulie wrote:
Hi all!
Quite a mystery. The following happened when I was on holiday, and no one
else has changed any configs of either Asterisk or the Cisco's in the
building...
The situation: Incoming works fine on all
Any help?
--- Sergio Chersovani [EMAIL PROTECTED] wrote:
You have to change the sip.conf and set context=sccp
for x-lite to be
able to dial 121
http://www.voip-info.org/tiki-index.php?page=Asterisk+Dialplan+Introduction
Sergio
___
Proxy servers can do that.
Brian Capouch wrote:
A group which my school is part of wants to start using DNS SRV
records to allow email-style dialing amongst members of the group.
I have gotten the records in our zonefiles, and things work pretty
much just fine.
However, since the DNS
search for [EMAIL PROTECTED] It works well and is very easy to install for
beginners like me.
Michael Felder wrote:
Can anybody recommend an Asterisk GUI to help a newbie confg ?
Kind regards
Michael Felder
IT Medic Australia Pty. Ltd.
P: 03 9557 2213
F: 03 9557 2214
M: 0419 568 217
E:
Javier Chia ha scritto:
Any help?
I should install [EMAIL PROTECTED] to reproduce your environment, I need
some time 'cause I'm busy rewriting the config parser.
Sergio
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I have an IVR application that works fine from multiple DID sources,
unless the call to that DID was from a Vonage service user. In this
case about half the DTMF tones never get recognized by Asterisk. Has
anyone else seen this? Suggestions? I'm running 1.0.9.
asterisk_on_oelf ha scritto:
I have found an new problem. I use a 7960+7914 and 3 lines configured for
testing, but only one line is shown on the 7960. The first button on
the 7914,
I'm rewriting the config parser, it will be configurable in the
sccp.conf in a little while.
Sergio
On Fri, Jul 15, 2005 at 08:23:14AM +0200, Florian Overkamp wrote:
Try:
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
The '.' is a wildcard match of unknown length. With your pattern you only
accept extensions of 1 digit long.
Perfect! Thank you!
Hmm...it appears it's not receiving ANI info
Dave Cotton wrote:
On Thu, 2005-07-14 at 16:31 +0200, Zoltan Szecsei wrote:
3) The alias suggestion
I did not understand this at the time I received it - Had I noticed
Tzafrir's pointer to the ethernet HOWTO, I would have realised that
alias in this context was not giving an alias to
Is [EMAIL PROTECTED] as functional as full blow Asterisk.
I am using this for my business.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel
Batista
Sent: Friday, 15 July 2005 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Asterisk shows failed to authenticate user.
This is clearly NAT related as the same user works fine inside the NAT with
no config changes
What phone? How is the server and proxy info configured? There is no
problem witht he setup assuming ports are set up properly. Sounds more
like a wrong
There is no brand on the phone.. it is from china.
[EMAIL PROTECTED] wrote:
On 7/15/2005, Bill Wong [EMAIL PROTECTED] wrote:
Hi,
I tested asterisk server with Xpro program, and all the function working
well ( like 3 way calling, transfer ). But on the VOIP phone, I
don't know
when i display in extensions.conf this :
exten = s,1,Answer
exten = s,n,DeadAGI(astcc.agi,${CALLERIDNUM})
exten = s,n,hangup
when a call comes the zap doesnot rerad the callerid
an give me this:
Jul 15 10:45:24 WARNING[6693]: chan_zap.c:5739
ss_thread: CallerID returned with error on channel
Guys.. I have a problem with 2 asterisks connected together but each time
one tries to call another I get this:
-- Executing Dial(SIP/fozy-dfbc, IAX2/voip-gw/201|60|mwtWT) in new
stack
-- Called voip-gw/201
-- Started music on hold, class 'default', on SIP/fozy-dfbc
-- Stopped
Hello,
Im trying Meet Me Feature. I read wiki , searched
google and i configured my extension.conf and meetme.conf. But I receive this
is not a valid conference number, please try again message, so what
could be the problem?
Thanks for your interest.
Erdem HAKI
Hey,
For the bridge issue, take a look at 'notransfer=yes' option in your
iax.conf.
