Hello,
Which tools are recommendable for monitoring VOIP, bandwidth, server
alarms, etc.?
Thanks,
Elliot
--
_
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New to Asterisk? Join us for a live introductory
Hello,
Can SRTP be set on a per device basis?
Thanks,
Elliot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://
s not
received in this amount of time, the call will autocongest"
Thanks,
Elliot
On Thu, Jul 5, 2012 at 12:31 PM, Olle E. Johansson wrote:
>
> 4 jul 2012 kl. 13:32 skrev Elliot Murdock:
>
>> Hello,
>>
>> I am trying to get clarity with the sip.conf timer configu
Hello,
I am trying to get clarity with the sip.conf timer configuration. The
current configuration states:
;--- SIP timers
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms o
Hello,
http://www.linuxinnovations.com shows the changes and updates between
the various versions of Asterisk.
--Elliot
On Sat, Jan 7, 2012 at 2:10 AM, Joseph wrote:
> On 01/06/12 16:35, Joseph wrote:
>>
>> On 01/06/12 18:15, Eric Wieling wrote:
>>>
>>> Putting in a Wait(n) is only (sometimes)
Hello,
What is the meaning of this DEBUG message?
chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and
clearing call
Thanks,
Elliot
--
_
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Ne
Hello,
Take a look at linuxinnovations.com for various differences.
Elliot
On Wed, Dec 28, 2011 at 11:10 PM, Danny Nicholas wrote:
> Can somebody point me to an explanation from Kevin or Tzafir or someone else
> "up the food chain" explaining the differences/benefits of 1.6/1.8 vs
> 1.4/10.0?
>
Hello All,
An Asterisk server sometimes responds to a BYE Request with a Status
481 Call leg/transaction does not exist response and is also appearing
to drop calls randomly.
What does this mean and how's one to handle it? Does this indicate
there is some internal bug with the Server, since it a
Hello,
What is the method for changing the country for indications (eg. ring,
busy, etc. tones) in a dialplan?
Thanks,
Elliot
--
_
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New to Asterisk? Join us fo
tlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Oct 23, 2011, at 5:46 AM, Elliot Murdock wrote:
>
>> Hello All,
>>
>> It seems from the Asterisk documentation, a User places phone calls
>> i
Hello All,
It seems from the Asterisk documentation, a User places phone calls
into the Asterisk server and a Peers accepts phone calls from the
Asterisk server.
However, according to the document describing the "register =>"
command for sip.conf, it seems that Peers can in fact place calls into
Hello,
When debugging SIP in Asterisk is it possible to send the SIP debug
log to a specific file instead of the general log file, or even
better, send each call into its own file for easier analysis?
Thanks,
Elliot
--
_
-- Band
Hello Kevin,
SS7 parameters to Asterisk variables.
--Elliot
On Tue, Jul 19, 2011 at 3:31 PM, Kevin P. Fleming wrote:
> On 07/18/2011 05:05 PM, Elliot Murdock wrote:
>
>> I am wondering if the Libss7 add-on for Asterisk also translates ss7
>> variables into the dia
Hello,
Is SS7 and PRI in any way compatible in that if the interface is
configured one it will work for the other (granted, it will not have
any of the ISUP, etc. parameters available if the line is PRI) or are
they two distince protocols that have incompatible signalling?
Thanks,
Elliot
--
Hello!
I am wondering if the Libss7 add-on for Asterisk also translates ss7
variables into the dialplans for routing, accounting, etc?
Thanks,
Elliot
--
_
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New
Hello!
Is it possible to retrieve ISUP and other variables from the SS7
channel and use them in contexts in Asterisk?
Thanks,
Elliot
On Thu, Apr 15, 2010 at 6:04 PM, Ngo-Vi Hoai-Anh wrote:
> Sangoma uses wanpipe. Channel drivers set upon wanpipe (very
> simplifiedly speaking). You can configure
the
> Bug Tracker, "Google Voice inbound calls still work, it is just coming
> from Google Talk that doesn't."
