Hello
Can anyone know where may download chan_cornet for interconnection Asterisk
and Hipath via IP
Thanks
Viktor
___
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Hi Chan,
1/ be sure to have correctly inputed your country zone
2/ disable the fax recognition in zapata.conf
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de chan (Alpha
Trilogies Networls)
Envoyé : lundi 27
Short version:
Flash device with latest SIP firmware (currently 1.04)
Set Network (I am using the LAN port only) and SIP config as
expected.
Set Line configuration so that the FXO is hotline to the asterisk
extension you want to ring with incoming PSTN calls (mine is set to
2020).
Set System
2006/2/26, hugolivude [EMAIL PROTECTED]:
I have a bunch of road warriors who I've set up with Xlite clients.
Unfortunately the sound quality has been intermittent at best.
What codec dis you use?? I think xlite support speex, that is the
better codec I've tested when connections are under hevy
Alexander Burke wrote:
At 05:03 PM 02/26/2006, you wrote:
I want to match the user from the users callerid. All users have DIDs.
You probably shouldn't do that for security reasons -- rather, match
them according to the SIP username/password pair they provide when they
register.
Hm,
Hi Victor
Looking for the same answers here too. We are regional distributors for
Hicom HiPath in this part of the world and until now we are still
waiting for chan_cornet to come around. So far we have successfully
interconnected via BRI (mISDN) and PRI (Zaptel) and it works great.
Let's
hi all,
maybe i am mistaken but it seems to me that the HiPath 2000 series is an Asterisk based system.
why am i saying this? because Siemens announce it is a Linux, Open Source system.
so, as i do not know any OTHER PBX Linux- Open Source system rather then Asterisk, does anybody know something
Hi,
if yo are looking a way to interconnect Asterisk with
a HiPath 4000 via IP, so you have 2 ways to do it.
- via oh323 (for HiPath 4000 version 1 and 2)
- since HiPath4000 version 3 you are able to
interconnect using sipQ (SIP Trunking)
--- Viktor Tatianin [EMAIL PROTECTED] wrote:
Hello
Hi again,
i don't think that the HiPath2000 is an Asterisk based
system. AFAIK the HiPath2K is only configurable using
a Web-based tool (no console access). For the moment
the HiPath2K will only be release with CornetIP (HFA).
No SIP (panned in a second step) and unfortunazely no
IAX are
Hi,
I did change the RING parameters to my country, but seems like no
improvement, so how to confirm the ringing frequency than from Telco, any
device to test it out?
Date: Mon, 27 Feb 2006 09:28:15 +0100
From: [EMAIL PROTECTED]
Subject: RE : [Asterisk-Users] Ringing Delay
Hi Chan,
1/ be sure
I'm having a bit of an issue with one of the bargain x100p clones, and
I'm not sure what the right approach is.
My symptom started as way loud offset delayed echo from voip hardphones - PSTN
through the clone card. I played with and then learned everything I could about
echo cancelling, and have
Hi Mickey
Seeing is beliving. Any clues to your claims?
Sstephen
Tele Cost Price Reducer wrote:
hi all,
maybe i am mistaken but it seems to me that the HiPath 2000
series is an Asterisk based system.
why am i saying this? because Siemens announce it is a Linux,
Open Source system.
Hi,
Can someone recommend an IAX provider for US DIDs who will:
1) Accept Canadian credit cards (rules out Junction Networks!)
2) Can do local number porting (LNP)
3) Have great audio quality
I tried Teliax, but the IAX audio quality was terrible - pops and clicks
galore! The Teliax SIP
Hi,
I have a problem dialing a SIP phone which is logged in as different
Astesrik machine from the one I am working with.
I want to call a phone in Another astersik machine in , if it answers,
calling a SiP phone registered in my ASterisk:
My dialplan is:
[mariaSIP]
exten = _1.,1,Wait(1)
exten
is where anyone who knows what is needed to get the pickupexten (*8)
running ?
gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff
I've activated it in features.conf (default *8) and also tested other
extensions
res_features.so is loaded
show features says:
Builtin Feature Default
Hi Stephen,You said that PRI works great. We are using HiPath 3550 and Siemens digital phone which using *11, *97 etc for function keys. However Asterisk uses the the * key plus one or two digits for function keys as well(it is common key combination for functions). So is it any way to
On 2/27/06, amna saleem [EMAIL PROTECTED] wrote:
umm..