It'll force * to stay in the path
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html
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Yes there is,
Record the user name in the voicemail setup, that's option 0 (zero) then
option 3.
Once you have recorded the users name with a voice prompt the directory
will use that recording instead of spelling the name.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hello
I am trying to get Asterisk to work with the
Junghanns Quad BRI ISDN card. I am progressing slowly!
Problem I am now experiencing is as
below.
gcc -pipe
-Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations
-g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6
Hi Peter,
Thanks for your reply.
Would srx show ccmsgs 1 help ?
Regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
Please send a Picture.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Wong
Sent: Friday, July 15, 2005 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Phone manual..
There is no brand on the
On Fri, 15 Jul 2005, David Wilson wrote:
Thanks for your reply.
Would srx show ccmsgs 1 help ?
I am not familiar with the Sirrix line of BRI cards. However, someone else
on the list may be, or you may be able to diagnose the problem yourself.
Peter
You might want to try this group out: http://groups.yahoo.com/group/pa1688/
Most of these Chinese phones are using the pa1688.
Cheers,
Storm.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Wong
Sent: Friday, July 15, 2005 12:34 AM
To: Asterisk
Title: Asterisk+errision PBX
Dear All,
I have a setup where three errision PBX's (head office 2 branch offices ) are @ three different location. Head of has a errision BP250 where the branch officers has Errision BP50's.
I wana connect all three PBX's through IP link's. My idea is to
The lights are supposed to go off
They will only come on if you have configured the span on in the
/etc/zaptel.conf
Sean
-Original Message-
From: Chee Foong [mailto:[EMAIL PROTECTED]
Sent: 15 July 2005 02:28
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] LED went off
Kevin P. Fleming wrote:
Joseph wrote:
This says the latest stable version is 1.0.7...
http://www.voip-info.org/tiki-index.php?page=Asterisk+Download
But it look like the latest stable version is 1.0.9.
Am I missing something?
Yes, nobody has updated the page, but everyone has the ability
We are starting to provide outsourcing services for our clients where we
make the outbound calls on behalf of the client.
Our clients want us to use 0800, 0845 and 0870 non-geographical numbers
for contact.
BT have advised me that we can only have one presentation number per
isdn-32
Periodically I will get this type of message in the * log:
WARNING[18535]: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Non-critical
Response)
The ip address listed sometimes is the * box itself and sometimes will
be a sip cisco sip phone.
When this happens often
Although the ipmid.cfg has been deprecated with SIP v1.5.2 (all the
parameters have been moved to sip.cfg), the firmware will still parse and
use the ipmid.cfg file until you specifically update your existing
configuration files.
If you have already updated the configuration files, then both of
Ill look into it and check the wiki for examples.
Thx!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Vincent Luba
|Sent: Viernes, 15 de Julio de 2005 03:19 a.m.
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] RE: 2 asterisks
I resolved this - I'm in the UK, and the problem was due to the cable (you need a two wire RJ cable) - I replaced it and it worked fine.
Thanks
On 7/13/05, Luki [EMAIL PROTECTED] wrote:
John,all this ringing makes me think that your PSTN Ring Timeout is too
low. Increase it by a second or two and
Title: OT (kinda): Justification for adding Asterisk to the business plan
Greetings all,
I'm trying to build a justification case to get the firm I work for to start working with Asterisk more. How could I build this case?
The argument I'm raising is that people need phones. PBX systems
I want to make sure that RTP is not going thru my asterisk.
I read you should avoid in the dial commands:
m music while ringing
t,T transfer calls from caller and called party
What else do I need to take care?
remote phone === registered to local asterisk === calling remote gateway
On Fri, 2005-07-15 at 04:17 -0700, /dev/null wrote:
I'm trying to build a justification case to get the firm I work for to
start working with Asterisk more. How could I build this case?
The argument I'm raising is that people need phones. PBX systems are
too expensive for fewer options and
Hello
Haki
I
fixed this problem following the instructions in
/usr/src/zaptel-1.0.9/README.udev.
