[EMAIL PROTECTED] wrote:
Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not very
well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I
suspect a lack of demand. Havng a E1 termintae in your
Gavin Henry wrote:
Dear All,
Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?
Voicetronix has an open sourced solution using their OpenPRI in Hi-Z mode.
http://www.voicetronix.com/open-source.htm#logger
Leo
Phil Menico wrote:
I used autodial to allow a user to make a call by clicking on a web
directory and placing a call file into the Asterisk outgoing
directory. That works perfectly for me.
What if I want to click on the web directory and transfer my existing
call? Is there a comparable
Just curios, does the CS1000 now support RFC2833? Previously, I know the
NRS can only support SIP-INFO.
Leo
Jerry Geis wrote:
Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming
calls just fine. However, using outgoing call files the CS1000 is
hanging up after I answer the
Steve Prior wrote:
I read this story and thought of Allison's prompt to try not to think
about blue eyed polar bears.
Will she be banned from foreign travel now?
I supposed it's ok since blue-eyed polar bears are fictitious and thus
protected by the first amendment :)
Leo
I, too, have heard about that best practice of using different
channels for different AP's on the same SSID. As far as I can tell,
This is standard textbook stuff. Read Cisco press's 'Deploying License
Free Wireless Wide-Area Networks' by Jack Unger.
it's BS. I don't know who started it,
Karsten Wemheuer wrote:
Hello,
Am Donnerstag, den 15.02.2007, 10:55 +0800 schrieb Leo Ann Boon:
1. The smallest mini-ITX case I found that accepts a PCI card is the
Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know
if it fits? I didn't find its width, and apparently
Bruce Reeves wrote:
In my experience having ap's with the same SSID and 3 channels of
separation overlapping worked if the phone could roam.
Recommended is 5 channels of separation.
Ronald,
Just be aware that even if the phone supports AP roaming, there's no
guarantee that the call will
1. The smallest mini-ITX case I found that accepts a PCI card is the
Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know
if it fits? I didn't find its width, and apparently, the C138 will not
accept a PCI card bigger than 17,52cm.
The C137 can fit 2 TDM400P with the
Matt wrote:
Eric,
I understand what you are saying about APIC... and from my
understanding the O/S takes over control of the IRQs.. but aren't
there still only 15 physical IRQs that you can set in the BIOS for
devices? I've never seen a machine in which I could go above 15 for
a device in
Matt wrote:
Leo,
Yes I did read this. And I have ACPI turned on. Unfortunately lspci
-vb still is showing devices sharing IRQs.
You mean IO-APIC? ACPI is a different beast altogether. lspci -vb and
lspci -v should show different results on a proper IO-APIC system.
lspci -vb shows what the
Matt wrote:
Leo,
I am sorry. Yes I mean IO-APIC. So basically the output of lspci -v
are the same as cat /proc/interrupts.
It is a riser, I will check on that.
So here's my questions then. If APIC routes the IRQs to 1-15 for real
world usecan you safely have two devices on, say,
Matt wrote:
I guess the question is... is it even possible to have a real-time
VoIP card running on PCIe? Or with 1,000 Interrupts a second.. does
it simply need to have its own IRQ?
Have you tried the Sangoma PCIe cards?
APIC is supposed to fixed the PCI IRQ problem. AFAIK, APIC is not a
I don't know anything about a line simulator but your description
certainly points to a problem with the simulator. As I'm also doing
tests on X100P, I'm interested to know what does a simulator give you
that your PBX doesn't. (I wish I had a PBX to play with.)
How about just using a
Klaverstyn, David C wrote:
Hi All,
I cannot get my TDM to work correctly.
In my /etc/zaptel.conf file I have
loadzone = us
defaultzone=us
fxoks=1
Shouldn't this be fxsks if you're using an FXO module as analog trunk?
Leo
___
Alyed Tzompa wrote:
Had the same issue time ago, but Eric shed good light on it, have a
look at:
http://lists.digium.com/pipermail/asterisk-users/2006-November/172079.html
Summary: sorry, no nice work around.
At least, not in the analog TDM world. Personally, I'll advise everyone
to use ISDN
Klaverstyn, David C wrote:
Hi,
Yes it should, I have changed it back and is still causing the same
problems.
Did you also missed out the following line in zapata.conf?
signalling=fxs_ks
Leo
___
--Bandwidth and Colocation provided by Easynews.com
Klaverstyn, David C wrote:
Yes, I have also since put that in and I get the error:
Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
signalling
And if I put in rxwink I get this error:
Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
rxwink
It's all very
Klaverstyn, David C wrote:
My original post does have the contents of the file exactly.
