Joel wrote:
I'm going to repeat this to you again, on the receiving side you need
to set:
Hi Joel
thanks for your advice, but I'm having trouble sending an SMS, not
receiving. Your suggestion seems to be only concerned with receiving?
/Per Jessen, Zürich
Yuan LIU wrote:
P[ 2] -- None
-- SMS[-1] RX 93 00 6D
-- SMS[0] TX 10 98 96 00 10 01 00 00 11 06 00 00 00 00 00 00 00
12
03 00 02 00 04 13 65 00 53 65 63 75 72 69 74 79 20 72 65 73 65 61 72
63 68 65 72 73 20 68 61 76 65 20 74 72 61 63 65 64 20 73 70 61 6D 2D
73 65 6E 64 69 6E 67 20
Hi all,
* i am using widows based asterisk pbx(AstWin) which i have down loaded
from www.asteriskwin32.com . we r using x-lite as a soft phone now .we have
created 10 sip users in sip.conf and configured extensions.conf too. all of
us could make calls through asterisk. we made 10 calls at the
I just finished up a perl script that connect via the * AMI and triggers
actions based when events fire. The triggers are then defined in the a
triggers.conf file and point to an action defined in actions.conf, these
actions can be any command line application.. pulling data from events is as
Robinson C P wrote:
Hi all,
* i am using widows based asterisk pbx(AstWin) which i have down loaded
from www.asteriskwin32.com http://www.asteriskwin32.com . we r using
x-lite as a soft phone now .we have created 10 sip users in sip.conf and
configured extensions.conf too. all of us could
Do you have the context numberplan-custom-1 in your extensions.conf file? I
think if you don't have it in extensions.conf then it goes back to using
default.
Yes, it is defined in extensions.conf
extensions.conf
.
[default]
exten = _X.,1,NoOp(This is default)
[numberplan-custom-1]
Per,
Have you tried the smsq app? That's what I use to send SMS, I don't
think I ever got the SMS app to work for sending. It receives fine.
/usr/bin/smsq --motx-channel=ZAP/1/0809101000 $RECIPIENT message here
works fine here.
___
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Hello everyone!
I've got a Eicon Diva Server BRI card into my * box working just fine,
but I wander if there's anything I can do to improve voice quality for
my operators. I'm thinking something along the lines of auto gain and
sudden noise suppression (like when you hit a fax machine or the
Wilson Pickett wrote:
Per,
Have you tried the smsq app? That's what I use to send SMS, I don't
think I ever got the SMS app to work for sending. It receives fine.
Hi Wilson,
yeah, I am using smsq, but part of my problem was my own confusion - as
usual :-)
I'm at the point now where I
Hi,
I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP
Trunk without MTP (media termination point). Howerver, Cisco 79xx phones
do not support RFC2833, they always notify CCM5 via SKINNY channel no
matter where they send RTP to.
For non-MTP trunk there's Out-of-band DTMF
I have had a TDM400 with 2 FXO and 2 FXS working for ages (12 months). It has
stopped working. All four green lights are still lit. I have rebuilt zaptel and
asterisk and restarted but the problem persists.
/sbin/ztcfg -
Zaptel Configuration
==
Channel map:
Channel 01:
On Thu, Apr 19, 2007 at 02:08:47AM -0700, kjcsb wrote:
I have had a TDM400 with 2 FXO and 2 FXS working for ages (12 months). It
has stopped working. All four green lights are still lit. I have rebuilt
zaptel and asterisk and restarted but the problem persists.
/sbin/ztcfg -
Zaptel
Putting my Westhawk Ltd protocol consultancy hat on.
Due to old age and good luck, westhawk as a full class C (256 ipv4
addresses)
so all our machines have routable adresses, putting us in a similar
position to
the way the rest of you would be when/if v6 takes off.
This is quite
Hi all,
I am trying to indentify a problem: I have 2 machines, one with Asterisk
1.0.11, the second with Asterisk 1.2.17. Both running with the same zaptel
(1.2.16). Asterisk 1.0.11 running on Sarge with AMP's dialplan and the 1.2.17
running on Etch with FreePBX's dial plan.
Now on both
I had the same problem with a sangoma card. Try to unplug channel 1 and
then restart zaptel and see if the problem persist. If so unplug 2 and so
on. My problem was that I had a FXO pluged into FXS port. Maybe someone
switched some cables and did not tell you ?
