Would you believe that I have just stumbled on to that :) Thanks very
much. I still find it confusing that the blinky light blinks when 5608
is rung, but you can't pick up 5608 ;)
Julian
Gordon Henderson wrote:
On Wed, 12 Nov 2008, Julian Lyndon-Smith wrote:
Man, I really feel
Hello,
I have another little problem with my ZAPs channels, in fact, when I
received a call, I heard no sound while in the CLI, sound is played:
-- Starting simple switch on 'Zap/4-1'
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack
-- Executing [EMAIL PROTECTED]:2]
On 12 Nov 2008, at 02:02, Don Fanning wrote:
I have a AS5200 that I have interfaced to a T100P via a T-1 Crossover
cable. I got it where the alarms are all ok/green but I'm unable to
dial out or dial into the AS5200.
Anyone have any suggestions as to where to begin troubleshooting this?
Hi,
How can I pass the following data to te queuelog via ami??
Agent,data.
??
I'm doing this:
Action: QueueLog\r\nQueue: queueprueba\r\nEvent: Login\r\n\r\n
And thath works fine getting the log with the event but I cant find how to
pass the agent and data parameters
Any
I'm trying to get sipusers working with a realtime odbc database on
Asterisk 1.6. We have sippeers working from the database, but need
sipusers to be in a separate table for other implementation reasons.
sip show user test load returns results from the database.
CLI sip show user test load
On Wed, Nov 12, 2008 at 6:44 PM, Sebastian Gutierrez
[EMAIL PROTECTED] wrote:
Hi,
How can I pass the following data to te queuelog via ami??
Agent,data.
??
I'm doing this:
Action: QueueLog\r\nQueue: queueprueba\r\nEvent: Login\r\n\r\n
And thath works fine getting the log
Hi all,
We have just set up trixbox latest with a Rhino r1t1 card, hooked up to
a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a
call from SIP to PSTN all sounds become unintelligible screeching or
static kind of noise on both ends, when we call PSTN to SIP the PSTN
side
On Wed, Nov 12, 2008 at 05:55:29PM +0800, lizhong zhu wrote:
the dmesg shows:
wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got
back 0x01
printk: 13709 messages suppressed.
wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got
back 0x01
Not if I have realtime, I'm inserting and deleting from queue_members table,
so I don't have that info.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins
Enviado el: Wednesday, November 12, 2008 3:16 PM
Para: Asterisk Users Mailing List -
Hi All,
I appreciate any help with this issue am facing.
first of all my topology is as such:
my asterisk box has two callcentric sip accounts on it.
as well as a PSTN line which is connected to asterisk through a Sipura 3102.
now my problem is as such:
I sometimes use my box as an
hello:
thanks for Tzafrir Cohen for dahdi testing.
I installed dahdi-2.1-r3c svn code and asterisk1-6
for testing OpenVox B400P and junghans card. i fund that there is bug (i think)
to dectect NT or TE mode. actually on the board,
i set it as TE mode, but after start wcb4xxp, but
it show the
Hi JR,
Tried with another motherboard: it works!!
Tried with Asterisk 1.2 and 1.4: it works!!
I cannot still believeit works...IT WORKS!!!
Need to make some other tests with other kernels/machines/cards but I'm
sure this is the right way!
Thank you thank you thank you!!! ::-))
Giorgio.
Hi everyone,
Currectly I'm having some troubles to get correct status of my calls throug
ISDN lines, when outbound calls don't get its destination I always receive
NO ANSWER as ${DIALSTATUS} despite the fact I know the target number
doesn't exists or is busy at that time.
Maybe there is
[EMAIL PROTECTED] wrote:
Hi All,
I appreciate any help with this issue am facing.
first of all my topology is as such:
my asterisk box has two callcentric sip accounts on it.
as well as a PSTN line which is connected to asterisk through a Sipura 3102.
now my problem is as such:
I
Hello,
One of our client company is providing hosted contact center solutions with
Cisco IPCC. To keep the Call Recording cost at low, they are planning to use
Asterisk / Free Switch. Can anyone integrate Cisco IPCC with Asterisk for
call recording ?