In my /etc/asterisk/zapata.conf file I have
[trunkgroups]
[channels]
context=from-pstn
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
Yuan LIU wrote:
Kind of do. There are times when it feels like trying to fit two
spinning wheels, though:-)
'Zee trick to fit two spinning wheels is to stop the wheels :)'. That
why, your first working system is the most important. It's easier to
built on once you have a solid foundation.
Yuan LIU wrote:
I remember a thread similar to this a while ago but couldn't find.
How do I make Asterisk to interact with an IVR? (Nothing fancy, just
plain predictable voice menus like a conference bridge.) I get stuck
at Dial(), which seems to wait for hangup after the other end picks up.
Yuan LIU wrote:
Another dumb question: Can a dial plan continue after local hangup
when using Dial()? For example,
[incoming]
exten = s,1,Dial(Zap/1)
exten = s,2,Congestion()
exten = s,3,Hangup()
---
Asterisk seems to insist that a dial plan is complete when Zap/1 hangs
up and do not go into
Stephen Bosch wrote:
The reason we have these complaints is not because Asterisk doesn't
detect the drop -- it's because a great many telephone companies don't
do remote party disconnect signalling, or they don't do it properly.
When people call for technical assistance they usually end up
Eric ManxPower Wieling wrote:
Leo Ann Boon wrote:
Eric ManxPower Wieling wrote:
You should not have quotes in Caller*ID info. MOST devices will
just ignore the quotes, but a few will refuse to accept Caller*ID
with quotes in it. At least one revision of SIP firmware for Cisco
phones does
Yuan LIU wrote:
From: Leo Ann Boon [EMAIL PROTECTED]
Yuan LIU wrote:
Related to callerid: I can't get text ID to work in an analog phone
on FXS. I tried the above format, it simply displays the entire
string in both numeric and text field (i.e., displays the same
string twice). Tried a few
Stephen Bosch wrote:
snip
...and have zillions of dollars :)
Industrial PCs are pretty expensive.
Over here, they're actually quite reasonably priced. A 2U rackmount P4
D930 3.0GHz, 1GB RAM system with 4 PCI (32bit) slots starts around US$1K.
Leo
Good question. Anyone knows if the TDM-400 actually detect loop drops?
Well, that's really what kewlstart (and loopstart) means. If it
couldn't, then Asterisk wouldn't know that the call had been hung up,
and hog the channel.
For loopstart lines, I don't think Asterisk detects loop
Shivram u wrote:
Hi,
An incoming call is redirected to another number by our asterisk
server. In the incoming call the caller name is present but when
redirect the call, the end receiver is not able to see the callerid
name. The caller id number is visible.
If you're calling PSTN, caller id
Alessio Focardi wrote:
Hi,
I'm looking for an hardware platform for an * installation that should
have at least 3 PCI slot with no irq sharing whatsoever.
Use an industrial PC with a backplane bus. You can easily get 3-4 usable
slots in a 2U and 10-14 slots if you use a 4U.
Leo
Yuan LIU wrote:
Related to callerid: I can't get text ID to work in an analog phone on
FXS. I tried the above format, it simply displays the entire string
in both numeric and text field (i.e., displays the same string
twice). Tried a few other ways, got varied results (some resulting in
Eric ManxPower Wieling wrote:
You should not have quotes in Caller*ID info. MOST devices will just
ignore the quotes, but a few will refuse to accept Caller*ID with
quotes in it. At least one revision of SIP firmware for Cisco phones
does this.
Thanks for the heads up. On the other hand,
Stephen Bosch wrote:
Hi, folks:
Can the loop drop detection threshold (normally defined in milliseconds)
be set on the Digium TDM-400 cards? Most PBXs let you set this value.
Good question. Anyone knows if the TDM-400 actually detect loop drops?
Leo
Kyle Gordon wrote:
snip
Hi Leo,
That appears to have done the trick. fxs_ls does seem to detect it hanging up
more reliably. I don't know what the difference is, but it works :-)
If there's any change, I'll be sure to let you know :-p
No problemo. Glad to know it worked for you. Like
Shane Spencer wrote:
I am very interested in the DACs capabilities of Digium cards, there
is no information anywhere on this. I could always do pri bridging
via libpri like you suggest however. But having hardware handle the
bridging onboard a single PCI card would help reduce my server
Shane Spencer wrote:
I wanted to know if there was a peekaboo factor to it all. You can
flow data under a glass window :)
Well - you can always use a logic probe :). Bridging does add a little
latency to the whole thing. Why don't you consider a passive tap
solution like the hi-z OpenPRI
chester c young wrote:
On a SIP phone is it possible to enter the dialplan when the user
picks up the phone without having to wait for the user to press an
extension?
