to determine agent's order.
Julian J. M.
On Nov 29, 2007 1:46 PM, Fernando Urzedo [EMAIL PROTECTED] wrote:
Hi,
I would like to implement a queue using a circular strategy, I mean,
using roundrobin or rrmemory strategies. However, I am not able to
define the order Asterisk will call the agents once
that timeout or remove it
completely. Just tell them you have a PBX on that line.
Julian J. M.
On 8/6/07, Alex Pankratov [EMAIL PROTECTED] wrote:
Hi guys,
I spent a couple of hours in Google, but the problem
appears to be uncommon, so I'd like to ask about it here.
The problem is exactly
Freepbx has devices and users concept. It may be what you're looking for.
You can have your users log in in any phone with their extension
number and password. After that, all calls to his extension would ring
on that phone.
http://www.freepbx.org
Julian J. M.
On 8/6/07, Olivier [EMAIL
What kind of switch are you connecting the phones to? I've seen that
behaviour with cheap Repotec switches (24+2Gigabit). Just replacing it
with a different one fixed the problem.
Julian J. M.
On 7/31/07, Tom Lanyon [EMAIL PROTECTED] wrote:
The issues:
Dropouts - by far the most
. There is no real
way it will work on FXO, unless you get an ISDN or all VoIP lines.
Actually some telcos use polarity reversals to signal answer and hangup states.
That's what answeronpolarityswitch and hanguponpolarityswitch
parameters in zapata.conf.
Julian J. M
You are using parameter b in ChanSpy arguments. That will only select
unbridged channels, Zap/73 is connected directly to the meetme
application. Remove that 'b' and try again.
Julián J. M.
On 3/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I have been successful using ChanSpy on a
for i in `seq 100 150` ; do asterisk -rx sip notify polycom-check-cfg
$i ; done
Julian.
On 3/12/07, Mike [EMAIL PROTECTED] wrote:
Hi,
I know that if you have Polycom phones properly configured, you can use sip
notify polycom-check-cfg SIP_REGISTRATION_ID to have the phones download
the new
I'm still using asterisk 1.0.x bristuffed at one site.. Is there
anything similar for this? When both channels are in use, 3rd call
doesn't recive busy signal, but a message fromt he TelCo (something
like The dialed number is not currently available).
Thanks,
Julián J. M.
On 2/3/07, Armin
I don't use chan_capi, but bristuff. http://www.junghanns.net/en/download.html
Julian.
On 2/3/07, Armin Schindler [EMAIL PROTECTED] wrote:
On Sat, 3 Feb 2007, Julian J. M. wrote:
I'm still using asterisk 1.0.x bristuffed at one site.. Is there
anything similar for this? When both channels
My voice is my passport; verify me. ;)
I don't think you'll get reliable results with 8khz sample rates. The
highest frequency wave you can achieve is a 4khz square wave.
Anyway, i don't think if such software exists ;)
Julian J. M.
On 1/19/07, Asterisk [EMAIL PROTECTED] wrote:
Hi all
Use
qualify=3000
For an acceptable lag of up to 3 seconds. That value _doesn't_ mean to
ping the peer every 3 seconds, btw. By default, It will be pinged
every 60s if ok, and every 10s if there is any problem (peer lagged,
unreachable, etc).
Julian.
On 1/4/07, Eric ManxPower Wieling [EMAIL
})
inbound call:
[from-pstn]
exten = _X,1,Set(SIP_CODEC=ulaw)
exten = _X,2,Answer()
Julian J. Menendez
On 10/15/06, Martin Joseph [EMAIL PROTECTED] wrote:
On 2006-10-14 20:00:30 -0700, Julian J. M. [EMAIL PROTECTED] said:
I've finally given up on trying to fax over my Digium TDM400 card.
I've found
:
[telco]
port=1
context=from-pstn
msns=*
Then, in extensions.conf:
exten = _,1,Set(CALLERID(num)=00)
exten = _.,2,Dial(misdn/g:telco/${EXTEN})
Julian J. M.
On 1/2/07, Remco Barendse [EMAIL PROTECTED] wrote:
On Fri, 29 Dec 2006, Julian J. M. wrote:
It's not necessary to recompile
.