Astawerks
VoIP Hardware sales
Hi,
Did someone ever managed to make Astertest
(http://www.asteriskguru.com/tutorials/astertest.html) work ? I followed
all the instructions of this tutorial and corrected the mistakes pointed
by the users but it still doesn't work. I can compile it and load
app_securax_cpuinfo.so. When
Thank you, it passed the 1024 limit today just fine :)
On 4/18/07, matteo brancaleoni [EMAIL PROTECTED] wrote:
Hi,
On Wed, 2007-04-18 at 15:56 +0300, Maysara A. Abdulhaq wrote:
hello,
i tried to increase the number in /proc/sys/fs/file-max , which was:
203511
and file-nr was
21120
Steve Finkelstein wrote:
Doug,
Were you also having issues specifically related to SIP authorization,
same as we're experiencing?
I didn't dig that deeply into it, I did a telnet into the phone, did a
show register and saw that it was having problems with the register and
the timers were
Dear
can Background, plays wav format , for any incomming,
codecs,
best
Mani
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Greetings list,
A quick question regarding extensions.conf #include behaviour if I may. I'm
sure someone will know the answer off the top of their head...
How does asterisk handle overloading of contexts. For example, say an
extension exists in extensions.conf as follows:
[incoming]
some
Thanks for all the discussions guys.
I guess all of these points are know and well researched or discussed.
What I'm really after are some discussions about how product design
needs to change to accommodate our new IPV6 address space. How can RD
departments take advantage of the coming change
Hi,
This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the
display screen. How can I set the New York time? What value I have to give to
GMT offset value?
Look forward to your response. Thank you.
Regards,
Chandra.
-
On Thu, Apr 19, 2007 at 11:56:18AM +0100, Chris Bagnall wrote:
Greetings list,
A quick question regarding extensions.conf #include behaviour if I may. I'm
sure someone will know the answer off the top of their head...
How does asterisk handle overloading of contexts. For example, say an
Hello,
As i was involved in the development i can say that it is for the moment
abandoned by the developers, we might get back to it later but are first
focussing on some other projects. (Idefisk being the main one)
I don't think it will work with any recent version of asterisk and even
if
Hi,
Check if your system has the /dev/files needed.
I think some installation didn't do it automatically.
Manolet Gmail wrote:
2007/4/18, Ronaldo [EMAIL PROTECTED]:
Hi Manolet,
You have to install zaptel in order to make MeetMe application to work.
MeetMe needs a kind of timer device
On Thu, 19 Apr 2007, Cosmin Prund wrote:
Hello everyone!
I've got a Eicon Diva Server BRI card into my * box working just fine,
but I wander if there's anything I can do to improve voice quality for
my operators. I'm thinking something along the lines of auto gain and
sudden noise
On Thu, Apr 19, 2007 at 10:24:18AM +0100, Tim Panton wrote:
Putting my Westhawk Ltd protocol consultancy hat on.
Due to old age and good luck, westhawk as a full class C (256 ipv4
addresses)
so all our machines have routable adresses, putting us in a similar
position to
the way the
Hi
I have previously had good success on smaller installations with TDM400P
cards. I now have a UK customer looking for 8-10+ lines and it seems
like a PRI would be most economical + reliable?
Has anyone here used PRI interfaces in the UK and can confirm that it
works well (using Trixbox
You can use the web interface and set it to -5 gmt. Google for free NTP
servers. I used to use time.nist.gov and got mixed results. I found
another one that works almost all of the time.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED]
Dean Collins wrote:
What I'm really after are some discussions about how product design
needs to change to accommodate our new IPV6 address space. How can RD
departments take advantage of the coming change and build new
functionality to suit.
As in what can I do with my IPV6 enabled electric
Hi Chandra -
This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the
display screen. How can I set the New York time? What value I have to give
to GMT offset value?
The GMT offset value is in seconds. So, for example, the value to use
for EST is -18000, because EST is -5
Just a thought, try kannal, use system in your dialplan and call lynx with a
properly formatted URL for Kannal.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Per Jessen
They are all terrible in their own way. Don't you have someone you can
delegate the Verizon babysitting responsibility to? I would consider
sales calls a little more important than being a babysitter.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
Hi
I've configured the queue on my asterisk box and everything is working fine.
In my queue I've 3 agents logged in the queue. When call comes they are able
to receive the calls without any problem. But some time they are on break
and there extension rings and no one is there to answer the call
I have previously had good success on smaller installations with TDM400P
cards. I now have a UK customer looking for 8-10+ lines and it seems
like a PRI would be most economical + reliable?