You need a phone with a hotline function. Consult your phone's user manual.
Leo
Shane Spencer wrote:
I am trying to do a wire level tap on T1 equipment using digum
equipment. So far most call monitoring hardware for call centers try
to stay on the analog side requiring a lot of rewiring. I have
already posted to the list about T1 bridging using DAC's support in
the zaptel
It is, and is identified by wcfxo as a Wildcard FXO: Wildcard
X100P. So much for The DigitNetworks X100P is detected as an actual
X101P card.
IIRC, there were 2 Digium single FXO cards - the X100P using the
Motorola SM56 and the X101P with Intel/Ambient 537. The X101Ps have 2
RJ-11 jacks.
[EMAIL PROTECTED] wrote:
Hmm. Nope. Still same thing. I added pedantic=yes both in the general
context in
sip.conf and in the user's context in sip.conf with no change. Just for fun, I
also
changed it to pedantic=no in each place with no luck either. (I stopped and started
asterisk
[EMAIL PROTECTED] wrote:
Thanks for the info, is there a patch available for version 1.2 that adds
the autofill option?
Gavin Hamill has back ported some of the 1.4 queue features into 1.2.
See his post to this list
[EMAIL PROTECTED] wrote:
Here's the debug output from the console, it's somewhat long. Could the key line be
(towards the bottom) this?
[Jan 28 20:39:10] NOTICE[31924]: chan_sip.c:13519 handle_request_invite:
Nothing to
pick up for OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
Ah ha -
Yuan LIU wrote:
From: Leo Ann Boon [EMAIL PROTECTED]
It is, and is identified by wcfxo as a Wildcard FXO: Wildcard
X100P. So much for The DigitNetworks X100P is detected as an
actual X101P card.
IIRC, there were 2 Digium single FXO cards - the X100P using the
Motorola SM56 and the X101P
Yuan LIU wrote:
A little googling made me realize that Asterisk demo may not be the
best application to look for caller ID because it tries to pick up at
first ring. So I zapped demo context with a plain one. This time, no
more failed success. But Asterisk only receives
New User,
no
Kyle Gordon wrote:
fxsks=1 #X100P
Is your line truly a kwelstart line? try fxsls
SNIP
busydetect=yes
You may need to add these 2 values to help the busydetect
busycount=3
busypattern=375,375
busypattern tells asterisk how your busy tone sounds like, in UK it
should be 400Hz 0.375s ON and
Tzafrir Cohen wrote:
On Sat, Jan 27, 2007 at 07:40:31AM +0800, Leo Ann Boon wrote:
Kyle Gordon wrote:
fxsks=1 #X100P
Is your line truly a kwelstart line? try fxsls
And if the line is ls, indeed, what harm is there in setting it up as
ks?
I understand ks is ls
Charlie Grosvenor wrote:
Yes the line is connected, a standard phone works fine when connected to
the line.
There're 2 ports on the card. Which port are you using? One of the ports
is for connecting another phone in parallel to the card.
Leo
___
Kyle Gordon wrote:
Hi all,
I'm currently on an NTL PSTN line, using Asterisk 1.2 and a X101P
cheapo card.
The problem lies with detecting when the far end has hung up. It fails
to detect it, and will only cleardown when the silence timeout has
been reached. Now, I've seen the thread at
snip
zaptel.conf
---
loadzone=uk
defaultzone=uk
span=1,1,1,ccs,hdb3,crc4,yellow
span=2,0,1,ccs,hdb3,crc4,yellow
I don't think yellow alarm is necessary unless you've been advised by
your carrier.
bchan=1-15,32-46
dchan=16,47
bchan=17-31,48-62
Kong Zhen Shin wrote:
i tried without yellow as well.. and according to zaptel drivers, the
yellow don't do anything, just put a yellow signal where there is
nothing from the provider.
and yes, i did put a pri_net on the span 2, the config is a typo..
thanks for reminding me..