>
> -Vladimir
>
>
> On 6/14/2011 5:51 PM, Elliot Murdock wrote:
>> Hello,
>>
>> Seems that it's been spotted and tracked at
>
ith detailed setup description.
>
> -Vladimir
>
>
>
>
> On 6/14/2011 1:26 AM, Elliot Murdock wrote:
>> Hello,
>>
>> To help clarify, Jabber is receiving the incoming packets, but
>> Asterisk does not seem to be associating it with the gtalk
>> configurat
be the problem?
Elliot
On Tue, Jun 14, 2011 at 2:02 AM, Elliot Murdock wrote:
> Hello,
>
> I am using 1.8.4.2 and while outgoing seems to work, incoming still
> does not route calls in to the appropriate context.
>
> Please advise.
>
> Thank you,
> Elliot
>
>
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
wrote:
> You must have 1.8+ its already been posted the 1.6 didn’t get a backport f
Hello,
When an INVITE packet is received, Asterisk immediately responds with
a 180 RINGING response before it receives a response from the
provider. Furthermore, when this occurs, no ring tone is created on
the caller's end.
I've played around with the progressinband setting, but to no avail.
A
Hello,
It seems that the outboundproxy parameter in sip.conf is what is needed.
--Elliot
On Tue, May 24, 2011 at 12:15 PM, Elliot Murdock wrote:
> Hello,
>
> Thank you for the reply.
>
> How would one go about adding a SIP proxy IP address in Asterisk for peers?
>
> The
11 at 11:11 AM, Olle E. Johansson wrote:
>
> 23 maj 2011 kl. 23.36 skrev Paul Belanger:
>
> > On 11-05-23 05:30 PM, Elliot Murdock wrote:
> >> Hello,
> >>
> >> I am wondering how to send a call to a specific IP address that is
> different
> >> t
Hello,
I am wondering how to send a call to a specific IP address that is different
than the host of the URI. For example, an invite to the URI is "
j...@phone.com" needs to be sent to the IP address 123.456.789.255, not to
the IP address of phone.com.
How is this done?
Thanks,
Elliot
--
__
packets make the SIP packets not processable by
Asterisk?
Thanks,
Elliot
On Mon, May 23, 2011 at 10:59 AM, Alex Balashov
wrote:
> On 05/23/2011 03:26 AM, Elliot Murdock wrote:
>
> There are some parameters in the ISUP data (coming into the network
>> via SIP-T packets) that nee
Hello,
There are some parameters in the ISUP data (coming into the network via
SIP-T packets) that need to be translated into SIP headers to be used by
asterisk for proper call routing. What gateways are available to accomplish
this?
Thanks,
Elliot
--
Hello,
Does Asterisk support the history-info header as well?
Also, what kind of ISDN mappings are available in 1.4 and 1.6.2 versions?
Thanks,
Elliot
On Thu, Jan 27, 2011 at 1:08 AM, Kevin P. Fleming wrote:
> On 01/25/2011 12:44 AM, Phil Lello wrote:
>>
>> Hi all,
>>
>> I'm looking at my opti
Hello All,
How does one go about creating a dahdi configuration file for multiple
PRI cards?
Thanks,
Elliot
--
_
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New to Asterisk? Join us for a live introduct
Hello!
I am trying to find out the device name and/or other identifying data
to be used in a context when a device transfers the call to new a
phone number. From running tests, it looks like the account code
variable (${CDR(accountcode)}) is set to the account code of the
device that placed the o
Hello!
The T.38 Digium Fax Driver sometimes responds with a successful
sending of a fax, when in fact, the fax did not go through.
1. Where does this problem lie?
2. How to go about fixing it.
Thanks,
Elliot
--
_
-- Bandwidth a
PM, Danny Nicholas wrote:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
> Sent: Wednesday, January 05, 2011 4:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discu
Hello!
I am trying to figure out how call transfers work in SIP. What
extension does the transferring and transferee devices go to?
Elliot
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New to Asteri
Hello!
I am wondering how the differences between G729, G729a, and G729b
effect call bridging and server interoperability. For example, can
one server use the G729 code with another server that uses the G729A
codec?
