Hi,
I would like to setup Asterisk to route incoming calls to ZAP on my TDM400P
to SIP phones. What is the best dial plan to use.
We are currently able to route outgoing calls to PSTN from SIP to ZAP.
Thanks in advance.
Walid
___
Asterisk-Users
Hi list!
I am able to successfully receive faxes with RxFax using OH323 (0.6.5)
channels and G.711. On the other side I have a Quintum CMS.
However, I don't know how to force the codec before I answer the calls,
i.e., I would like to answer the voice calls with G.729 and the fax calls
with
We've got a EuroISDN (32 channels) with a TE405p, running cvs head as of
5 days ago.
In the past couple of days, we've hit a scenario where incoming calls to
the * pbx from the PSTN are being marked as busy, but outgoing calls
work just fine. When we reboot *, the problem goes away. Has
Hi Geoff Manning,
It's true that Zoom x5v won't work with
Asterisk because only work Voip with Global Village Service ?
Thanks
gottelf at gmail.com
On 6/24/05, snacktime [EMAIL PROTECTED] wrote:
On 6/24/05, Nazareno Pereyra Lima [EMAIL PROTECTED] wrote:
Hi
Some time ago (with previous releases of Asterisk) I had the same problem
with broadvoice, so I added a cron job that reloads the sip every 1 hour.
I know this is not the best solution, but at the time this seed fine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Wed, Jun 22, 2005 at 02:58:54PM -0400, Jerry wrote:
Mike M [EMAIL PROTECTED] wrote:
Think opposite. Green modules are fxs and should be handled with the
fxo signaling. Red modules are fxo and should be handled with fxs
signaling.
Note the red and green colors here:
Hello,
I have Cisco
CallManager 3.3.4 and [EMAIL PROTECTED] latest
version. I have earlier tried getting Asterisk to register with CCM via H323 and
failed. Back then, I learned that this is a known bug in Asterisk. Also people
who tried doing that had also succeeded in getting calls to go
On Wed, Jun 22, 2005 at 06:22:57PM -0400, Jerry wrote:
Hi Alessandro,
But all ports are green!
p1 -green
p2 - green
p3 - green
p4 - green
I think he means the daughter card color, not the LED on the card slot.
What color are the actual daughter cards?
Indeed, I was
It's not just you. Same thing happens here. I went back to 1.0.7.
Stefan Gofferje wrote:
Hi folks,
I used to have some constructions like
exten = number/callerid,1,Goto(somewhere)
After updating to 1.0.8 those does not work any more.
Any hints?
Regards,
Stefan
Use a gatekeeper and have both boxes register with the gatekeeper. That
way you can specify what numbers go where. From everything I have
tested, * will NOT register with CCM. When I added in a gatekeeper and
had both sides register with it, everything works.
Walid Azab wrote:
Hello,
I
Hi
On Thu, Jun 23, 2005 at 05:52:44PM -0700, beonice wrote:
Okay, so what makes more sense:
1) a remote management card that will let me
actually log in to the machine to monitor it as well
as to reboot it
vs.
2) a remote-accessible powerstrip that will allow me
to remotely
Hi, Do anybody knows how to display the original caller's callerid when
transfering a call to another extension on that extension's phone?
Usually the extension who is transfering the call would display as
callerid.
Thanks.
___
Asterisk-Users mailing
On Fri, Jun 24, 2005 at 11:57:17AM -0400, Julio Arruda wrote:
2- Out-of-band is as safe/unsafe as having the conversation recorded,
including pin, by the hacker, if no encrypted voice path is being used.
I haven't given much thought to this earlier, so I hope the following is
not total crap:
It depends really. There's about 3 different ways to send DTMF with SIP. One
is inband, as audio. Another is rfc2833, which is not as audio - but still
goes via the RTP stream as separate packets. The last one is info, which
sends it over the control stream as SIP packets.
- Joshua Colp.
On
Thank you for your reply.
Did moving to a newer version fix the problem?
I am still asking here if it is a problem with asterisk or am I not doing
something right?