but still i
C F wrote:
1. When they tell you that they are putting all your lines in a hunt,
it realy is not a hunt but just CallForwarding No Answer/Busy, what
Some PBX implement line hunting that way. So, you need to Answer before
you do anything else. Otherwise the PSTN switch will cheerfully go on
C F wrote:
On 1/22/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
C F wrote:
1. When they tell you that they are putting all your lines in a hunt,
it realy is not a hunt but just CallForwarding No Answer/Busy, what
Some PBX implement line hunting that way. So, you need to Answer before
you do
Andrew Joakimsen wrote:
Most of the Cisco phones sold cheap are UNLICENSED (global spare)
thus you would not be able to purchase (or at least aren't supposed
to) the smartnet contracts, you need to buy the license ($100+) and
the contract ($10 or so)
I'm always surprised by by the number of
Voip Asterisk wrote:
I know that NAT is something no one really likes to talk about, but
does anyone know how work with it elegantly? There are many providers
which deal with it on a daily basis in fact they cater to it, is this
possible to do with asterisk or does it require other exotic
Antoine Fressancourt wrote:
I will sum up the results of my investigations :
- When canreinvite is set to yes, I manage to make a video call
between the 2 parties, when I emit a DTMF signal, it triggers the
playback of a sound clip correctly, but I can't playback a video clip.
What's the
Andrew Joakimsen wrote:
I have some Audiocodes units which appear to be running Linux,
according to the unit's own System Log
kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
Googling turns up:
http://www.jungo.com/openrg/openrg.html
OpenRG is a Linux based device
Antoine Fressancourt wrote:
Hello,
Thank you Leo for your answer,
I manage to do what I want perfectly when both the caller and the
callee are set in SIP with canreinvite=no using SIP INFO method for DTMF.
Now, I can't figure out why this can't work when I set canreinvite =
yes with the
3. It seems to be only incoming calls that have an echo and only on the
inside, the outside never hears one, what does this mean?
Why don't you record the call at asterisk? Leave the zaptel settings as
default, i.e. standard echo cancel and rxgain=txgain=0. Don't use
MixMonitor, just leave
Eric ManxPower Wieling wrote:
*sigh* Any time a call hits an analog 2-wire circuit there will be
echo. In normal PSTN only situations the echo is so FAST that you do
not hear it. It is only where there is a high latency path in the
circuit like a VOIP phone where you will hear the echo.
Colin,
Thanks mate for the first laugh of the day.
Colin Anderson wrote:
I got a requirement list just now, with my comments inline: (showing
it just for a giggle)
User requirement: 1) Directory set up by name - If person calling does
not know employee's name, how will they access?
exten = 1234,1,Dial(SIP/1234)
exten = 5678,1,Dial(SIP/5678)
The SIP phones (X-lite) are configured to send DTMF's using RFC 2833
mechanism.
I want to know if it is possible in Asterisk to catch a DTMF event
sent by one of the phone to trigger an action, for example to play a
sound/video
David Thomas wrote:
This is by far the most volotile list I have ever been on. I'm not
sure that's exactly the reputation Digium/Asterisk is shooting for,
but even so it does provide some much needed comedy relief.
Alas, it was't even related to the OP's problem. He was just trying to
figure
Erick Perez wrote:
The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx.
Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw
phones +voicemail and *no* call recording?
Make sure there're no interrupt sharing issues. My old EPIA 1GHz with 2
LAN and 6 USB, had
Xue Liangliang wrote:
Hi, all
I am a pabx vendor from Singapore. Recently we are going to implement
a failover solution for our customers using heartbeat, the asterisk
server can failover perfectly, however the g729 codec canot work,
because it is binded the mac address, we have bought two
Xue Liangliang wrote:
Hi, actutally it is kind of shareing storage, because we use drbd and
vserver technology, the fail over is at vserver level, and vserver is
synced through drbd storage.
drdb - that's what I suspected. Off the top of my head, the fastest way
is to reactivate using the new
I have followed the (very brief) instructions on voip-info.org titled
Asterisk Call parking. Basically, I confirmed that features.conf was
already set up properly, and made sure parkedcalls was included in my
local context. If I dial in via the FXO and answer the call on x102,
then hit
Erick Perez wrote:
what if I go with full g711-no transcoding?
remember that I will have an E1 coming in, so my usage can be up to 30
channels at once.
if that is an overkill machine config, and for obvious reasons I cant
use old hardware, what are your suggestions?