Can you please tell me what phone u r talking about??i mean does it support
IAX.
Actually i am sick and tired of my DIAX and want a new IAX phone...
I am using an older version of * like 1.0.3
I hope u will not mind replying to me
It
On 2/26/06, mustardman29 [EMAIL PROTECTED] wrote:
According to this blurb I found on the Asterisk Wiki, it was supposed to be
fixed so it still works after a reload. Your suggestion is all fine and
dandy but does nothing to rectify a server reboot. If phones have to be
rebooted everytime the
- Original Message -
From: Doug Lytle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 23, 2006 9:33 PM
Subject: Re: [Asterisk-Users] mpg123 alternative?
Matt Roth wrote:
We were using the
Chris Stenton wrote:
I've had issues with Native MOH when using IAX trunking and Placing a
caller into a parking slot. Sound is awful. So, for parking I use
mpg123 and everything else I'm using Native MOH.
Have you placed a bug report about this?
No,
I can't always reproduce it.
I know this shouldn't be the place to ask this, but I've just
tried to upgrade my IP600 with bootrom 2.6.2 and SIP 1.5.2 and I'm getting
intotrouble here (I chose not to go to the higher software levels since
there's a warningabout using"secure" links.. I am not trying to
change anything
Hello everyone
The PBX is connected using 4 Line FXO card to the PSTN.
I wish to send calls that come to extension X to an external phone
number, i.e. call the comes from Line1 would go out using Line{2,3,4}.
I wish the user that the extension belongs to him be able to set it.
Can this be done
Greetings,
Complete newbie question so apologies here. I am trying to connect our test
Asterisk server with a number of SIP clients to a H323 PSTN gateway, the
basic connection of SIP Asterisks works a treat however the h323 is causing
problems. Box is a Cisco IP-IP gateway running in
I solved that problem for Polycom phones with the patch at:
http://bugs.digium.com/view.php?id=6047
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If Siemens claims it is Open source, they also
should provide the
download link for the
software...otherwise it wouldn't be OPEN source
- Original Message -
From:
Tele Cost Price
Reducer
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Hello,
I have a request from a customer that I'm not sure how to implement.
They have a Snom-360 as receptionist phone and SPA-941 for all other
phones. They use the SPA-941 DND function when they are away from their
desks, which happens often due to the nature of their business.
They
Hello,
while testing the following scenario, I ran into trouble:
One * box with two AVM active controllers in Point-to-Point-Mode is
connected to another * box with ZapHFC/Quad-BRI cards using bristuff in
NT-mode.
All is working fine, I can call from one box to the other and vice
versa.
But if
On 2/27/06, Darren Ellis [EMAIL PROTECTED] wrote:
Hello,
I have a request from a customer that I'm not sure how to implement.
They have a Snom-360 as receptionist phone and SPA-941 for all other
phones. They use the SPA-941 DND function when they are away from their
desks, which happens
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always works
fine for an unlimitel.ca account.
Someone else has seen this too:
When I try to call from asttapi one number, I get message No one is available
to answer at this time (1:0/0/0). Immediately after that I try to call the
same number from SIP phone (the same one that is used with asttapi) and call
goes true.
What have I done wrong?
This is how it looks on CLI.
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always works
fine for an unlimitel.ca account.
Someone else has seen this too:
Hi everybody.
Just noticed that when calling a mobile phone, Asterisk
doesnt bridge the voice message by telco if mobile is unreachable, but
keeps on ringing till it receives a hangup signal. I think this is due to the
fact that the message is played without the call has been answered,
Rich Adamson wrote:
I find that DTMF does not work reliably if jitterbuffer=on for certain
Can anyone suggest a workaround (other than jitterbuffer=off)?
Might try turning off trunking (assuming you have it turned on) and
test again. Seems a couple of parameters interact and probably has
On Mon, 27 Feb 2006, Karsten Wemheuer wrote:
while testing the following scenario, I ran into trouble:
One * box with two AVM active controllers in Point-to-Point-Mode is
connected to another * box with ZapHFC/Quad-BRI cards using bristuff in
NT-mode.