Regards
Cecília
-Mensagem original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]Em nome de Erdem
HAKIEnviada em: sexta-feira, 15 de julho de 2005
05:11Para: 'Asterisk Users
Michael Graves wrote:
I have a number of Polycom phones to setup with my * server. For my
initial few phones I hand wrote configs. Does anyone here who uses
Polycom phones have some form of management utility for automating
their setup?
I wrote myself a very simple script that makes
Hi,
I´ve been following some information about the latest
chan_sccp drivers for a few weeks. I installed and tried
about any variant of driver that exists in the chan_sccp system.
Despite a few little changes in the CLI options I am still
encountering a lot of problems with the phone.
When
On Fri, 2005-07-15 at 06:37 -0400, Joseph wrote:
Periodically I will get this type of message in the * log:
WARNING[18535]: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Non-critical
Response)
The ip address listed sometimes is the * box itself and sometimes will
Ive tried both with
and without Answer as the first line, same result. When I was searching
through the archives, I believe there was a post from Steve Underwood that said
to always use Answer as the first line.
---
Yes, the permissions
The fact that Asterisk is soft and you're trying to sell to
an IT Company..
Just to clarify, we make up the IT company and we'd be selling it to our
customers that may or may not be IT based. The company does run VoIP but
does not use Asterisk (using VoIP to add additional local lines in
Title: Differences between System 75 and Asterisk
I remember ATT System 75 PBX systems back in the day and was amazed with how easily everything worked and was reconfigurable on the fly. Asterisk is also approximately the same. What are some of the differences between both units? Has anyone
On Fri, 2005-07-15 at 14:03 +0200, Armin Lediger wrote:
Hi,
I´ve been following some information about the latest
chan_sccp drivers for a few weeks. I installed and tried
about any variant of driver that exists in the chan_sccp system.
Despite a few little changes in the CLI options I
I tried, but it doesnt work. You can
see my conf files, is there a problem related to conf files? Could you check
it?
My meetme.conf file
[rooms]
conf = 1000
conf = 4000
conf = 9000
conf = 9001,123456
My extensions.conf file
exten = 9000,1,MeetMe(9000)
Thanks
Hello,
I have already configure the zaptel.conf
and ztcfg -vv shows all 124 channels are configured.
Its just the light was turn off when wct4xxp is loaded (with no error).
CCF
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Sean Lowry
Sent: Friday, July
Armin Lediger ha scritto:
When not using the phone for a while, it disconnects from
asterisk (1.0.7) after a few hours and tells on the display
what chan_sccp are you running?
Sergio
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On Wednesday 13 July 2005 18:19, jltaylor wrote:
TNT's have DS3 cards and the DS3 config is cheaper than multiple T1 config.
The Lucent MAX TNT is a true carrier class machine.
I totally agree with you, that's why I was asking why Jay was so adamant that
they're crap. We've never had issue
Hi,
I have a problem with DTMF in incoming SIP call (from pstn2voip provider).
Every phone keypress is detected by asterisk as doublepress (I
press 1 and get 11).
I connect to pstn2voip provider by SIP, allow only alaw codec.
dtmfmode=inbound.
After pressing '1' while asterisk executes WaitExten
I´ve got just one room configured.
These´s my configuration files ..
My
extensions.conf file
; Or a conference room (you'll need to edit meetme.conf to
enable this room);exten = 8600,1,Meetme(1234)
My meetme.conf
file
[rooms];; Usage is conf =
confno[,pin];conf = 1234
Hope it
helps
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Wilson Pickett
Sent: Friday, July 15, 2005 1:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * behind NAT and local subnet
Asterisk
will the lsmod list show you ztdummy modul?
if not, modprobe ztdummy
I think without a timer source meetme won't work
Erdem HAKİ wrote:
Hello,
I’m trying Meet Me Feature. I read wiki , searched google and i
configured my extension.conf and meetme.conf. But I receive “this is not
a
As for MeetMe, I do not have a Zaptel card. My kernel above
2.6. so is ztdummy required? Because I configured conf files but still doesnt
work. I think that something is missing.