Julian J. M.
On 12/29/06, Remco Barendse [EMAIL PROTECTED] wrote:
On Thu, 28 Dec 2006, Gavin Hamill wrote:
On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote:
vzaphfc is not a complete replacement of bristuff. It replies on most of
it. Rather, it replaces the zaphfc subdirectory
Why don't you try app_swift?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Swift
This one even compiles on 1.4, and has buffering, meaning that it
doesn't have to wait for the tts to generate the complete output.
http://www.loopfree.net/app_swift/
exten = s,1,AGI(getinfo.php)
exten
Hello,
Has anyone managed to compile app_nvfaxdetect on asterisk 1.4?
Is there any other way of detecting incoming fax calls on non-Zap channels?
Julian.
___
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asterisk-users mailing list
To
to a value slightly lower than the
router timeout.
Julian J. M.
On 11/22/06, Pavel Jezek [EMAIL PROTECTED] wrote:
qualify=xxx in sip means, consider peer as OK if delay reply is bellow
xxx (ms)
qualify checks (POKE) is every 60s (and is not configurable in sip.conf)
qualify setting in iax.conf
Try forcing asterisk recalculate those costs:
CLI show translation recalc 20
Julian J. M.
On 11/5/06, Avi Miller [EMAIL PROTECTED] wrote:
Hey gang,
I'm hoping someone can help me out here. I've just noticed that on
two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm
getting
Yes, digitmap... If you just want to allow any digit pattern, use this digitmap:
xx.T
x - Any valid digit
. - 0 or more ocurences of previous charracter
T - Default timeout (3 seconds)
Any digit followed by a 3 second timeout will match. You can include
pattern to match * and #.
Hi,
I've finally given up on trying to fax over my Digium TDM400 card.
I've found that fax over VoIP is quite more reliable (at least I can
receive the faxes).
My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes
everyday (just ocasionally), i pretend on using g729, unless a fax
to that of the original call,
and the Local channel will become a zombie and be removed from the
active channels list. This is desirable in some circumstances, but can
result in unexpected dialplan behavior if you are doing fancy things
with variables in your call handling.
Julian J. M.
On 8/5/06
Have you tried CLI show application background ?
exten = s,1,Background(myfile|n)
Julian.
On 7/31/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello friends,
I want to use the background(playfile) application without the channel
being answered. I dont want playback because I would
I didn't test it with a Sipura, but a TDM400. You can check this page
for configuration codes for the F251M.
http://blog.julianmenendez.es/configuracion-fct-ericsson-f251m (In
Spanish). If the SPA-3000 supports detecting polarity reversals,
you'll need them.
Julian.
On 7/26/06, Jon Farmer
it in the logs. According to chan_sip.c, around line 12508:
} else if (!strcasecmp(v-name, localmask)) {
ast_log(LOG_WARNING, Use of localmask is no
long supported -- use localnet with mask syntax\n);
}
Julian J. M.
On 7/22/06, Robert Jenkins [EMAIL
Have you made sure you are also setting localnet in sip.conf?
externip=1.2.3.4
localnet=192.168.0.0/255.255.255.255
Asterisk won't use externip for devices on your local network.
Julian.
On 7/22/06, Robert Jenkins [EMAIL PROTECTED] wrote:
Hi,
I've recently got asterisk running on it's own
BRI ISDN is 2 channels, what would you want to do with a 3rd call?
Julian
On 6/30/06, francesco giuliani [EMAIL PROTECTED] wrote:
Does any boby knows how to manage a 3° incoming call in a BRI ISDN line
by chan_modem?
___
--Bandwidth and Colocation
Check /var/log/http/error.log
Usually, asterisk-stat fails because it tries to use more memory than
allowed in php.ini.
Julian J. M.
On 6/26/06, Chris Earle (CBL) [EMAIL PROTECTED] wrote:
yep
I don't know exactly which things the php-gd is used for, but like I said,
someof the pages work
Hi,
I also remember reading that.. but i'm not sure if it was Digium's
word ;) It had to do with some SCSI and SATA controllers taking
control of the PCI bus for too much time, and causing frame-slips or
IRQ losses on TDM hardware.
Julian.