Also, which version is Asterisk set up to use?
Thanks!
Elliot
--
_
linuxinnovations.com is also a good place to seek out the differences
between the versions.
On Thu, Dec 16, 2010 at 7:00 PM, Leif Madsen
wrote:
> On 10-12-15 09:46 AM, bilal ghayyad wrote:
>>
>> Hi All;
>>
>> I need to know which version of asterisk to use, if to be 1.4 or 1.6 or
>> 1.8?
>>
>> Fo
Hello!
http://linuxinnovations.com lists the evolution of the various
commands, applications, and other essential parts of Asterisk from
version 1.4 until 1.8, so you may find it a good resource for helping
you make a decision.
--Elliot
--
___
Hello!
You may want to check out http://linuxinnovations.com, a simple
reference describing the practical differences between the various
versions of Asterisk. Seems it includes now version Asterisk 1.8.
--Elliot
On Sun, Nov 14, 2010 at 8:17 AM, Kyle Kienapfel wrote:
>
>
> On Sat, Nov 13, 2010
Hello All,
To help out with dependency issues, a wiki at
http://asteriskdependencies.linuxinnovations.com was set up. It's
fairly new, so any contributors interested would need to fill it with
the proper data.
-Elliot
On Mon, Sep 27, 2010 at 10:15 PM, Leif Madsen
wrote:
> On 10-09-26 02:55 P
Hello all!
After a recent upgrade of Asterisk 1.4.18 to 1.4.33.1, everything
looks ok, except when fax T.38 calls are being passed through the
server, the fax transmission fails, giving a "train failure" message
when invoking "fax show stats" in the fax server (Asterisk 1.6.0.26).
Rxfax and Txfax
asteriskuptospeed.linuxinnovations.com is also a good resource for
spotting many practical differences between the various versions.
On Wed, Feb 24, 2010 at 8:36 PM, Tilghman Lesher
wrote:
> On Wednesday 24 February 2010 10:16:25 Miguel Molina wrote:
>> Gergo Csibra escribió:
>> > Wednesday, Febr
Hello Klaus,
Just for your quick reference, here are some other changes in the AEL
from versions 1.4 to 1.6 (source:
http://asteriskuptospeed.linuxinnovations.com/core1.4-1.6.2.html and
CHANGES file) :
Macros are now implemented underneath with the Gosub() application.
Heaven Help You if you wrot
Hello!
I remember a while back I saw a way to answer a call from a device
that is not from the one ringing, but I don't remember what how to do
it. Any help would be great!
Thanks,
Elliot
___
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Hello Gordon!
Thanks...it works now.
How did you notice that it needed input from the keyboard?
Do you think I should make the developers aware of this issue?
Thanks,
Elliot
On Sun, Sep 6, 2009 at 2:12 PM, Gordon
Henderson wrote:
> On Sun, 6 Sep 2009, Elliot Murdock wrote:
>
>> He
Hello!
I have a very simple bash script:
#!/bin/bash
asterisk -rx "sip show peers" > /var/log/devices
When I run it in bash shell, everything works fine, but if I
background it (by adding & or using bg), nothing appears in the
/var/log/devices file.
Any reason for this behavior or help would be
Hello!
What are the nat_sip modules you mention?
When I set up a linux router some time ago and configured sip.conf
with net=yes, everything went smoothly just like any other router.
Elliot
On Mon, Aug 3, 2009 at 8:45 PM, Gordon
Henderson wrote:
> On Mon, 3 Aug 2009, Ketema Harris wrote:
>
>> m
Hello Everyone!
Thank you for all the information.
I am wondering how the Asterisk community has been working on
solutions to deal with the asymmetric quality of ADSL. Voip is
becoming popular and a bottleneck does exists on the ADSL upload side.
Elliot
On Sun, Aug 2, 2009 at 3:17 PM, Kevin P
possible to use G729 for upload,
but keep the higher quality codec, G711, for download.
Thanks,
Elliot
On Sat, Aug 1, 2009 at 11:06 PM, Alex Balashov wrote:
> Elliot Murdock wrote:
>
>> I am wondering how to configure Asterisk and devices so I can use
>> different codecs for upstr
Hello!