All is working fine, I can call from one
Hello, sometimes I'm experiencing autocongest error due slow response,
anyone knows, what this means?
Second or third attempt after that happens pass successfully...
this happens ever in fastethernet lan, so no problem with lag in wan
environment,
I'm using idefisk 1.32 on client side (winxp
At first sight there is no problem, your code looks good, no warnings
etc, is just that nobody picks up in the other end. Do you have access
to the other Asterisk server? what does the console shows up? I have
not used the manager with Java, but that does not seems to be your
problem. I guess you
Marco,
Which versions of Asterisk will that patch work with?
Douglas.
-Original Message-
From: Marco Maiolini [mailto:[EMAIL PROTECTED]
Sent: Monday, February 27, 2006 6:36 AM
To: asterisk-users
Subject: Re: [Asterisk-Users] BLF not working after reload
I solved that problem for
Hi
You
may write trunk withdial digits *11 which connect to asterisk via
PRI or BRI
Viktor
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Isaac
XiaoSent: Monday, February 27, 2006 1:44 PMTo:
asterisk-users@lists.digium.comSubject:
Hi,
Probably a stupid syntax problem.. but can't seem to make
these files work for more than the first T1 (ok, if I comment out
the 2nd, 3rd, and 4th span).
Can someone proofread this for me?
Thanks!
==
zaptel.conf
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,0,0,esf,b8zs
On 2/25/06, Anton Krall [EMAIL PROTECTED] wrote:
Does anybody know how to set polycom's default ring volume ? Everytime you
restart a polycom phone, ring defaults to a very low volume setting which is
kind of annoying having to set everytime you reboot.
IIRC, You have to set it in the XML file
I have also issues with jitter over wan (cdma),
I'm trying to debug how dejitter buffer is working (using iax2 jb
debug), but nothing happens/no debug output on asterisk console :-(
is any way how to monitor iax jitter buffer? thx
PJ
___
--Bandwidth
Teehee... no I didn't do any of that.. mostly because it's a feature I
don't use all that often, and at the moment I can't upgrade :) so...
On 2/24/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Matt wrote:
I too have noticed this but received no solution =\ I was running 1.2.0
Did you try
Thanks for pointing that bug out to me BJ. At least I understand what is
going on now. It's definitely not limited to the Polycom Phones.
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Monday, February 27, 2006 3:57 AM
To: Asterisk Users Mailing List -
Can you explain this?
What country?
In this case it's not asterisk but the telco that has to do the Answer.
To every mobile? or just that provider?
On 2/27/06, Francesco Angi [EMAIL PROTECTED] wrote:
Hi everybody.
Just noticed that when calling a mobile phone, Asterisk doesn't bridge the
So why use DND? As far as the phone knows, they are all
internal/external. You should realy look into an asterisk side
extension that will block incoming calls.
On 2/27/06, Darren Ellis [EMAIL PROTECTED] wrote:
Hello,
I have a request from a customer that I'm not sure how to implement.
They
Hi all,
I have just setup automon functionality on my asterisk box and when
trying to activate the feature by pressing *1 on my analogue phone
within the conversation it does not work.
That is strange because with SIP phone it works OK.
Does anyone know what could be wrong?
Thanks,
Ondrej
Hi guys,
Matt gave the advice belowfora way to cause MoH to rewind and play from the beginning for each call that comes in.
However the music doesn't restart.
Here was my first attempt:-
mode=customdirectory=/var/lib/asterisk/mohmp3application=/usr/local/bin/madplay -Q -o raw:- --mono -R 8000
Try pressing FLASH, then *1 and then FLASH again.
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
We are working on a smartDND agi script which will do this. Should be
coming out this spring :)
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, February 27, 2006 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello, I would like apply different call rate (tarif) per outgoing
number (or group of phones, based on prefixes),
I'm playing with astpp, but seems, that this feature isn't available here,
can you recommend any other open-source billing (A2billing, AstBill?),
that this can do?
thank you!
PJ
As a follow on from my last email, it appears that Asterisk restarts the player application if the process terminates.
Does anyone know a way to stop that?