Running a cron job every hour just seems way to hokey for something that
as far as I know is supposed to work.
besides that
On Thu, Jun 23, 2005 at 02:57:36PM +0200, René Ott wrote:
Hello,
I am interested in some management-performance issues:
1st Scenario:
A management tool (for example a webbased one) has the following process:
- write in database
- read with script (for example perl) data from db and
You can reload most anything individually, despite not having a CLI command.
You just need to execute reload filename. Example: reload chan_iax2.so
That would reload IAX2... Yay!
- Joshua Colp.
On 6/25/05 1:23 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Jun 23, 2005 at 02:57:36PM
On Fri, Jun 17, 2005 at 03:48:41PM -0300, Christian Callejon wrote:
[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# modprobe wcfxo
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
FATAL: Error running install command for wcfxo
[EMAIL PROTECTED] ~]# ztcfg -vvv
Hello,
we can read in README.odbcstorage
The database name (from /etc/asterisk/res_odbc.conf)
is in theodbcstorage variable in the general
section.
where is general section in res_odbc.conf ?
I think of app_voicemail lookup table voicemessages
but which databases
Harry
update...
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jun 25, 2005, at 12:18 AM, Steve wrote:
Still looking for some help here.
Is this problem due to asterisk, the two week old version of CVS-
HEAD I'm
running?
Or is
Hi Steve.
I think the proxy authorization is just for WWW access(tcp 80 and 443),
if some VoIP port is open you will be able to access your provider
without auth.
Denis.
On 25 de jun de 2005, at 02:22, Steve wrote:
I keep getting asked by people if these types of wifi phones are
capable
I have set up realtime for Asterisk just as the instruction provide. Everything works, except it apearer that SIP devices do not regisert correctly. I can place a call from a SIP device, but not place a call to a SIP device.
If a I use sip.conf everything seems to work.I have not posted all the
On Jun 25, 2005, at 8:58 AM, E Fierro wrote:
Hi, Do anybody knows how to display the original caller's callerid when
transfering a call to another extension on that extension's phone?
Usually the extension who is transfering the call would display as
callerid.
Thanks.
SHOW APPLICATION dial
I want to compile the G729 codec to try it out with firefly.
I don't have Visual C++ 6 compiler. Is there a way I can obtain the link.exe
alone for use with cygwin, or a substitute program?
I don't look forward to installing the whole Visual C++ just for the link.exe
Not always. Some use a www capture page. When you log in through that
page, it opens up that mac/ip for a specified length of time. We're
doing that here using nocat (http://nocat.net) Without logging in, no
traffic goes through from that mac/ip.
- Dan
Denis Galvão - iSolve wrote:
Hi
Ok. You're right.
Denis.
On 25 de jun de 2005, at 15:07, Dan Perik wrote:
Not always. Some use a www capture page. When you log in through that
page, it opens up that mac/ip for a specified length of time. We're
doing that here using nocat (http://nocat.net) Without logging in, no
Daryl Jones wrote:
It's not just you. Same thing happens here. I went back to 1.0.7.
There is definitely breakage in 1.0.8 in this area; please test the
patch below and report back the results here so we can get a new release
made.
diff -u -r1.45.2.2 pbx_config.c
--- pbx/pbx_config.c
Stefan Gofferje wrote:
The patch is rejected here.
My email client ate the tabs; the patch should apply with '-l' to ignore
whitespace changes, or you can manually do it (it's only four lines).
___
Asterisk-Users mailing list
I tried to reload chan_zap.so but it didn't work. Do you know a way how to
reload it?
René
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Joshua Colp
Gesendet: Samstag, 25. Juni 2005 18:26
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users]
In CVS you can reload chan_zap, but not totally... I believe you can't
change the signalling type without restarting asterisk.
- Joshua Colp.
On 6/25/05 3:59 PM, Rene Ott [EMAIL PROTECTED] wrote:
I tried to reload chan_zap.so but it didn't work. Do you know a way how to
reload it?
René
Stefan Gofferje wrote:
Tested:
[incoming_ISDN]
exten = msn/number,1,Zapateller
exten = msn/mymobileno,1,DISA(no-password|client_int_unrestricted)
(...)
[internal_ISDN_clients]
exten = s/isdnphoneno,1,DISA(no-password|client_int_unrestricted)
(...)
Looks fine!