I am wondering how to configure Asterisk and devices so I can use
different codecs for upstream and downstream packets.
Thank you,
Elliot
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15
command work?
Thanks,
Elliot
On Thu, Jul 23, 2009 at 5:08 PM, Ishfaq Malik wrote:
> Hi
>
> You can retrieve it in real time using the AMI from a script
>
> http://www.voip-info.org/wiki/view/Asterisk+manager+API
>
> Ish
>
> Elliot Murdock wrote:
>> Hello Philipp,
&
Hello Philipp,
Thank you.
I could set that up, but is that status (of qualifying) stored
anywhere (besides the log files) that a script could use?
Regards,
Elliot
On Thu, Jul 23, 2009 at 12:47 PM, Philipp
Kempgen wrote:
> Elliot Murdock schrieb:
>> I am looking for a way to test
Hello!
I am looking for a way to test if a SIP device is still alive or not.
I want to add this functionality in an AGI or independent script in
order ensure all the SIP phones are properly connected to the system.
Thank you,
Elliot
___
-- Bandwidth an
Hello!
I set my devices to only use g77a, but I am getting this when I run
show channel"
NativeFormats: 0x8 (alaw)
WriteFormat: 0x8 (alaw)
ReadFormat: 0x4 (ulaw)
Why is ulaw (g711u) showing up for the Read Format?
Thanks,
Elliot
___
-- Bandwidt
:
> Elliot Murdock wrote:
>
>>Hello Again!
>>
>>I'm back again. I just checked the general settings in sip.conf and
>>noticed that both the "disallow" and "allow" parameters are commented
>>out. I would assume this would allow all codec
d to disallow=all, and allow=alaw, allow=ulaw,
and allow=g729 only then does Asterisk allow consider g729 as a usable
codec.
So are my suspicions correct that by default, Asterisk only allows
gsm, ulaw, alaw, and H263 and one needs explicitly allow also g729?
Thanks you,
Elliot
On Thu, Jul 2, 2
2009 at 7:08 PM, Kevin P. Fleming wrote:
> Elliot Murdock wrote:
>
>> [Jul 2 16:56:26] VERBOSE[13420] logger.c: --- (12 headers 12 lines) ---
>> [Jul 2 16:56:26] VERBOSE[13420] logger.c: Sending to 216.48.184.50 :
>> 5060 (no NAT)
>> [Jul 2 16:56:26] VERBOSE[
Hello Everybody!
Here are the full SIP logs!
<--- SIP read from 216.48.184.50:5060 --->
INVITE sip:6587972772285...@82.80.231.238:5060;user=phone SIP/2.0
Call-ID: 699864475636237-1246542986-18105
From: ;tag=24794
To:
Content-Type: application/sdp
CSeq: 1 INVITE
Via: SIP/2.0/UDP 216.48.184.5
Hello!
Here is the 200 OK response from my server:
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 10646 10646 IN IP4 212.80.21.238
Owner Username: root
Se
Thu, 2 Jul 2009, Elliot Murdock wrote:
>
>> Hello!
>>
>> Which RFC specifies the corresponding number of the formats?
>>
>> Where in the Asterisk source code does it state the SDP formats?
>>
>> Does Asterisk follow the formats of IANA?
>> (h
Hello!
Which RFC specifies the corresponding number of the formats?
Where in the Asterisk source code does it state the SDP formats?
Does Asterisk follow the formats of IANA?
(http://www.iana.org/assignments/rtp-parameters)
Thank you,
Elliot
On Thu, Jul 2, 2009 at 3:44 PM, Elliot Murdock
Asterisk?
Thanks,
Elliot
On Thu, Jul 2, 2009 at 3:25 PM, Kevin P. Fleming wrote:
> Elliot Murdock wrote:
>> Hello!
>>
>> I noticed that the SIP packet contains this line:
>>
>> m=audio 6 RTP/AVP 18 98 96 97 101 13
>>
>> However, there is
Hello Kevin,
Gotcha...however, the mime is G.729a with an extra period, so it
doesn't get recognized.