Thanks
Dan
On 27/02/06, Dan Journo [EMAIL PROTECTED] wrote:
Hi guys,
Matt gave the advice belowfora way to cause MoH to rewind and play
Yep, that much I know but do you know which setting to use? Manual doesn't
mention anything.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Wilson Pickett
|Sent: Monday, February 27, 2006 10:12 AM
|To: Asterisk Users Mailing List - Non-Commercial
Hello,
On Mo, 27 Feb 2006, Armin Schindler wrote:
On Mon, 27 Feb 2006, Karsten Wemheuer wrote:
In detail:
When all lines are connected, the first two calls are placed on line 1
(which is on controller 1). The next two calls are placed on line 2 (on
controller 2)
If I'll cut line 2,
C F wrote:
Can you explain this?
What country?
In this case it's not asterisk but the telco that has to do the Answer.
To every mobile? or just that provider?
I too have seen something similar in the past. When calling Verizon
(408-489) numbers, when there is no answer and it rolls over to
On Feb 27, 2006, at 6:09 AM, Dr. Michael J. Chudobiak wrote:
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always
works fine for an
Okay everyone
Im moving away from using sipura 841 phones. Im
starting to test with Polycom IP 501 phones. We plan to upgrade our server to a
dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it.
So my question is will upgrading the IP phones with my
On Feb 26, 2006, at 8:03 PM, Joseph Blake wrote:
We're setting up asterisk at the office (really doing some testing
right now) and it is going to be hosted on a dual G5 XServe running OS
X.
I love it. Glad to hear it. Should be a monster.
We're an apple certified solutions provider, etc.
Thanks Chris and Mike for the great ideas.Richard"Chris A. Icide" [EMAIL PROTECTED] wrote: Or you could skip the overhead associated with an AGI and use thedialplan command availabe after installing asterisk-addons MYSQL.exten = _X.,1,Read(PO-NUMBER,enter-yr-po-num)exten = _X.,2,MYSQL(Connect
Im starting to get a lot of errores from asterisk when transfering calls
from one phone to another:
Incoming call: Got SIP response 500 Internal Server Error back from
What does this error usually mean?
I have exactly the same problem with Polycom phones while transferring...
Strange...
B.
Hi,
I
need to send RTP from asterisk to one IP and signalling to another IP. In this
case, can you help me to arrange that configuration on sip.conf
[]
type=friend
username=
secret=
host=
dtmfmode=rfc2833
disallow=all
allow=g729
Atenciosamente
Diretoria
I'm trying to find a way in
extensions.conf to match ANYTHING dialled, including characters such as
*.
The following works for
numbers...
exten =
_X.,1,AGI(script)
but doesn't catch when someone dialls *
first. I tried this:
exten =
_.,1,AGI(script)
which catches when someone dials
On Feb 27, 2006, at 2:55 AM, Dr. Michael J. Chudobiak wrote:
Can someone recommend an IAX provider for US DIDs who will:
snip
3) Have great audio quality
This is somewhat a meaningless question, as the route from you to the
call terminating service can make or break the quality.
You
[EMAIL PROTECTED] is believed to have said:
Can you explain this?
What country?
In this case it's not asterisk but the telco that has to do the Answer.
To every mobile? or just that provider?
Well,
it's funny because here, now (Italy; Telecom Italia PSTN calling Wind
mobile), I do get the
Nora,
If you have issues with choppy calls, most
likely your issue isnt with your phones or TDM400, but it sounds like
you have some issues with your voip trunks and/or network connectivity issues.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nora
On Mon, 27 Feb 2006, Karsten Wemheuer wrote:
Hello,
On Mo, 27 Feb 2006, Armin Schindler wrote:
On Mon, 27 Feb 2006, Karsten Wemheuer wrote:
In detail:
When all lines are connected, the first two calls are placed on line 1
(which is on controller 1). The next two calls are placed
I'm moving away from using sipura 841 phones. I'm starting to test with
Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but,
for now we have a digium tdm400P with 4 analog lines coming into it. So
my question is will upgrading the IP phones with my existing digium
tdm400 card
Thanks dewey. Any feedback on how to debug
this issue ?
-nora
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dewey Straughn
Sent: Monday, February 27, 2006 11:14
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
I am using an
external voice mail system. I'd like to be able to light the message
waiting light on SIP and SCCP phones. Can someone point me in the right
direction? Is there a manager command or and AGI app that does this.