Thanks
Erdem HAKI
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL
I am running asterisk CVS-HEAD from 11-22-04, and I am
noticing that when I run check the process list asterisk appears to only be
using 172megs of ram but when I run top it shows it as using 400megs of ram. Am
I missing something here or is there a potential memory leak in this revision
Hi,
Ive strange problem when Im making a call from SIP (Cisco 7960) to capi
(Fritz PCI). When I call a national number, Im hearing the ringtone when
the called party is ringing but when I call an international number, I dont
hear the ringtone and Ive a silence until the called party answers.
Title: Asterisk+errision PBX
Yes, it can be done.
You will need:
1) Asterisk
server
2) ATA devices (
recommended: Linksys PAP2-NA - 1 device for every two lines connected to
the PBX)
3) An appropriate amount
of time to become familiar with Asterisk and the ATA devices
You could use
Ipmid still is being processed, sip.cfg contained the same information.
I've removed it just to clean things up.
Setting the class to the correct value solved the problem, I can't
believe that I missed it.
Thanks,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Greetings,
I would encourage you to consider this item VERY carefully as customers
could get very irritated with Asterisk very quickly. For some context,
we just finished a 3 month rollout of Asterisk across 40 handsets and
three remote locations. While it works now, it was by far the worst
Erdem HAKİ wrote:
As for MeetMe, I do not have a Zaptel card. My kernel above 2.6. so is
ztdummy required? Because I configured conf files but still doesn’t
work. I think that something is missing.
Thanks
ztdummy is required for meet-me room timing
Doug
I did what you said, but still not working :(
[EMAIL PROTECTED] ~]# modprobe ztdummy
[EMAIL PROTECTED] ~]# lsmod
Module Size Used by
ztdummy 3924 0
md5 4161 1
ipv6 259201 20
parport_pc 28421 1
lp
C F wrote:
The IP-501 AFAIK comes shipped with SIP1.5.2 which does NOT use
ipmid.cfg. You have to get new *.cfg files for the ip-501 or the older
phones that run 1.5.2.
Sorry to tell you but that is not a correct. The IP-501 I have about 15 of
them new and they came with 1.4.2 also they do
I try to use a SIP trunk from a VOIP provider to make land to mobile
calls. If I do these from a ZAP channel, using an analogue phone, after
few seconds of silence (I don't like to generate fake [r]inging) I ear
the ringing tone from the mobile operator along with any message the
mobile
We have a strange make problem :
In file included from /usr/include/gtk-1.2/gtk/gtkarg.h:31,
from /usr/include/gtk-1.2/gtk/gtkobject.h:31,
from /usr/include/gtk-1.2/gtk/gtkaccelgroup.h:35,
from /usr/include/gtk-1.2/gtk/gtk.h:32,
Michael Felder wrote:
Is [EMAIL PROTECTED] as functional as full blow Asterisk.
I am using this for my business.
Yes I feel that the actually name should be [EMAIL PROTECTED] or SOHO.
I have a few clients running there business on it.
Mike
-Original Message-
From: [EMAIL
Did you remove the r option from your Dial line?
Hugo Begglo wrote:
Hello again everyone,
I'm having this same issue with Asterisk. Any ideas ?
Hugo
Cullin J. Wible wrote:
After all of your feedback and a discussion at Teliax we have fixed this
issues.
It appears that when dialing a PSTN
On Fri, 2005-07-15 at 16:44 +0300, Erdem HAKİ wrote:
[EMAIL PROTECTED] ~]# modprobe ztdummy
[EMAIL PROTECTED] ~]# lsmod
Module Size Used by
ztdummy 3924 0
md5 4161 1
ipv6 259201 20
parport_pc 28421 1
Hi,
As you may know, there is a problem with REGISTER between Asterisk and
Cirpack using SIP. (see
http://lists.digium.com/pipermail/asterisk-dev/2004-December/007843.html
)
I would like to know if a patch is somewhere ? I have seen one but it
was the patched file and not the diff, so I prefer
I have a Grandstream Budge Tone 100 SIP phone connected through a NAT
firewall to an Asterisk server. I successfully connected the phone via
NAT to the server but when I dial the extension to an AGI script, it
does not kill the process as soon as I hang up. As a result, the next
time I pickup, it
Hi,
I was wondering which would be the best GUI to use for Asterisk management?
astGUIclient or AMP?