Regards,
Kashif Naeem
Business Development
On Wed, Nov 12, 2008 at 12:25 PM, Peter Lindquist peter.lindquist.th@
gmail.com wrote:
Hi all,
We have just set up trixbox latest with a Rhino r1t1 card, hooked up to
a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a
call from SIP to PSTN all sounds become unintelligible
This should now be fixed. If you want to force an update, you can do something
like `yum clean metadata; yum update`
Jason Parker wrote:
It apparently isn't built with IMAP support. That would be a bug in my
packaging. I'll see what I can do with it.
Jason Lixfeld wrote:
I'm having some
Hi Dave,
that actually makes sense..
I had probs in figuring out my disconnection dial tone, till the point I
stoped trying to figure out..
so ur right that might be the problem..
thanks for your help ill give it a try :)
best,
Roland
--
From:
On Tue, Nov 11, 2008 at 11:54 PM, Shaun Ruffell [EMAIL PROTECTED] wrote:
If you are a user of the B410P card, and are able, please test these release
candidates in your environment. To test you will need version 1.4.4 or
greater of libpri and version 1.6.0 or greater of Asterisk.
Shaun, this
On Wed, Nov 12, 2008 at 7:31 PM, Sebastian Gutierrez
[EMAIL PROTECTED] wrote:
Not if I have realtime, I'm inserting and deleting from queue_members table,
so I don't have that info.
As am I.
I posted a patch that fixes this, so you could be interested in
keeping it in mind (if not even
Just FYI, this issue has been fixed.
On Tue, Nov 4, 2008 at 3:12 PM, Igor Zamocky [EMAIL PROTECTED] wrote:
http://bugs.digium.com/view.php?id=13645
Thanks Igor, we'll keep an eye on it.
--
exvito
___
-- Bandwidth and Colocation Provided
Investigate the externip= option for your SIP peers to the provider.
Jim Dickenson wrote:
I have an Asterisk server, running 1.6.0, on my home network. It has an
IP address of 192.168.0.201. The server is behind my Linksys router that
does NAT for my home devices. I have a softphone on my
Add nat=yes for your provider [proxy01.sipphone.com] should do the trick.
Qualify=60 is good for keeping firewall ports open.
You could supplement that with externip= in the [general] section but it is
probably not needed.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
c james wrote:
Mark Michelson wrote:
c james wrote:
Mark Michelson wrote:
c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap
Daniel Lynes wrote:
You'll need to lose the double quotation marks in the assignment:
Set(CALLERID(name)=Fred) becomes:
Set(CALLERID(name)=Fred)
If it still doesn't work, then it means that your particular provider
does not support the ability to be able to set the caller ID name, or
I have an Asterisk server, running 1.6.0, on my home network. It has an IP
address of 192.168.0.201. The server is behind my Linksys router that does
NAT for my home devices. I have a softphone on my Mac that is on the same
NATed network as the Asterisk server. It has an IP address of 192.168.0.1.
Would this be part of 1.6.1 release???
Regards
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins
Enviado el: Wednesday, November 12, 2008 5:12 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users]
AST -- FW NAT -- CARRIER
sip.conf
externip=PUBLIC IP GW OF ASTERISK
nat=route
localnet=IP AND LOCAL MASK(ex .192.168.0.0/255.255.0.0)
;Carrier example
[Carrier]
type=friend
host=CARRIER IP
fromdomain= CARRIER IP
context=incoming
disallow=all
allow=g729
canreinvite=no
insecure=very
Good Luck
I have done some large installs where people are going to be in the office,
sometimes out, work from home, it always changes sorta thing..
I have found that setting all device profiles to Nat=yes Just Works
whether they are on the LAN or not and this is even on larger scale systems
with
Steve Totaro wrote:
I have done some large installs where people are going to be in the
office, sometimes out, work from home, it always changes sorta thing..
I have found that setting all device profiles to Nat=yes Just Works
whether they are on the LAN or not and this is even on
sean darcy wrote:
Tried it with and with quotes. Same result - exactly. Works with dummy
variable, doesn't if set in subroutine.
It works fine for me, I use the below:
exten = 3175797960,1,Gosub(get_name,s,1)
[get_name]
;
;* Connect to local
On Wed, Nov 12, 2008 at 4:47 PM, Alex Balashov [EMAIL PROTECTED]wrote:
Steve Totaro wrote:
I have done some large installs where people are going to be in the
office, sometimes out, work from home, it always changes sorta
thing..