Great, we'll get it into CVS
I have downloaded and tried a number of astcc.agi filesregarding this problem, but I can not get Astcc to bill for calls. If I apply a connection chrage Astcc billsfor the connection charge only, hence I get to make a 10 minute call for the rate of the connection charge only, because Astcc does
Do you have the notransfer and reinvite lines set properly? I had this
same problem with ASTCC but found that if I removed asterisk including
the source and did a clean reinstall it worked suddenly.
Darren
Ade Agbero wrote:
I have downloaded and tried a number of astcc.agi files regarding
I am trying to get an iaxy device to connect to my asterisk box
over the public cloud however
It fails register and I cannot figure out why. Below is my
iax.conf, iaxy setup file and out from iax2 debug.
My iax.conf
[u7403]
type=friend
accountcode=iaxy
host=dynamic
secret=u7403p
We have successfully connect * .9x 1.0.x with CCM 3.3.x and up using
both gatekeeper and no gatekeeper.. Using SIP usually with CCM 4.0 and
up.. With CCM 3.3.x, there is a limitation where the gateway H323 in
your case cannot use IP addresses, so the Asterisk box has to have
correct DNS entries
I am trying to get an iaxy device to connect to my asterisk box over the
public cloud however
It fails register and I cannot figure out why. Below is my iax.conf,
iaxy setup file and out from iax2 debug.
My iax.conf
[u7403]
type=friend
accountcode=iaxy
host=dynamic
secret=u7403p
context=from-iaxy
I have traveled with my laptop and have found all ports except port 80 are
blocked by wayport in hotels. All browsing is redirected to their signup /
authorization page. You can not redirect any packets past their system as I
have tried to do. (not 80, not 443, not 25,110 or any way out in
Hi there
I'm trying to get asterisk to automatically dial my home number and connect
it to my mobile using a a .call file but am having problems bridging the
calls. What I do is to send an sms from my mobile where I have an external
script write out a .call file and place it in Asterisk's
I have Callmanager 3.3(5) linked with Asterisk using H323.
Note that even though I'm using Stable Asterisk from 12/06/05, I am
using H323 from CVS Head 30/09/04. I found that later versions would
have one-way audio problems.
I've also had it working with OH323 but noticed higher audio latency
Obelix wrote:
I want to compile the G729 codec to try it out with firefly.
I don't have Visual C++ 6 compiler. Is there a way I can obtain the link.exe
alone for use with cygwin, or a substitute program?
I don't look forward to installing the whole Visual C++ just for the link.exe
The .net
I am attempting to get an iaxy device to connect to my
asterisk box over the public cloud however
It fails register and I cannot figure out why.
Below is my iax.conf, iaxy setup file and debug output from
iax2 debug.
My iax.conf
[u7402]
type=friend
accountcode=iaxy
host=dynamic
Hi all,
yesterday afternoon, I called through my provider (teliax). but from the
evening, I get this error. (below). then I checked in My Account page ans
support page in teliax. and I saw that they have given new setting (to
another proxy sever). I followed new settings. my Asterisk server is
Hi,
I am building for a
customer an * solution that will use 2 Digium cards.
1 x TE110P (T1 ISDN
PRI)
1 x TDM40B (4 analog
ports, 2 for faxes, 2 for extensions)
The system will be
connected to the PSTN through the T1 ISDN PRI interface. All customer
extensions will be SIP phones
Is there anybody else here that still has anything with Livevoip? They
are down and apparently have no idea when they will be back up. Has
anybody talked to them? I wouldn't care at all if it was not that I
have 2 DIDs that I've been unable to transfer away. :-(
Darren Wiebe
[EMAIL
I have a UK Livevoip DID that is down, and has been for several days.
I'm looking to replace my London DID, low usage but need at least 2
channels and a local London number.
Please email me off list if you can provide this.
J
On 6/26/05, Darren Wiebe [EMAIL PROTECTED] wrote:
Is there
Download FREE
from: http://ipswitchboard.thorben.dk
IPSwitchBoard is
an Operators Panel for the Asterisk PBX. IPSwitchBoard is a FREE Windows.NET
application which gives you:
*
Unattended/attended transfers.
* Park calls and
retrieve/forward them again.
* Organize all
your SIP,
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