On 6/5/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Hi,
You can have a look here http://blog.julianmenendez.es/sipura
It's drupal based provisioning system for linksys and sipura phones.
You'll need to register an account to use it.
Basically, you have profiles (linksys na-pap2, sipura spa-3000, etc).
You choose one to create a base
Try adding 'r' to the dial options. According to show application dial:
r- Indicate ringing to the calling party. Pass no audio to the calling
party until the called channel has answered.
exten = 3058472194,1,Dial(SIP/1035SIP/[EMAIL PROTECTED],50, r)
Julian.
On 4/1/06,
I have 2 different instalations with 1 Billion HFC Card (1port), and 1
TDM400. Asterisk 1.0.10+bristuff+florz patch.
Only issue is that you must load all modules (wcfxs, zaphfc) before
runing ztcfg, otherwise nothing works.
Everything works ok, even faxing.
Julian.
On 3/30/06, Chris Earle
Are both protocols enabled? I remember I had to first send an SMS with
the Domo (an analog phone with sms capabilities) before I could even
receive them.
Maybe protocol 1, even if it's implemented, needs to be enabled someway.
Julian J. M.
On 3/29/06, Fran [EMAIL PROTECTED] wrote:
Telefónica
The h323 channels doesn't have any support for NAT. You'd need to
register with a properly configured gnugk for that.
Julian J. M.
On 3/29/06, Alberto Sagredo [EMAIL PROTECTED] wrote:
If you open h323 port and rtp ports, it should work.
Il Neofita escribió:
There is a proble to put an H323
That ATA cannot do 2 simultaneous calls with g729. The second call is
probably trying to use ulaw, alaw or g723. Are you sure any of them
are enabled for that extension?
Julian.
On 3/27/06, Tofik Suleymanov [EMAIL PROTECTED] wrote:
Hello,
How to reproduce this bug (?) :
1. register sipura
For converting email to fax, you have asterfax (http://asterfax.sf.net)
For fax2email, app_rxfax is well documented. Check
http://scottstuff.net/blog/articles/2004/03/28/faxing-with-asterisk
You can also use hylafax (with iaxmodem or chan_fax). It may give you
finer control of incoming faxes.
. Return values:
0 Channel is down and available
1 Channel is down, but reserved
2 Channel is off hook
3 Digits (or equivalent) have been dialed
4 Line is ringing
5 Remote end is ringing
6 Line is up
7 Line is busy
---
Julian J. M.
On 3/10/06, Christian B [EMAIL PROTECTED] wrote
it when
asterisk answers, that would explain your problem.
BTW, is it a pstn line? or a gsm fct? If the later, you need to set it
up for proper hangup detection in asterisk.
Julian J. M.
On 3/7/06, Carlos Prieto [EMAIL PROTECTED] wrote:
Hi !
I have some issues, i don't know exactly if it's
what about this?
[incoming]
exten = DID1,1,Goto(incoming1,${EXTEN},1)
exten = DID2,1,Goto(incoming2,${EXTEN},1)
Julian.
On 3/5/06, Tele Cost Price Reducer [EMAIL PROTECTED] wrote:
hi Zach,
i would use GOTOIF to forward the DID from within the [incoming] context to
the other context. i
You don't seem to have disconnect supervision enabled.
Julian.
On 3/2/06, Warren Burstein [EMAIL PROTECTED] wrote:
[...]
One additional mystery is that I don't know why these calls persist.
When I hang up either of the bridged extension on my test system, the
bridged call ends. When a single
If you can read Spanish, check
http://blog.julianmenendez.es/asterisk-hylafax-iaxmodem
Julian.
On 2/23/06, Bob McDowell [EMAIL PROTECTED] wrote:
I think I'm very close to getting IAXModem and Hylafax going, but my
current inbound hylafax logs show this:
Feb 23 10:09:37.98: [ 3638]: MODEM
if they roll the changes across) with
firmware greater than 1.0.13 (not publically available at time of
writing, due out in October 2005)
I've used that with my GXP-2000, and seems to work ok. I had, however,
to adapt it to my needs.
Regards
Julian J. M.
On 12/17/05, William M. Sandiford [EMAIL
Try removing the Answer() before the Dial... e.g.:
[spa2100]
exten = _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN})
exten = _X.,2,Dial(SIP/netvoice-102)
exten = _X.,3,Hangup
Regards
Julian J. M.