If not, what would I have to do to interface with * and have the
Martin Joseph wrote:
snip
3) Have great audio quality
This is somewhat a meaningless question, as the route from you to the
call terminating service can make or break the quality.
Sure, but some carriers have problems inside their own networks. I can
optimize the routing to the provider
Douglas Garstang schrieb:
I'm trying to find a way in extensions.conf to match ANYTHING dialled,
Hi,
your subject is probably not correct. You want to catch
anything except h, t, ...?
Maybe you want to get matched the digits and *.
Thus try:
_[*0-9].
This will match any dialed string,
Hi, i have a HT 488 and I want using this like an FXO for Asterisk.
I have find some configuration in the list archive google but my HT
with these config not work.
my sip.conf
[HT-488]
username=400
type=peer
secret=wowowow
qualify=yes
port=5062
nat=no
host=192.168.1.157
fromuser=400
What is your setup? There are a lot of
variables. How many VOIP trunks do you have? What is your Internet connection?
Are you using G.729 for your voip trunks to cut down on bandwidth usage? Anytime
you implement a phone system and are using more then just POTS for calls (IE.
Voip
which doesn't work. So, what exten regex can I use that would catch anything
dialled, or how can I stop Asterisk from executing the AGI script a second
time when I use _.?
I think you can just add an extension h in that context, something like
exten = h,1,Hangup
hth
I am having a couple of (unrelated) problems with my polycom 501.
1. The buddy watch is just not working. It tells me that everyone is
online, whether or not they are.
Here is an example directory entry for one of the peers (whose phone is
not registered).
item
Has anyone done any integration work with Covad's hosted solution ? I
am considering Covad's hosted solution and want to be able to use
Asterisk to develop some other apps. Anyone else tried this ? how did
Covad react. I know they use MGCP.
Another thing, the Cisco reseller rep tells me
I'd like to use the AGI command CHANNEL STATUS to check the status of a
channel. However, the dial() command doesn't return -1 until after the call has
hung up. If that's the case, how is channel status supposed to return statuses
like:
status values:
0 Channel is down and available
1
I'd like to use the AGI command CHANNEL STATUS to check the status
of a
channel. However, the dial() command doesn't return -1 until after the
call has hung up. If that's the case, how is channel status supposed
to
return statuses like:
status values:
0 Channel is down and available
1
MC,
But the channel status command is documented as an AGI command itself.
If you look at http://www.voip-info.org/wiki-Asterisk+AGI, you'll see the
'channel status' command listed there as an AGI command.
I can't post my dial plan, as I don't really have one. Well, I do, and it looks
At 12:07 PM 02/27/2006, you wrote:
Can you explain this?
What country?
In this case it's not asterisk but the telco that has to do the Answer.
To every mobile? or just that provider?
My knowledge of SS7 is limited, but this has to do with opening the
audio path before a call-answered event
I see that you are playing with [EMAIL PROTECTED] How is
it going ? Sorry I have not called you. Been very
busy.
Dovid
--- Tele Cost Price Reducer [EMAIL PROTECTED] wrote:
Manoj,
just look in AMP to Inbound Routing, fill in the
DID, define the softphone
as extension X and send the call to
MC,
I think I worked out that I need to use ${DIALSTATUS} anyway. Don't really see
what 'channel status' is for...
-Original Message-
From: Douglas Garstang
Sent: Monday, February 27, 2006 1:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Howdy:
Does echo only occur on analogue PSTN lines, or can it also occur on PRI and
BRI lines? If so, for the same reasons? This is a part of our consideration
to transition to BRI.
Sincerely,
Brent A. Torrenga
[EMAIL PROTECTED]
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana
Ok, here's a weird one.
I have an AGI script where one user calls another. The call is answered.