Thanks.
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you can also use answer as the ring type instead of ring-answer if you just
want it to pick up.
I would keep Ring_Ans the same throughout for simplicity
exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans)
add an alert info type (say type 5)
alertInfo voIpProt.SIP.alertinfo.5.value=Ring_Ans
[EMAIL PROTECTED] ~]# modprobe ztdummy
[EMAIL PROTECTED] ~]# lsmod
Module Size Used by
ztdummy 3924 0
md5 4161 1
ipv6 259201 20
parport_pc 28421 1
lp 12489 0
parport
On Fri, Jul 15, 2005 at 10:14:39AM -0400, Ariel Batista wrote:
Michael Felder wrote:
Is [EMAIL PROTECTED] as functional as full blow Asterisk.
I am using this for my business.
Yes I feel that the actually name should be [EMAIL PROTECTED] or SOHO.
Originally it was intended to integrate
astGUIclient is not a configuration tool, it is an end-user-interface that
extends the functionality of your phone through a web browser.
We recommend AMP if you need a web-based config utility.
MATT---
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday,
I'm in search of useful ACD type statistics from the queues. Ie talk
time, ratio's, dropped calls etc.
The flat file queue_log is nice, but more useful would be the data in
Postgres or Mysql. Unfortunately the queue module does not yet
support ODBC DB logging (yet). In the meantime this
Kevin wrote:
Is the pager filed in the vm config still for the outcall destination or
where do you specify the number to call for the outcall?
Sorry.
You use notify= option in voicemail.conf:
3532 = 8711,Toni Hawkins,,,tz=central,notify=15045556389
-Original Message-
From: Eric
On Fri, Jul 15, 2005 at 10:11:46AM -0400, Andriy A. Yerofyeyev wrote:
We have a strange make problem :
In file included from /usr/include/gtk-1.2/gtk/gtkarg.h:31,
from /usr/include/gtk-1.2/gtk/gtkobject.h:31,
from
On Fri, 2005-07-15 at 17:36 +0300, [EMAIL PROTECTED] wrote:
I was wondering which would be the best GUI to use for Asterisk management?
astGUIclient or AMP?
I'd use AMP - mainly because [EMAIL PROTECTED] uses it - so the user base
and knowledge base should be bigger...
--
. . ___. .__
as it turns out if the agent (with callbacklogin) is logged into SIP/123 it
will try to dial 123
in local context.
only variable I can get so far that is passed on is callerid
If i want auto answer I set caller id to include a specific string
In local context I have a dialplan matching 123
On Fri, 15 Jul 2005, Cyrille Demaret wrote:
Hi,
Ive strange problem when Im making a call from SIP (Cisco 7960) to capi
(Fritz PCI). When I call a national number, Im hearing the ringtone when
the called party is ringing but when I call an international number, I dont
hear the ringtone
Thanks for the update. I had made that assumption after looking at the
script but checked as I can't seem to get it to call. I added the
variable to the general section, created the script, made it executable
and no call. I wait the 10 minutes and monitor the asterisk and system
messages log.
On Jul 14, 2005, at 11:20 PM, Time Bandit wrote:
Is the problem that the technology isn't mature, that the load on the
computer is too high, or simply that it doesn't work well in a poorly
designed network?
YMMV. I like the portability of a softphone, but sound may jitter
because of other
Hi!
Hi again, folks. I've been getting feedback from this list and
elsewhere that softphones are generally not considered good enough
for hardcore business use. Can someone point me to where I can find
more detail on this debate?
- you comp needs to have its speakers turned on in order
I have used the mayday2005 and easter2005-testing versions of the
chan_sccp driver for weeks on end making and receiving flawless calls
(in uLaw only) in an asterisk cvs that I think was from around february
of this year. A lot of the soft keys weren't implemented, but the main
calling
It's all about the testing before rollout.