On 12/9/05, George Pajari [EMAIL PROTECTED] wrote:
Eric ManxPower Wieling wrote:
T/t/H/h
No... It applies without problems (just a little offset)
Julian.
On 10/27/05, Giovanni Miano [EMAIL PROTECTED] wrote:
Any problems with bristuff ?
2005/10/26, Julian J. M. [EMAIL PROTECTED]:
You can try this patch
(www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your
You can try this patch
(www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your
telco sends your polarity reversals on answer and hangup.
Julian J. M.
On 10/26/05, Giovanni Miano [EMAIL PROTECTED] wrote:
I've TDM400P with 2 cards fxo and asterisk 1.0.9 + zaptel 1.0.9.2
All works
For hylafax to answer a call, you need to use faxgetty.. Add this 2
lines to your /etc/inittab and run init q to force a reload:
IAX:2345:respawn:/usr/local/bin/iaxmodem ttyIAX
modem:2345:respawn:/usr/sbin/faxgetty ttyIAX
Change the paths according to your system.
Julian J. M.
On 10/25/05
Run memtest86 from the boot menu. You may have faulty RAM. I had the
same problem installing CentOs 4...
Julian J. M.
On 8/6/05, Kumara Jayaweera [EMAIL PROTECTED] wrote:
Hi all,
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success
stories? my Intel 865 M'd+ Intel 3.0GHz
Try Florz patch with your bristuffed asterisk. Better support for
missed interrupts.
Julian J. M.
On 7/22/05, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I tried to install a tdm400P and a monoBRI. I loaded zaphfc and wcfxs
modules and everything seemed allright but linux log shows
Make sure, you include remote office's lan in the localnet directive
(otherwise, they'll use the wan ip address, and that may be the
problem...)
Julian.
On 7/15/05, Peter Osborne [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk for my PBX, I have a remote office that is connected by a
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
To do attended transfers with Snom 360, you need to put the current
call on hold, dial the dest extension, tell him/her something, and
press the Transfer button.
I don't think it'll work with asterisk call parking, though...
Julian J. M.
On 7/12/05, Patrick Friedel [EMAIL PROTECTED] wrote:
OK
Have you tried googling for asterisk e164 ?
Julian.
On 7/13/05, Will Velez [EMAIL PROTECTED] wrote:
Hi my name is Will Velez.
Does Asterisk support E164?
Thanks
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Try using insecure=very in your peer definition. That makes asterisk
not require authentication from your peer if comes from the ip address
give in host=xxx.xxx.xxx.xx directive.
That helped me receiving calls from my sip provider, which had exactly
the same problem.
Julian.
On 7/10/05, Peter
Why don't you just use Dial(SIP/125)??
Or better, if you have your extensions defined in context e.g.
[from-internal], just do:
exten = 9876,1,Goto(from-internal,125,1)
Julian.
On 7/8/05, Mark Edwards [EMAIL PROTECTED] wrote:
Hi.
I have the following line in the default context of all
Where are you located? What's not working is the remote party hangup
detection, and callprogress only works on selected countries.
Please, load your wcfxs (or wctdm) module with debug=1, and check
/var/log/messages to see if the card is detecting polarity reversals
when you answer the PSTN line
I guess the wrong word in the original mail was URGENT...
Julian ;)
On 7/7/05, Michael L Smith [EMAIL PROTECTED] wrote:
Who are you to decide what Information can and cannot be legitimately be
sought here:?
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Asterisk-Users mailing list
You can try insecure=very for your peer (in sip.conf). Make sure, they
don't have to register - host=123.123.123.123 instead of
host=dynamic.
Julian.
On 7/5/05, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
Is there any AGI supported calls authenticated by IP address?
Many thanks.
Newbie
exten = _X.,1,Dial(Zap/1/0www${EXTEN})
That doesn't wait for dialtone, just dial 0, sleep for 1,5sec, and
dial the number.
Julian.
On 7/5/05, Accursio Avona [EMAIL PROTECTED] wrote:
Hi all,
I'm newbe with asterisk and i'm facing with this problem that i'm not
able to solve.