However, as I asked before, does Asterisk map any of the RTP/AVP
profiles or does every format need to be defined in the SDP with a
rtpmap attribute?
You see, the incoming SDP supplies format 18
with the G729 codec, since the calls fails at
pickup.
Thank you,
Elliot
On Thu, Jul 2, 2009 at 11:15 AM, Philipp
Kempgen wrote:
> Elliot Murdock schrieb:
>> Thank you for that piece of information. Which RFC does it state that
>> the audio name is "G729"?
>
> htt
format is there any way for
Asterisk to negotiate it *WITHOUT an explicit rtpmap?
Thank you,
Elliot
On Thu, Jul 2, 2009 at 11:20 AM, Elliot Murdock wrote:
> Hello!
>
> I noticed that the SIP packet contains this line:
>
> m=audio 6 RTP/AVP 18 98 96 97 101 13
>
> However, th
Asterisk to negotiate it within an explicit rtpmap?
Thank you,
Elliot
On Thu, Jul 2, 2009 at 9:34 AM, Elliot Murdock wrote:
> Hello!
>
> Thank you for that piece of information. Which RFC does it state that
> the audio name is "G729"?
>
> Thanks,
> Elliot
>
Hello!
Thank you for that piece of information. Which RFC does it state that
the audio name is "G729"?
Thanks,
Elliot
On Thu, Jul 2, 2009 at 12:16 AM, Kevin P. Fleming wrote:
> Elliot Murdock wrote:
>> Hello!
>>
>> I have a sip device that is sending in the SDP:
Hello!
I have a sip device that is sending in the SDP:
rtpmap:98 g729a
It does not seem like Asterisk is negotiating the codec properly,
because while the call rings, the rtp lines fail. However, on other
sip devices that have "rtpmap:18 g729" in their SDP, things work fine
with Digium's commer
Hello!
I am looking for a way to dynamically redirect a call while it is
ringing to another device. Basically, if a person is far away from
his desk, he should have the option to use another phone and pick up
the call.
Thanks for any suggestions,
Elliot
_
ly answers the local call,
but the final dial at priority 3 will remain unanswered.
Thanks,
Elliot
On Tue, Jun 30, 2009 at 12:31 PM, Elliot Murdock wrote:
> Hello,
>
> Oddly enough, sound is sent to original caller if it is a registered
> SIP device on the server. If the caller is
Hello,
Oddly enough, sound is sent to original caller if it is a registered
SIP device on the server. If the caller is remote, than nothing is
passed back.
Any help will be greatly appreciated,
Elliot
On Tue, Jun 30, 2009 at 12:13 PM, Elliot Murdock wrote:
> Hello!
>
> I am trying to
Hello!
I am trying to set up a dialplan that uses the Local channel type:
[default]
exten => s,1,dial(local/2...@dialplan/n)
[dailplan]
exten => 220,1,saydigits(123)
exten => 220,2,dial(SIP/120||m)
The calling party does not hear any of the digits nor the music on
hold. What should be done
Hello!
I have a 64 bit Asterisk system and am wondering how to use Digium's 32 bit
fax driver. Is there some kind of emulation that can be used?
Thanks!
Elliot
___
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asterisk-users mail
Hello!
My DID provider has multiple IPs addresses that is sends packets from. How
to do associate more that on IP address to a sip device in sip.conf (or any
other ideas)?
Thanks,
Elliot
___
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Hello Michiel,
Yep...that's the reason. I changed those other extensions and
everything is fine now. Asterisk is pretty clever, just need to keep
up with it.
Thanks
Elliot
On 5/27/09, Michiel van Baak wrote:
> On 14:49, Wed 27 May 09, Elliot Murdock wrote:
>> Hello!
>>
&
27/09, Elliot Murdock wrote:
> Hello!
>
> I am having an odd problem in that when the caller dials extension "2"
> in a dialplan, the system waits 3 to 4 seconds before proceeding.
> This doesn't happen when any other other extensions are dialed,
> including
Hello!