Best bet would be to talk to insert telco of preference and ask them what
they recommend. For anything
KPML is now an RFC -- http://www.ietf.org/rfc/rfc4730.txt
Asterisk doesn't support KPML today. That doesn't mean it can not be
developed if there is sufiicient interest. The true value of adding KPML
support in Asterisk is when it is acting as a 'softswitch' (call controller
without media
On Thu, Apr 19, 2007 at 08:36:12AM -0400, Steve Totaro wrote:
Just a thought, try kannal, use system in your dialplan and call lynx with a
properly formatted URL for Kannal.
Or indeed Kannel (www.kannel.org)
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US
Any ideas on this?
Thank you.
Phil Menico | Chief Technology Officer | 212-951-7632
XTEND Communications | 171 Madison Avenue, New York, NY 10016 |
www.xtend.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Menico
Sent: Tuesday, April 17,
Steve Totaro wrote:
Just a thought, try kannal, use system in your dialplan and call lynx
with a properly formatted URL for Kannal.
Thank, that's interesting, I might just try that.
I've actually got SMS-sending up and running, just not over Asterisk. I
use sms_client and an analog modem,
If your phone is getting its parameters by DHCP from a linux server, add
the NTP server option to that server:
in /etc/dhcpd.conf
option time-servers 192.168.0.3;
If your phone is getting an NTP server setting by DHCP server, you can't
override that from any setting. I came across
Kevin P. Fleming wrote:
Eric ManxPower Wieling wrote:
Any updates on this?
The code is done and initially tested; it is being reviewed internally
and should be available on Friday or Monday.
Under what circumstances would this clipping be present? Is this patch
going to be recommended for
Hi, Dean:
Dean Collins wrote:
Or if you are a writer who has published something on this exact topic
that has been run at a national print level……want a gig?
Not on this exact topic -- but I'm a professional writer AND a network
consultant who has even had some exposure to IPv6.
What's the
Thank you all for your response, but it appears that some of you
didn't understand my question. I know I can schedule a cron to check
the status (I can even use asterisk -rx sip show peers | grep
UNREACHABLE if I use a cron) but that is not what I want. I want
either a way that just as asterisk
In article [EMAIL PROTECTED],
Chris Bagnall [EMAIL PROTECTED] wrote:
A quick question regarding extensions.conf #include behaviour if I may. I'm
sure someone
will know the answer off the top of their head...
How does asterisk handle overloading of contexts. For example, say an
extension
Hi, now i can log in ok on my xlite, somebody calls me and everythink
its okey. i hear and the caller hear. (the pc with the xlite have
DMZ).
But now i close xlite and put the same extension on a grandstream 286
(dont have DMZ). When somebody calls me the caller can hear me. but i
cant hear!
I use modprobe ztdummy, next i restart asterisk and now works fine,
modprobe is to load the driver rigth? what i need to do in order to
load automatically, not at the boot time but when asterisk start?
2007/4/19, Ronaldo [EMAIL PROTECTED]:
Hi,
Check if your system has the /dev/files
Hi,
Is it possible to design an IVR using SER ? If yes please advice.
thanks
arun
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Thu, 19 Apr 2007, Arun Kumar said something to this effect:
Is it possible to design an IVR using SER ? If yes please advice.
One could certainly incorporate SER into the infrastructure of an IVR
system, particularly if it's distributed, redundant, or involves shuttling
calls between
Matt is that a fact or an ASSumption?
On 4/17/07, Salvatore Giudice
[EMAIL PROTECTED] wrote:
Can anyone recommend a VoIP provider who supports LNP? I need to move to a
new provider for inbound calling and I want to keep my current numbers. My
current provider is a gaggle of retards.
Any
Best bet would be to talk to insert telco of preference and ask them what
they recommend. For anything more than about 8 channels a PRI is likely to be the
most cost-effective route if you need physical lines on-site, especially if that's
likely to grow beyond 8 lines in the future.
Salvatore,
On 4/17/07, Salvatore Giudice
[EMAIL PROTECTED] wrote:
Can anyone recommend a VoIP provider who supports LNP? I need to move to a
new provider for inbound calling and I want to keep my current numbers. My
current provider is a gaggle of retards.
Most VoIP telephony service
hi, to get it work i change under sip.conf
nat: route
Allow RTP reinvite:update
with that i can hear, without dmz... but... why?