I have found that setting all device profiles to
Steve Totaro wrote:
While not taking the time to look, and if memory serves me correctly,
LAN devices appear on the correct ports even with nat=yes. I may be
wrong I will have to double check this when I have a moment.
That is not my understanding from the code.
Also, I am curious -
On 13/11/2008 10:23 a.m., sean darcy wrote:
Tried it with and with quotes. Same result - exactly. Works with dummy
variable, doesn't if set in subroutine.
This is incoming. I'm setting the CID(name) based on the incoming
CID(num). Nothing to do with the provider.
Are you maybe seeing
On Wed, Nov 12, 2008 at 5:25 PM, Alex Balashov [EMAIL PROTECTED]wrote:
Steve Totaro wrote:
While not taking the time to look, and if memory serves me correctly,
LAN devices appear on the correct ports even with nat=yes. I may be
wrong I will have to double check this when I have a
I've replied to two emails in the last two days and haven't seen them yet.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
-- Bandwidth and
On Wed, Nov 12, 2008 at 9:40 PM, Sebastian Gutierrez
[EMAIL PROTECTED] wrote:
Would this be part of 1.6.1 release???
AFAIK yes. It's already in branch.
However you might be confused about when it will be. Digium has
changed numbering for releases, 1.6.1 is next major release, so it
won't be out
Doug Lytle wrote:
sean darcy wrote:
Tried it with and with quotes. Same result - exactly. Works with dummy
variable, doesn't if set in subroutine.
It works fine for me, I use the below:
exten = 3175797960,1,Gosub(get_name,s,1)
[get_name]
Hi Michael, thanks for the response.
Michael Graves wrote:
I have the snom m3 system with two handsets at the moment. My wife
would take issue with your characterisation of the snom handsets as
nice. She thinks that they're too small and the smooth buttons are a
bother. I have no such
Doug Lytle wrote:
I've replied to two emails in the last two days and haven't seen them yet.
Please ignore, I must be getting blind.
I seem to have missed them.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
On Thu, Nov 13, 2008 at 1:23 AM, Doug Lytle [EMAIL PROTECTED] wrote:
Doug Lytle wrote:
I've replied to two emails in the last two days and haven't seen them yet.
Please ignore, I must be getting blind.
I seem to have missed them.
Settinu up bounce upon your post in list settings and then
Hi All,
Have any way to asterisk forward the 487 Request Cancelled in SIP TO SIP
call?
In a SIP to SIP call when the called peer B send 487 to Asterik, Asterisk
return to calling peer A 603
PEER AASTERISK PEER B
| INVITE |
I would be curious to know the circumstances in which a receiving peer
is sending a 487 and what the justification for that would be. A 487 is
usually a response code to verify receipt and execution of a CANCEL
transaction from the other side.
Bruno Rodrigues wrote:
Hi All,
Have any
I think what I meant to imply there but did not say is that Asterisk
considers this a bizarre thing for an endpoint to do as a final response
to an INVITE (let alone one which has already been picked up with a 183)
and understandably produces its preferred feedback to the calling leg
for all
Steve Totaro wrote:
I believe that if you are speaking of code and Asterisk's implementation
of the SIP RFC it is already very borked in many many ways. I speak
from what I see in userspace, real-world, although, as I said, I am
going from memory and could be wrong.
Yeah, I know. But
We have a weird thing happening with the caller id when a call is dialed
to a SIP device registered to 1.4.22. We're preparing 1.4.22 on a
development machine for switch to live.
The callerid number displayed on the SIP device (Polycom or soft phone)
is a full SIP URL, i.e. sip:[EMAIL PROTECTED]
Tzafrir Cohen wrote:
Interesting. This part was originally ifdef-ed out in chan_zap:
http://bugs.digium.com/13786
I get a 404 NOT FOUND on that link.
I'll dig up an older version of that code and compare.
Thanks for the tip.
Jim
___
--
On Wed, Nov 12, 2008 at 6:57 PM, Alex Balashov [EMAIL PROTECTED]wrote:
Steve Totaro wrote:
I believe that if you are speaking of code and Asterisk's implementation
of the SIP RFC it is already very borked in many many ways. I speak
from what I see in userspace, real-world, although, as I
On Wednesday 12 November 2008 18:34:45 Steve Totaro wrote:
On Wed, Nov 12, 2008 at 6:57 PM, Alex Balashov
[EMAIL PROTECTED]wrote:
Steve Totaro wrote:
I believe that if you are speaking of code and Asterisk's
implementation of the SIP RFC it is already very borked in many many
ways. I
To answer my own question after reviewing chan_sip.c.
sipusers has been de-implemented in 1.6.0.1 and doesn't do anything
anymore other than appear in sip show settings.