I've to put an
Recheck your zaptel.conf. That's not the correct setup for a T1 trunk.
You need to know the signalling the channel bank uses, and specify the
voice channels (bchannel=1-24), and the signalling channel
(dchannel=25). Those numbers are bogus, as I've never worked with T1
;)
BTW, why are you using
for the TDM and the HFC
Card. They usually launch ztcfg.
2) In the init script, load both modules manually (modprobe wcfxs zaphfc)
3) Issue the ztcfg command
4) Load asterisk
That way it worked without problems.
Julian J. M.
On 6/27/05, David Masure [EMAIL PROTECTED] wrote:
Hi all
I've made a backport of this patch for asterisk stable. You can get it
here: http://www.maxosystem.net/asterisk . The page is in Spanish, but
you just need to download and apply the patch to chan_zap.c. It also
works with bristuff patch applied.
Julian J. M.
On 6/9/05, Neil and Fiona [EMAIL
I guess that's Early Media Connect, i.e., if the phone supports that
(not all do), the channels get bridged just after dial completed, (SIP
183), and what you hear is the remote ring tones (from your telco),
not locally generated (as if it received SIP 180 Ringing).
What IP phones are you using?
I've just checked the download page, and the latest firmware available
is 1.0.1.8. Where did you find 1.0.1.9?
This phone has some nasty bugs, one of them being that the other end
HEARS you after you press the Transfer button and you hear a dialtone.
It doesn't send any message to asterisk so
Isn't it easier to talk to your Telco, and tell them to just ring the
first free line, instead of all 4?
Julian J. M.
On 6/8/05, Erwin Lubbers [EMAIL PROTECTED] wrote:
Hi,
I have connected 4 analog public telephone lines to an Asterisk server using a
Digium TDM400P card and that working
I've used that feature in asterisk HEAD, and it has worked for me (i
needed to apply a little patch for it to work for incoming calls
also), but i also used answeronpolarityswitch=yes. Maybe it's a logic
bug in the code. Try with that option and tell us the results ;)
BTW, it doesn't matter is
Hello,
I've been fighting one-way-audio issues with asterisk and SIP
extensions for some time..., and I want to share with you my findings
;)
My setup:
* 1 ADSL router (Zyxel)
* 1 Asterisk box with private IP, and interesting ports forwarded to it.
* Several extensions, some local
Try this:
1) You're on a call
2) Push a Line button, so that you get dialtone
3) Dial the boss extension #
4) Hey boss, you have a call from XXX
5) Push Transfer
6) You can select which call to transfer (if you have more that 1 on hold)
7) Push transfer again.
Julian.
On 6/3/05, Christian
Are you sure you have context=from-pstn in your zapata.conf for the
fxo channels?
Julian.
On 5/12/05, fhunter [EMAIL PROTECTED] wrote:
I don't think my first posting went thru.
I am trying to set up Asterisk for the first time. I am new to this.
I am using [EMAIL PROTECTED]
I have a
In /etc/asterisk/logger.conf, add this:
full = notice,warning,error,debug,verbose
Then watch /var/log/asterisk/full getting really big ;)
Julian.
On 5/11/05, Anton Krall [EMAIL PROTECTED] wrote:
Guys.
Is there a way to output the same information shown on the console when
invoked as
and hanguponpolarityswitch=yes
in zapata.conf), and asterisk detects it alright.
I've been unable to find an admin manual for this fct (or any of the
250 series). Can someone point me to the manual or just give me the
activation code to dial?
Thanks in advance,
Julian J. M
But that only works when SIP/201 receives a call, right?
What if SIP/201 is making a dialout call, does it show as busy in the
phone's keypad?
Julian J. M.
On 5/7/05, Thorben Jensen [EMAIL PROTECTED] wrote:
Could you please give us some more detail as to what you did, in terms
%
8192 samples in 8192 sample intervals 100.00%
8192 samples in 8192 sample intervals 100.00%
8192 samples in 8192 sample intervals 100.00%
--- Results after 19 passes ---
Best: 100.00 -- Worst: 99.987793
I have yet to try spandsp, but I think i'll work without problems.
Julian J. M
Add some 'w' before the number, i.e., Zap/g0/ww1812121212
Julian J. M.