2007/4/19, Manolet Gmail [EMAIL PROTECTED]:
Hi, now i can log in ok on my xlite, somebody calls me and everythink
its okey. i hear and the caller hear. (the pc with
On 2007-04-18 at 22:57:39, Steve Finkelstein [EMAIL PROTECTED] wrote:
I was only able to get a stable setup after I moved my 7940 back to SIP
version 7.5
I'm working with Steve on this issue, and the phone was running 8.6 when we
originally tried it. Just for kicks i downgraded it to 7.5 this
I am not sure of the best way to do this, so I thought I would query the list.
I have about 100 internal extensions ranging from 2000 - 2100. Each
internal extension has a external DID number. For example: 2001 =
5552871620. As you can see the internal externsion and DID don't
match in any
How can I add extra digits to go through different carriers?
If it is long distance, but not a toll free number, then add 10 15
xxx.
--
Thanks
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To UNSUBSCRIBE or
Chris Mason (Lists) wrote on 4/19/07 6:10 AM:
If your phone is getting its parameters by DHCP from a linux server, add
the NTP server option to that server:
in /etc/dhcpd.conf
option time-servers 192.168.0.3;
If your phone is getting an NTP server setting by DHCP server, you
On Thu, 19 Apr 2007, Forrest Beck said something to this effect:
I thought of maybe adding a key for each extension to the astdb and
have a Macro query the astdb. Any other ideas?
That would work, and is certainly the easiest, since you can bulk-load
the DID - extension maps via external
If there was something useful in ones kettle having an ethernet
connection, it would probably already have it. After all, with NAT'ing
there's no real shortage of IP-addresses. And perhaps we would already
have K2K networks, with K2K proxies etc.
It'd be great if I could get my kettle to
Brian McEntire wrote:
A follow-up with the solution in case anyone else is looking for this
answer:
I created two contexts in my zapata.conf file, since each VOIP line is
terminated by a VOIP adapter and then just comes in hardwired to the
TDM400 via RJ11 line, I know which VOIP number is
Steve Totaro wrote:
You can use the web interface and set it to -5 gmt. Google for free NTP
servers. I used to use time.nist.gov and got mixed results. I found
another one that works almost all of the time.
If you use pool.ntp.org (or a regional variant thereof, such as
ca.pool.ntp.org) it
On Thu, 19 Apr 2007, [EMAIL PROTECTED] said something to this effect:
How can I add extra digits to go through different carriers?
If it is long distance, but not a toll free number, then add 10 15
xxx.
Use IF conditionals in the dial plan to manipulate strings and/or create
new dial
Hi, Ed:
Ed W wrote:
Agreed. My experience is that quality is higher on Voip than it is via
a TDM400p. However, my experience hasn't been that VoiP is as reliable
as copper lines and so unless you can tolerate the odd outage once per
month or two then you might want to stick to copper for
[EMAIL PROTECTED] wrote:
How can I add extra digits to go through different carriers?
If it is long distance, but not a toll free number, then add 10 15
xxx.
Here are a couple of good reads for you
http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Planning
Stephen Bosch wrote:
Steve Totaro wrote:
You can use the web interface and set it to -5 gmt. Google for free NTP
servers. I used to use time.nist.gov and got mixed results. I found
another one that works almost all of the time.
If you use pool.ntp.org (or a regional variant
Hi all,
I want to make a SIP trunk between a Cisco 2811 router and a Asterisk.
Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk
2XX). Now I want to configure a trunk so that 2811 users can call *
users. I've been reading a lot but I'm still confused.
Hope you can help me,
--
Hi guys,
i just came to know that CDR(dst) field is set to current extension instead
of the dialed no. i need to set it to DNID because our every user has 5 dids
and i want to show the caller at the end of the month which numbers he
dialed for every call, along with other cdr info. Our rating
Grigoriy wrote:
I'm trying to connect Asterisk 1.4 and Cisco CallManager 5
using SIP Trunk without MTP (media termination point).
Howerver, Cisco 79xx phones do not support RFC2833, they
always notify CCM5 via SKINNY channel no matter where they
send RTP to.
If you are running the phone
Hi
I run asterisk 1.4.2 with zaptel 1.4.1.
Zaptel is only needed for the ztdummy driver to get the Meetme()
application to work. I don't have any specific hardware.