I would suggest you go for a
Lenz wrote:
You are correct, this is more or less the scenario involved - the
problem is that people want to call a personalized line AND speak to
the same subset of agents preferably.
I have never seen such a setup myself - I have seen CCs with 30 or 40
queues, never 200 - so I was
lenz wrote:
HI Gavin,
wish we could do that! :) the problem is that they want to have
personalized agents too - so that each client has its own line AND his
own agents, so that they get back to speaking to the same people all
of the time. SO we need many different queues to accomodate all
Troy - Purple Oranges wrote:
Hi all and Happy New Year.
I have a couple of interconnected asterisk boxes connected to several
providers. With one provider in particular (ATP in Australia) there
are two ringing tones heard on outbound calls. It is not the end of
the earth - I am not reselling
Lee Jenkins wrote:
Moises Silva wrote:
use agi debug command from the Asterisk CLI to see what is going on.
Also, the last time I checked, \n is needed at the end of any
command sent to Asterisk.
Regards.
Hi, sorry I have already done that, but did not mention it. The
output that is
Eric ManxPower Wieling wrote:
Leo Ann Boon wrote:
Phil Finkler wrote:
Hi all,
I’m trying to incorporate using the i extension in my callplan to
determine if someone enters an invalid extension. My internal
extensions are all 3 digits (100-104). The problem is, the callplan
doesn’t see
Eric ManxPower Wieling wrote:
Leo Ann Boon wrote:
Hi all,
I'm using 'show translation' to help dimension my system, but I
confused by the results I get. My 2 test systems (results below): an
AthlonXP 2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz)
produced similar results (D930
Vicky wrote:
I tried it on a intel 3 ghz p4 box and a athlon 3000 768 mb ram
running vista and host for centos 4 ( vmware ) considering the load
on athlon running asterisk ( that too under vista plus vmware ) while
intel 3 ghz p4 1 GB ram box was sitting idle with centos , there was
hardly
Phil Finkler wrote:
Hi all,
I’m trying to incorporate using the i extension in my callplan to
determine if someone enters an invalid extension. My internal
extensions are all 3 digits (100-104). The problem is, the callplan
doesn’t see that say, extension 600 is invalid, it just goes back
Tzafrir Cohen wrote:
If you had just one call, then adding extra CPUs wouldn't have helped.
'show translations' mainly helps you compare different codecs. It is
also handy as a benchmark because it's there. However
I agree with you that with 1 call, more CPU won't help. I'm just
surprised
Conrad Wood wrote:
On Thu, 2006-12-21 at 12:07 -0700, Douglas Garstang wrote:
I'm no C programmer, but is this 32 limit just an array definition somewhere?
Wouldn't it be a no brainer to track it down and increase it so some very large
number?
I think pickupgroup is defined as
Hi all,
I'm using 'show translation' to help dimension my system, but I confused
by the results I get. My 2 test systems (results below): an AthlonXP
2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar
results (D930 is slightly faster). Googling shows that others have
Douglas Garstang wrote:
I just know someone is going to ask 'why would you ever want to do that?'.
Here's my answer.
We have two companies, each with a dialplan similar to what's below. In the event that the
number being dialled does not match any number within our OWN company, we want to set
yusuf wrote:
Hi,
I just got hold on an Orion E1 30 port GSM Gateway, and I am having
problems trying to get the E1 link to come up. I am using Asteisk
1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both
the Digium and Samgoma types, as I have successfully hooked up to many
Jesus Mogollon wrote:
Hi all
Does anyone know of any motherboards with PCI slots that can take
the TE412P card? Is there such a MB for Athlon 64 or P4 procs?
I have a TE410P working with an ASUS P5MT mobo with Intel Pentium D
processor.
___
Matt wrote:
I see that the digium card doesn't share the IRQ however Digium
has recommended diabled USB still... additionally the Digium card is
on 169 which isn't a valid IRQ.. how can I find out what it is sharing
with?
the tdm card is not sharing an interrupt with your USB. It's your LAN
Matt wrote:
So you are saying that the card is on it's own IRQ and is not sharing
anything with anything? I realize the eth0 and usb are sharing, but
am not too concerned about that.
What's your zttest result and did zttool reported any irq misses? If
zttest is mostly 99.98%, then the zap
Dovid B wrote:
- Original Message - From: Leo Ann Boon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, December 08, 2006 12:07 PM
Subject: Re: [asterisk-users] Running Asterisk on a Home rotuer
Dovid B
Julian Lyndon-Smith wrote:
snip
this works well, with one exception: when I take the call on the
mobile, the callerid info is the number of my switchboard. I presume
that this is because I am dialling out from the switch board.