On 5/4/05, Ronan Eckelberry [EMAIL PROTECTED] wrote:
Does anyone know of a way to put a wait or a pause in a .call file?
When my * tries to make an outgoing call on a Zap channel, it does not
wait for a dialtone. It just
Shouldn't it be: ?
bchannel = 9,10
dchannel = 11
bchannel = 12-13
dchannel = 14
Julian J. M.
On 4/27/05, Thomas Andrews [EMAIL PROTECTED] wrote:
On Wed, Apr 27, 2005 at 11:08:46AM +0200, Thomas Andrews wrote:
I have 2 Billion cards and I can't get the hfc driver to work. I get
this error
%
99.975586% 99.987793%
Thanks
Julian J. M.
On 4/26/05, Steve Underwood [EMAIL PROTECTED] wrote:
Why would you expect a bunch of fax modems to work any better than
spandsp? If spandsp doesn't work reliably your system is very likely broken.
I have had hundreds of complaints about spandsp reliability
Hello Colin,
Did setting the latency timer really helped? What latency do you set
for the rest of pci devices? just 0?
Julian J. M.
On 4/26/05, Colin Anderson [EMAIL PROTECTED] wrote:
2. ZTTEST is a critical metric. I was getting disconnects on about 20% of
faxes until I looked at the output
Make sure you have canreinvite=no in your sip peers definition, and/or
that you pass 't' or 'T', to the Dial statement.
Julian J. M.
On 4/25/05, Tim Pushor [EMAIL PROTECTED] wrote:
Hi all,
I am still unable to initiate a call transfer with the keypresses
defined in features.conf in a couple
I haven't worked with PRI, but could it be related to an invalid callerid?
What about:
exten = _X., 1, SetCallerId(123123123)
exten = _X., 2, Dial(Zap/g1/${EXTEN})
Julian.
On 4/22/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On April 22, 2005 11:48 am, Mark Phillips wrote:
Nothing
]: Image resolution: 7700 x 7700
Apr 19 21:05:23 DEBUG[7768]: Transfer Rate: 9600
Apr 19 21:05:23 DEBUG[7768]:
==
Am I doing something wrong? Is my system doomed? ;)
Julian J. M
Hello,
In FC3, i had to set wctdm options in /etc/modprobe.conf (it may be
modules.conf in other distros):
options wctdm boostringer=1 debug=1
Julian J. M.
On 4/18/05, Ian Pattison [EMAIL PROTECTED] wrote:
2. Low ringing voltage still (~44V AC). I have used the boostringer=1 option
when
You can use (at least in asterisk CVS), this:
Channel: Local/[EMAIL PROTECTED]
then in extensions.conf
[from-internal]
exten = 1234,1,Dial(whatever)
exten = 1234,2,Dial(otherprov)
Not testet though ;)
Julian J. M.
On 4/14/05, Mystery Glitch [EMAIL PROTECTED] wrote:
Can I use the .call files
Just set qualify=yes in sip.conf
On Apr 12, 2005 3:41 AM, Ronald Wiplinger [EMAIL PROTECTED] wrote:
Is there a possible settings for a remote SIP phone, so that a router
will not close the connection due to long time inactivity?
___
Asterisk-Users
if i'm wrong ;)
Julian J. M.
On Apr 10, 2005 6:20 PM, cmisip [EMAIL PROTECTED] wrote:
1. Qos is all about managing upload packets ( and download packets
indirectly by managing upload packets).
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Asterisk-Users
Try:
canreinvite=no
in your sip user definition.
Julian J. M.
On Apr 8, 2005 4:23 PM, Marlène Beray [EMAIL PROTECTED] wrote:
When I call from an IP Phone registered to the Asterisk server, the
connection is established and I can hear what the other person says but this
other person does
I'm having this problem too, with a Swissvoice IP10... No nat between
asterisk and the phone... I don't have any problems with the phone,
outgoing and incoming calls work as expected...
Could it be related to qualify=yes?
Julian J. M.
On Apr 6, 2005 1:39 PM, Eric Wieling aka ManxPower [EMAIL
Create serveral contexts, e.g. from-sip-group1, from-sip-group2, etc...