On Nov 12, 2008, at 9:04 AM, Eric Chamberlain wrote:
I'm trying to get sipusers working with a realtime odbc database on
On Wed, Nov 12, 2008 at 7:58 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 12 November 2008 18:34:45 Steve Totaro wrote:
On Wed, Nov 12, 2008 at 6:57 PM, Alex Balashov
[EMAIL PROTECTED]wrote:
Steve Totaro wrote:
I believe that if you are speaking of code and Asterisk's
Matt Riddell wrote:
On 13/11/2008 10:23 a.m., sean darcy wrote:
Tried it with and with quotes. Same result - exactly. Works with dummy
variable, doesn't if set in subroutine.
This is incoming. I'm setting the CID(name) based on the incoming
CID(num). Nothing to do with the provider.
Are
On Wed, Nov 12, 2008 at 07:33:36PM -0500, Jim Duda wrote:
Tzafrir Cohen wrote:
Interesting. This part was originally ifdef-ed out in chan_zap:
http://bugs.digium.com/13786
I get a 404 NOT FOUND on that link.
But 13786 , that is: http://bugs.digium.com/view.php?id=13786
--
hello, users:
I tried to change to hardhdlc in system. but i still can not make calls. the
port 4 led still can be be on.
=system.conf==
# Autogenerated by ./dahdi_genconf on Wed Nov 12 19:22:36 2008 -- do not hand
edit
# Dahdi Configuration File
#
# This file is parsed
Steve,
Thanks, contacted Rhino and they say it is a card firmware/driver
version issue which is causing the EC to mishandle alaw/ulaw. Waiting
for them to provide me with new firmware/drivers.
No problems can be seen with interrupts, debugging the span, etc. All
looks normal, but all one
From IRC:
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as screeching,
feedback, static, or other useless terms. If users report static
on a system where there cannot be static (all digital, PRI, SIP, etc),
you
Alex Balashov wrote:
Steve Totaro wrote:
I have done some large installs where people are going to be in the
office, sometimes out, work from home, it always changes sorta thing..
I have found that setting all device profiles to Nat=yes Just Works
whether they are on the LAN or not
Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
chan_iax (maybe other channels) translate the protocol specific causes
to a Q.931 hangup cause.
I would recommend you check the doc directory in the Asterisk source
code for channelvariables.txt, but for some reason that I
On Wednesday 12 November 2008 23:15:01 Eric ManxPower Wieling wrote:
Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
chan_iax (maybe other channels) translate the protocol specific causes
to a Q.931 hangup cause.
I would recommend you check the doc directory in the
Still shows that I'm running the same version as I was previously.
On 12-Nov-08, at 2:44 PM, Jason Parker wrote:
This should now be fixed. If you want to force an update, you can
do something
like `yum clean metadata; yum update`
Jason Parker wrote:
It apparently isn't built with IMAP
Eric,
Thanks, I did see that on IRC - and yes the problem is with the EC. Did
try lowering
the gain to no avail. Rhino confirms that an upgrade of firmware and
driver will solve
the problem. Just waiting for my on-site man to be in place to reboot
the server.
//Peter
Eric ManxPower Wieling
Tilghman Lesher wrote:
On Wednesday 12 November 2008 23:15:01 Eric ManxPower Wieling wrote:
Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
chan_iax (maybe other channels) translate the protocol specific causes
to a Q.931 hangup cause.
I would recommend you check the
Hi All,
I have setup asterisk 1.4.22; so far everything good.
Except, I am still searching for voIP phones.
Which grandstream phone should I buy, this is going to be for small
office for testing purposes.
I am on a budget, hoping to find someone here who has some used to
sell or point me in
On Wednesday 12 November 2008 23:44:49 Eric ManxPower Wieling wrote:
Tilghman Lesher wrote:
On Wednesday 12 November 2008 23:15:01 Eric ManxPower Wieling wrote:
Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
chan_iax (maybe other channels) translate the protocol
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