Everything is peachy. If the call is terminated by the CALLEE hanging up the
call, then Asterisk returns control back to where the Dial() command left off,
and I can check the return code of
Hello Armin,
Am Mo, den 27.02.2006 schrieb Armin Schindler um 20:23:
On Mon, 27 Feb 2006, Karsten Wemheuer wrote:
Hello,
On Mo, 27 Feb 2006, Armin Schindler wrote:
This is not a bug, just normal behaviour.
chan_capi does not know about the status of the ISDN line, it assumes to
Has anyone done any integration work with Covad's hosted solution ? I
am considering Covad's hosted solution and want to be able to use
Asterisk to develop some other apps. Anyone else tried this ? how did
Covad react. I know they use MGCP.
Don't know anything about them.
Another
Douglas Garstang schrieb:
If dial() doesn't return until after the call completes,
it means the channel status AGI command is a waste of time.
Hi,
you are right, dial will block, so you won't get the channel
status by that method when having an outbound call.
You can use the manager. But
Ok, this one has me stumped. This setup was working fine Friday and now
today it's just stopped working.
Details:
Dell 2850 running Asterisk 1.2.4. Phones are SIP phones (Cisco 7940s).
Timing is done via a WCTDM card (also tried ztdummy.) All traffic in and
out of this box to the PSTN is via
Ooops. This was meant to be sent direct and not to the
list. Sorry.
Dovid
--- Dovid Bender [EMAIL PROTECTED] wrote:
I see that you are playing with [EMAIL PROTECTED] How
is
it going ? Sorry I have not called you. Been very
busy.
Dovid
--- Tele Cost Price Reducer [EMAIL PROTECTED]
I recently upgraded a Cisco 7960 to the SIP firmware, it worked fine without
a call-manager. I just put the SIP firmware and associated config files
in the TFTP directory of my asterisk server so that the phone could pull the
firmware off of my asterisk server via TFTP. It took me about 5
The manual mentions that headset, handset, speaker volume are reset between
calls to comply with some regulation and there is a setting to prevent this.
However it too like the ring volume is completely reset between phone reboots.
The file MAC Address-phone.cfg is where the phone would store
Douglas Garstang schrieb:
...
HOWEVER, if the CALLER hangs up the call, it seems
Hi,
did you try the dial command option g?
I did not neither, but when I understand the voip-wiki right,
it might help you.
Roger.
Voip-wiki page about dial:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Does echo only occur on analogue PSTN lines, or can it also occur on PRI and
BRI lines?
Yes, it can occur on any type of line.
If so, for the same reasons? This is a part of our consideration
to transition to BRI.
It is the result of 4-wire to 2-wire conversion somewhere between
your end
I have a detailed procedure on migrating from locked MGCP state to SIP, if
you get really stuck email me and I will dig it up.
Cory Andrews
Purchasing Manager
++
VOIPSupply.com
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There is an option that you can add to your dhcp server option 150 IIRC.
-Original Message-
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: 2/27/06 4:37 PM
Has anyone done any integration work with
Omar A. Sabek wrote:
Like BJ, I'm sorry you had bad luck Phil. I have been playing with
this phone all weekend, and I have had minor problems. The voice
quality is as good as my cisco and polycom sip phones. I asked a
friend to guess what kind of phone I was talking on and he said it
sounded
Ok, this one has me stumped. This setup was working fine Friday and now
today it's just stopped working.
Details:
Dell 2850 running Asterisk 1.2.4. Phones are SIP phones (Cisco 7940s).
Timing is done via a WCTDM card (also tried ztdummy.) All traffic in and
out of this box to the PSTN is
Hi,
does any know why?
i can make call out with my asterisk and voipstunt
but i can't getcall in on my voip in number
i get rejected.
if i use Sipura without asterisk i get in calls
here is my sip.conf
Just tried it. No difference.
Here's the console output when the callee hangs up:
*CLI
-- Executing AGI(SIP/3254102-bb27, ipt/iptrouter.py|FromOnNetPhone) in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/ipt/iptrouter.py
-- AGI Script Executing Application:
snip
HOWEVER, if the CALLER hangs up the call, it seems as if Asterisk immediately
kills the AGI script. My script seems to terminate immediately and therefore
execution does not continue after the Dial() command.
/snip
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI
Douglas Garstang schrieb:
...
HOWEVER, if the CALLER hangs up the call, it seems
Hi,
did you try the dial command option g?
I did not neither, but when I understand the voip-wiki right,
it might help you.
Roger.
I've used the 'g' option and as far as I can tell it works just
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