The problem with the run of the mill IT guy (especially ones involved
in web sites) they tend to think that testing something means you try
a few calls and if it works it's all fine.
Testing isn;t beating into them in the same way it is in the
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i
don't know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
The old lock-in model of the telecom world was nasty and monopolistic and
expensive, but by jove it just worked.
Because as you've seen from this list ANYTHING and EVERYTHING can, and
does, go wrong with this technology.
Yes, true, but this is the nature of the beast. It (Asterisk) is an
Mojo with Horan Company, LLC ha scritto:
I have no 7920 to work with so I just can guess the 7920 comands :-)
Also in my experience, Asterisk masquerades as Callmanager 1, not 0.
I always see the following, but it could be my situation uniquely:
Connecting to Callmanager 0 (for like 4s)
Any idea if this applies to Mac OS X clients? We are a strictly Mac
company, and OS X's Unix core allows for preemptive multitasking. If
I am unhappy with the performance of the soft phones, I should be
able to tweak the priority of the phone so that it gets more compute
cycles.
I don't know
Add the debug option to the rxfax line
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Danz
Sent: 15 July 2005 13:13
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Re:
SpanDSP rxfax, no tiff
Ive tried
both with and without Answer as
It looks like you are running a old firmware or the locale skinny file
(SCCP-dictionary.xml) is not present in the TFTP server.
Oh, good ideas. I hadn't justified the cisco contract yet for
firmwares, but maybe I soon have :) I'll check out the dictionary first.
Thanks!
Sergio
sipsubscribe-20050715.rev779.txt enables:
* monitoring of other lines (using the 'hint' priority)
- LED off when monitored phone idle
- LED on when monitored phone busy
- LED blinking when monitored phone ringing
* display of caller id on monitoring phone
* call pickup by pressing function key
Ed,
There are two main drawbacks to the softphone, as I see it:
1.) User interface - The interface to the softphones is really less
than ideal. This includes the problem mentioned earlier about not
hearing ringing unless you have your headset on, dialing with the
mouse, not having
The script will create a file in /var/spool/asterisk/outgoing
That is the file that makes Asterisk make the call. This this file
exist when there should be a pending call? Also make sure your
externnotify= is set to the full path of the script.
Kevin wrote:
Thanks for the update. I had
I have seen various other problems with the cards not detected, but I seem to
suffer a different fate. The system boots and recognizes fine, but when I call
in from an external line, all I hear is a horrible static. The system works
fine when only 1 card is present. I have already moved the new
Ok, thank you.
It is strange that no body have installed any cisco
sccp phone in [EMAIL PROTECTED]
--- Sergio Chersovani [EMAIL PROTECTED] wrote:
Javier Chia ha scritto:
Any help?
I should install [EMAIL PROTECTED] to reproduce your
environment, I need
some time 'cause I'm busy
http://chan-sccp.berlios.de/
20050715 ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050715.tar.bz2
- cisco 7936 initial support
- gcc 2.95 compatibility
- fixed a removing session issue (introduced with the new socket code)
- minor code cleanups
- added (very) experimental support to CALL
Chris, as I look over my stack of unopened Polycom 501s, I think to
myself that I would enjoy seeing your provisioning script if you
wouldn't mind sharing it.
Chris Mason (Lists) wrote:
Michael Graves wrote:
I have a number of Polycom phones to setup with my * server. For my
initial few
1) reinvite=yes is incorrect syntax? Check the info here:
http://voip-info.org/wiki-Asterisk+sip+canreinvite
You can keep canrenvite=yes, but why do you want that?
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go
Have you set correctly the externip and localnet keywords in sip.conf?
Julian.
On 7/15/05, Damon Estep [EMAIL PROTECTED] wrote:
I have an * box behind a NAT router (static NAT, port ACLs set up correctly)
Most of the SIP users are on the local subnet with the * box, they work fine
Take
Ditto, I only have a couple of the polycomm phones spent the better part of
1 day figuring out how to get them configured properly.
--
Ted Serreyn Phone:262-432-0260 Fax:262-432-0232
Serreyn Network Services, LLChttp://www.serreyn.com/
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