Enter RDNIS. I added an extra line to the dialplan
snip
2
Dovid B wrote:
tacking pn = adding on - sorry for not being more specific.
I have seen that people in the past have used a linksys router to run
asterisk. It would be to expensive to bring in a PC for every
location. So we want to import cheap home routers put asterisk on
them as use them as
Michael Iedema wrote:
Greetings list,
Does anyone have any information (providers' support) about G.729E?
Voip-info.org came up empty, the implementers guide from the ITU wants
my credit card and the rest of the pages I found simply made a few
comparisons between it and iLBC.
From what I
snip
I have tried setting another variable as a counter with some logic
tests to see the number of attempts to call the agent, but this is
failing as the variable appears to be lost when the call goes back to
the queue.
Local variables are destroyed once the call terminates. You'll have
yusuf wrote:
Hi,
I realise this might be an insane noob question, but I'm on a huge
brain freeze, and I'm trying to decide this:
Is Asterisk a SIP Gateway or SIP proxy?
Short answer: Gateway.
This has been discussed to death many times on this list. Please search
the archive for more
Brad Templeton wrote:
snip
My understanding was that the port= field on a particular SIP
channel defines the port used at the remote end, ie. The
user's phone will be talking on port X of their IP address, it
does not alter what SIP port Asterisk is listening on on the
Asterisk box.
The
Noc Phibee wrote:
Thanks Giogio,
but no i don't have this module
bye
Check your zapata.conf. Your signalling and channel settings are wrong
for FXO module.
signalling=fxs_ls
channel= 4
FXO module use fxs signalling, FXS module use fxo signalling.
Leo.
Noc Phibee wrote:
thanks for this information, but no change:
Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4:
No such device or address
Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No
such device or address
here = 0, tmp-channel = 4, channel = 4
Jesus Jimenez wrote:
Hi ,
I have a problem with a X100, i do a external call to the
asterisk server . The dialplan its simple answer and hangup..
when it's done , the telephone which i did the call , is in line but
asterisk server is finish.
I'll apreciate all your suggestion.
Pavel Jezek wrote:
is possible to control ci$co gateway from asterisk via mgcp? i.e.
asterisk as mgcp call agent?
PJ
I've tested the old Cisco ATA-186 MGCP (firmware 2.16) with Asterisk
1.2. Works pretty well.
Leo
___
--Bandwidth and Colocation
Marcus Franke wrote:
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Hash: SHA1
Michael Welter wrote:
Has anyone tried recording to a ramdisk? To an NFS mount? Was there a
benefit?
RAM disk? Interesting idea, but what to do in case of a server crash
loosing these recorded files?
Or use
kjcsb wrote:
I have Asterisk listening for sip traffic on port 5060. I want to
allow users to use either port 80 or 5060 if they want. Hopefully
this will avoid some firewall issues.
If you're think that by sending SIP on port 80 will fool the firewall
into thinking it's HTTP traffic,
Steve Davies wrote:
*bump*
No suggestions at-all? Does anyone use this facility in a similar way
and NOT have problems?
Check the gain on your ISDN interface. The monitor command doesn't
modify the volume by default. Have you tested calls via IAX to your cell?
Leo
Evert wrote:
Hi! :)
Thanks for the tip. I'm almost there now, the only problem that I have
left is that I do NOT want Asterisk to check whether the extension
entered is valid. In the current setup Asterisk will refuse to forward
the call since it thinks the extension is invalid... :-/
Is
Zeeshan Zakaria wrote:
By default asterisk install rtp.conf with following settings:
[general]
rtpstart=1
rtpend=2
I usually change rtpstart to 10001 so 1 can be used for webmin. On
some servers I keep rtpend on 14000 (no
You should stick to even numbered ports. For each even
Zeeshan Zakaria wrote:
I'll keep that in mind for future. I read about using 10001 as start
port on Nerd Vittles website.
Is there some good material online to read more about RTP, SIP, RTCP
and UTP?
Search the RFCs.
Leo
___
--Bandwidth and
shadowym wrote:
Just to follow up on this,
After some testing tonight I found the following. Watching the Asterisk
CLI, when making a call from an extension to a ZAP channel the channel shows
as answered as soon as the zap line starts ringing. That would explain
why Followme was not working.
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