I am having an odd problem in that when the caller dials extension "2"
in a dialplan, the system waits 3 to 4 seconds before proceeding.
This doesn't happen when any other other extensions are dialed,
including an identical dialplan on other another extension!
Is this a bug?
Later,
Ellio
need to mount the voicemail directory on both servers.
Thanks,
Elliot
On Fri, May 22, 2009 at 3:32 PM, Jeff LaCoursiere wrote:
>
> Lets start from the beginning. Why are using a network share for your
> voicemail in the first place?
>
> j
>
> On Fri, 22 May 2009,
s kind of like me deciding to use ntfs on a local partition on
> linux box instead of ext3/4, jfs, reiserfs, etc.
>
> --
> Matt
>
> On Thu, May 21, 2009 at 5:06 AM, Elliot Murdock wrote:
>
>> Hello!
>>
>> Thanks...I set up a Samba mount, which works ok, except
is saving files with the last three characters, wav, as uppercase, WAV.
What is the procedure to ensure all the files are saved as is in Samba?
Thanks,
Elliot
On Thu, May 14, 2009 at 5:12 PM, Tilghman Lesher <
tilgh...@mail.jeffandtilghman.com> wrote:
> On Thursday 14 May 2009 08:14:
.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Elliot Murdock
> *Sent:* Wednesday, May 13, 2009 1:09 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Voicemail and remote directory with SSHFS
>
>
>
> Hello!
&g
Hello!
I am trying to mount a remote directory for voicemail using sshfs. However,
whenever Asterisk attempts to write the file, it fails, because SSHFS cannot
lock the directory. Is there a solution to this problem or an alternative
method for using a remote directory for voicemail?
Thanks,
El
so
>> CLI>load chan_zap.so
>>
>> it will tell you now what's the problem with registering the channels
>>
>> Martin
>>
>> On Thu, Apr 2, 2009 at 7:18 PM, Elliot Murdock wrote:
>>> Hello!
>>>
>>> Here is all I got:
>&
nfigured with span= and bchan= and dchan= keyword
> and /etc/asterisk/dahdi*.conf with channel => keyword.
>
> check it out ... this is the first step...
>
> dahdi_cfg -vv should show all your 64 channels
>
> Martin
>
> On Thu, Apr 2, 2009 at 6:36 PM, Elliot Murdock
Hello!
Here is what I get for pri show version:
libpri version: 1.4.9
Thanks,
Elliot
On Fri, Apr 3, 2009 at 2:54 AM, Dave Poirier wrote:
>
>
> On Thu, Apr 2, 2009 at 4:36 PM, Elliot Murdock wrote:
>>
>> Hello!
>>
>> I am trying to configure my digium TE220 d
Hello!
I am trying to configure my digium TE220 dual-span pci express card
with Dahdi. I seemed to have managed to set up the card with the
Dahdi kernel, as demonstrated by executing dahdi_scan:
[1]
active=yes
alarms=RED
description=T2XXP (PCI) Card 0 Span 1
name=TE2/0/1
manufacturer=Digium
devi
; On Sun, 8 Mar 2009, Elliot Murdock wrote:
>
>> Hello!
>>
>> There will be disk writing in these areas:
>> 1. Logs
>> 2. CDRs
>> 3. MYSQL Call logs
>> 4. Faxes and voicemail
>
> I'd not consider these to be a heavy load myself...
>
>
wrote:
> On Sun, 8 Mar 2009, Elliot Murdock wrote:
>
>> Hello!
>>
>> Oh, yes, I will be mirroring the harddrives in case of any failures.
>>
>> What is your opinion about using (software) RAID? Do you think the
>> overhead impacts performance too much?
>>
, Mar 8, 2009 at 4:26 PM, Jay Milk wrote:
> Elliot Murdock wrote:
>> Hello Everybody!
>>
>> I am currently setting up an Asterisk server for medium to high load
>> (approximately 20-35 concurrent phone lines).
>>
>> Do you think the following specs will suff
Hello Everybody!