And it does work nicely. When I reboot the machine however I have to
manually load the driver like this:
[EMAIL PROTECTED] ~]#
Greetings;
We are trying to get Asterisk up and happy at our site-we tried VOIP
using Sphere about a year ago, spent a *boodle* on expensive hardware
and services from a local expert, but it never was happy.
I'm brand-spanking new at VOIP, and I've learned a *ton* getting
Asterisk breathing
David Olsen wrote:
On 2007-04-18 at 22:57:39, Steve Finkelstein [EMAIL PROTECTED] wrote:
I was only able to get a stable setup after I moved my 7940 back to SIP
version 7.5
I'm working with Steve on this issue, and the phone was running 8.6 when we
originally tried it. Just for
On Mon, 16 Apr 2007, Martin Joseph wrote:
Just a warning for you all that are using Nokia series E phones for SIP
function.
I updated my phones firmware today using the Nokia Updater, and now the SIP
functionality, which previously worked pretty well is completely broken.
The phone no
Dan Austin wrote:
If you are running the phone loads that shipped with CCM5,
then your skinny phones have 'support' for RFC2833. CCM
figures out during the call if the call will traverse a
SIP trunk and instruct the phone to use RFC2833 for DTMF
I have a CCM5-Asterisk trunk setup for MeetMe
Manolet Gmail wrote:
I use modprobe ztdummy, next i restart asterisk and now works fine,
modprobe is to load the driver rigth? what i need to do in order to
load automatically, not at the boot time but when asterisk start?
Funny. I just posted exactly the same question. How someone has an
On 2007-04-19 at 12:29:50, Doug Lytle [EMAIL PROTECTED] wrote:
I was mistaken. I just checked the phone, it's 7.4
Odd. No dice either there. I must be doing something else wrong.
-d
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Hi All
According to http://bugs.digium.com/view.php?id=6457 this has been
resolved since 04-11-2006 and I have seen mentioned since 1.2.7. I have
tried using mixmonitor on asterisk 1.2.13 and 1.2.17 with the exact same
results. The WAV files are recorded but are cut short. I am using a
David Olsen wrote:
On 2007-04-19 at 12:29:50, Doug Lytle [EMAIL PROTECTED] wrote:
I was mistaken. I just checked the phone, it's 7.4
Odd. No dice either there. I must be doing something else wrong.
Do you see anything weird when logging (telnet to the ip) into the phone
and
I'm using zabbix (http://www.zabbix.com/) as a complete monitoring solution
zabbix agent has the possibility to specify custom checks that are run
as often as you wish
(maybe an asterisk -rx sip show peers | grep UNREACHABLE | wc -l)
the output of the script is sent to zabbix server which can
Grigoriy wrote:
Dan Austin wrote:
If you are running the phone loads that shipped with CCM5,
then your skinny phones have 'support' for RFC2833. CCM
figures out during the call if the call will traverse a
SIP trunk and instruct the phone to use RFC2833 for DTMF
I have a CCM5-Asterisk trunk
Theo,
Unless things have changed significantly in the newer releases, you must
load zaptel prior to loading ztdummy. Additionally, the zaptel devices
are not created instantly, so after you load zaptel you must wait a few
seconds before loading ztdummy. You can perform some sort of polling
I don't have any usefull information to add, just that as can be see in a mail
from yesterday I have the same result, some files are being shorter than its
must be.
Regards,
-Original Message-
From: [EMAIL PROTECTED] on behalf of Garth van Sittert
Sent: Jue 19/04/2007 12:04 p.m.
To:
On 2007-04-19 at 13:09:51, Doug Lytle [EMAIL PROTECTED] wrote:
Do you see anything weird when logging (telnet to the ip) into the phone
and doing a show register?
I see as follows:
cisco-7960 show reg
LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERING
line APR state
However, my experience hasn't been that VoiP is as reliable
as copper lines and so unless you can tolerate the odd outage once per
month or two then you might want to stick to copper for the main
carrier? Does this match with the experience from others?
Until recently, I'd have agreed
In the zaptel source make config will install the zaptel init script
in /etc/rc.d/init.d for many distros.
Matthew J. Roth wrote:
Theo,
Unless things have changed significantly in the newer releases, you must
load zaptel prior to loading ztdummy. Additionally, the zaptel devices
are not
Edoardo Serra wrote:
I'm using zabbix (http://www.zabbix.com/) as a complete monitoring
solution
zabbix agent has the possibility to specify custom checks that are run
as often as you wish
(maybe an asterisk -rx sip show peers | grep UNREACHABLE | wc -l)
the output of the script is sent to
Hi,
I'm new to this list. For the last couple of days I was searching for
a good solution using AsteriskNOW. I noticed that in the configuration
steps of the server, they asked for a service provider. We don't
really need one.