Then in that context, include the features you'd like for each group,
and give each sip user the correct context.
Julian J. M.
On Wed, 30 Mar 2005 09:30:16 -0500, Matt [EMAIL PROTECTED] wrote:
How would I go about giving
Have a look at http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision
Julian J. M.
On Mon, 28 Mar 2005 11:21:09 +, Robson Ribeiro [EMAIL PROTECTED] wrote:
After the call is finished if the user doesn't press # the line hangs forever
Maybe the first digit is dialed before the dialtone, try adding a 'w'
before ${EXTEN..., e.g.
exten = _91NXXNXX,2,Dial(${TRUNK}/w${EXTEN:${TRUNKMSD1}})
Julian J. M.
On Mon, 28 Mar 2005 13:19:03 -0500, Kellner, Peter
[EMAIL PROTECTED] wrote:
When I dial a long distance number (916503270309
).
Julian J. M.
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the # key works fine
atxfer = *
Attended transfers are only supported in CVS, not 1.0.X
Julian J. M.
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If you want to authenticate by IP, you need to add: insecure=very
Julian J. M.
On Tue, 15 Mar 2005 17:19:17 -, Kanishka Somaratne
[EMAIL PROTECTED] wrote:
I want to enable SIP calls from an ip address, direct calling without
registering, the ip which sends the calls will not change. i
Why not chown to the user asterisk is running under? That way you
don't give write access to everybody. AMP does that.
Julian J. M.
On Sun, 13 Mar 2005 13:31:12 -0600, Matthew Boehm [EMAIL PROTECTED] wrote:
As such, chan_zap is unable to work due to bad permissions. Is it safe to
simply
Try merging both and use type=friend
Julian.
On Sun, 13 Mar 2005 21:07:06 +0100, Pepe Aracil [EMAIL PROTECTED] wrote:
I only can get outgoing or incoming calls work well, but not both.
How can i solve this problem?
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Hello,
I don't know if your zaptel.conf and zapata.conf setup regarding your
isdn is correct, but if you use the default AMP setup, you need to
assign your channels to group 0 for dialing out, and assign it to
context from-pstn if you want to receive calls.
group = 0
context=from-pstn
channel =
Yes, you can, but asterisk needs to be reloaded (sip reload) when your
ip changes.
Julian J. M.
On Thu, 3 Mar 2005 14:57:15 -0500, Giovanni Powell
[EMAIL PROTECTED] wrote:
Can i use a domain name instead of an IP address for externip
(sip.conf) Because im using dynamic dns. Not sure what i'm
canreinvite=no in your sip phones sections, as was
suggested above.
Julian J. M.
On Wed, 2 Mar 2005 23:26:56 +1100, Rudolf Ladyzhenskii
[EMAIL PROTECTED] wrote:
Hi, all
Still trying to get NAT working.
I have following setup:
PHONE 1 -- * BOX
|
NAT
Just add this to /etc/modprobe.conf:
options wctdm opermode=TURKEY
Julian J. M.
On Wed, 2 Mar 2005 18:15:24 +0200, Soner Tari [EMAIL PROTECTED] wrote:
Sorry for littering the maillist, I've found it myself, I've changed the
wctdm.c file and make install'ed zaptel drivers, now it shows
If you really need it, you can create an alias that send that mail to
the addresses you want.
Julian.
On Thu, 03 Mar 2005 07:07:17 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
On Thu, 2005-03-03 at 06:32, Randy Johnson wrote:
Is there a way to send a voicemail to two different email
It can be done with FOP (flash operator panel), which you can download
from www.asternic.com. Also, FOP is included in AMP (Asterisk
management portal) http://amp.coalescentsystems.ca/
Julian J. M.
On Wed, 2 Mar 2005 15:39:32 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
How do those callcenter
=yes
.
.
Julian J. M.
On Sun, 27 Feb 2005 11:04:25 +0100, Michiel van Baak
[EMAIL PROTECTED] wrote:
On 20:52, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote:
Thanks for suggestion.
Unfortunately did not work.
What does this option do anyway?
I cannot explain it as clear
.
.
Julian J. M.
SIP debug shows that phone registers with public IP address of the site,
while calls somehow go to local address.
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