I am currently setting up an Asterisk server for medium to high load
(approximately 20-35 concurrent phone lines).
Do you think the following specs will sufficiently satisfy this system?
CPU: XeonQC3220 2.4GHZ 8M
RAM: 2X2GB/800
Harddrive: 1X250GB
I could add harddrives and part
Hello Everyone!
According to voip-info.org the correcy syntax for the realtime function is:
REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read
REALTIME(family|fieldmatch|value|field) on write
It seems from the syntax that it is only possible to retrieve a full
row according to the val
Hello Everybody!
My server is attempting to connect to a SIP device, but is not
succeeding to. I checked the actual packets traveling back and forth
with ngrep and I noticed some odd packets coming in.
Here is the outgoing INVITE packet sent to the SIP device:
#
U asteriskserver:5060 -> sipdevi
Hello,
Sorry, just to avoid confusion, in my last post, the proper smsq command is:
smsq --motx-channel="Zap/g3/7285267" 7286657 "test"
and not to the number 2285267 as stated in the previous post.
Elliot
On Fri, Dec 26, 2008 at 12:26 PM, Elliot Murdock wrote:
> Hello
23] NOTICE[2647]: pbx_spool.c:351 attempt_thread: Call
completed to Zap/g3/7285267
Thanks for any help,
Elliot
On Sun, Dec 21, 2008 at 1:57 PM, Hans Witvliet wrote:
>
> On Sat, 2008-12-20 at 19:27 +0200, Elliot Murdock wrote:
> > Hello!
> >
> >
> > What kind of sms
Hello!
What kind of sms text messaging capabilities does Asterisk have?
I do not know very much about about SMS technology, but I am looking for the
following features:
1. mobile SIP devices can send and receive SMS messages
2. Asterisk server be able to accept and send SMS messages through PRI
have just
> one call for agent??
>
> Thanks
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
>
> [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
> Lesher
> Sent: martes, 25 de noviembre de 2008 03:37 p.m.
> To: Asterisk Users Mailing List - Non-Com
extension is busy, redirect call to local extension "516".
> How this can be achieved?
>
> Right now I have:
>
> exten => 1234567,1,Dial(SIP/515)
> exten => s-NOANSWER,n,Dial(SIP/516)
> exten => s-BUSY,n,Dial(SIP/516)
>
> But this does not work (I guess
;
>
>
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Elliot Murdock
> *Sent:* martes, 25 de noviembre de 2008 11:04 a.m.
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Disabling Call-Waiting
>
>
>
> Hello!
>
> I hav
Hello!
I have a few sip devices and it is necessary for me to disable call-waiting
and immediately return a busy signal if the sip's channel is busy on them.
What is the procedure to do so in Asterisk 1.4?
Thank you,
Elliot
___
-- Bandwidth and Coloca
2008 at 11:25:33AM +0300, Tzafrir Cohen wrote:
>> On Sun, Aug 03, 2008 at 10:05:46AM +0300, Elliot Murdock wrote:
>> > Hello,
>> >
>> > My Digium wct4xxp suddenly stopped working. Here are some of the logs:
>> >
>> > zap restart
>> > [Aug
Thanks Tzafrir,
This is what I get:
module unload chan_zap.so
-- Unregistered channel -2
-- Unregistered channel 1
...
-- Unregistered channel 122
-- Unregistered channel 123
-- Unregistered channel 124
CLI> module load chan_zap.so
[Aug 3 11:35:40] ERROR[5518]: chan_zap.c:941
Hello,
My Digium wct4xxp suddenly stopped working. Here are some of the logs:
zap restart
[Aug 3 10:02:55] WARNING[15050]: chan_zap.c:903 zt_open: Unable to
specify channel 1: Device or resource busy
[Aug 3 10:02:55] ERROR[15050]: chan_zap.c:7164 mkintf: Unable to open
channel 1: Device or res
Hello Fellow Users,
I am looking for a way - using certain software or other techniques - to
monitor, measure, and improve the quality of service for Asterisk system.
During the last while, it seems the quality has decreased and am trying to
look for ways to get things going well again.
Thanks,
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