We had something in mind like installing two Asterisk servers and
Edgar A. Luna Diaz wrote:
I'm having high load, choppy sound and slow responsives with an
asterisk server (version 1.2.12.1) that make a peak of 90 channels
(around 60 phones calling at max, isn't necessary to reach this peak
to get the problem). All the traffic is SIP, with recording for
If your SIP server loses REGISTERs then it cant place an inbound SIP call.
Try changing the REGISTER frequency to lower value.
When you see incoming SIP call fail, you might want to check whether the
REGISTERs are working.
Thanks,
Neel
-Original Message-
From: [EMAIL
David Olsen wrote:
On 2007-04-19 at 13:09:51, Doug Lytle [EMAIL PROTECTED] wrote:
Do you see anything weird when logging (telnet to the ip) into the phone
and doing a show register?
I see as follows:
cisco-7960 show reg
LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state:
Maybe that's a little over the top but Minotaur as a broadband provider
could offer a new range of electronic door bell devices that also
control access to key strike locking plates.
As part of selling customers their broadband IPV6 package they are given
free access to this device (or minimal
Hi Tim -
I'm new to this list. For the last couple of days I was searching for
a good solution using AsteriskNOW. I noticed that in the configuration
steps of the server, they asked for a service provider. We don't
really need one.
We had something in mind like installing two Asterisk servers
Well thanks for answering,
When I test, I use my GSM and call the number my provider gives me.
How often it works or not, I didn't make test like 10 calls per hour
for a pretty long time so I can't exactly tell. When I test, well
sometimes it works great, sometime, the incoming call is
Had an appointment for these schmoes to come out and install another
line. Was supposed to be 8-12. Its now 6PM and not even call.
Missed
3 sales calls waiting on these jerks.
No wonder customers were jumping ship to Vonage.
I once had to oversee Verizon install a PRI line in Manhattan.
Hello and thanks for answering,
As I just answer to Yuan LIU, what I don't understand, is that I can
place an outbound call from asterisk to a gsm at the same time I
can't get asterisk thought a inbound call. But I'll try what you
advice me.
I'll tell you the result of it
Jean-Marc LE
Chris Bagnall wrote:
However, my experience hasn't been that VoiP is as reliable as
copper lines and so unless you can tolerate the odd outage once per
month or two then you might want to stick to copper for the main
carrier? Does this match with the experience from others?
Until
On 4/19/07, Forrest Beck [EMAIL PROTECTED] wrote:
I thought of maybe adding a key for each extension to the astdb and
have a Macro query the astdb. Any other ideas?
That's how we do it. We created a MySQL DB that maps DIDs to extensions, and
a php script to write our configuration files
On Thu, Apr 19, 2007 at 06:28:51PM +0200, Theo Band wrote:
Hi
I run asterisk 1.4.2 with zaptel 1.4.1.
Zaptel is only needed for the ztdummy driver to get the Meetme()
application to work. I don't have any specific hardware.
And it does work nicely. When I reboot the machine however I have
Hi Diego -
I want to make a SIP trunk between a Cisco 2811 router and a Asterisk.
Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk
2XX). Now I want to configure a trunk so that 2811 users can call *
users. I've been reading a lot but I'm still confused.
I don't know if
Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones.
Local outbound calling works fine, but ATT requires clients enter 7 digit code
for long distance. All calls with 7 digit code are lost within 20 seconds of
the call. This is the message Im getting:
Apr 19
What is the output of:
ls -l /sys/class/zaptel
ls -l /sys/class/zaptel
total 0
drwxr-xr-x 2 root root 0 Apr 19 20:05 zapchannel
drwxr-xr-x 2 root root 0 Apr 19 20:05 zapctl
drwxr-xr-x 2 root root 0 Apr 19 20:05 zappseudo
drwxr-xr-x 2 root root 0 Apr 19 20:05 zaptimer
drwxr-xr-x 2 root root 0
On 2007-04-19 at 14:16:49, Doug Lytle [EMAIL PROTECTED] wrote:
This what mine looked like before the firmware downgrade. Except the
timer and expires were huge numbers.
Firewall issue?
I've also had issues when compiling my own kernel and thinking it was a
good thing to enable 'SIP
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