All Polycom phones use the same firmware and bootroms - one reason why
the sip.ld is so damn large for them.
Thanks Rob.
Alleluia! Rob, I will take your word for it - it solves all my worries
in deploying different models to the same environment like IP5XX and
IP6XX.
Can't you just use the same bootrom for all your polycom phones?
PaulH
On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote:
I have a question about DHCP and boot server supporting more than 1
model of Polycom phones.
According to Polycom standards, Polycom phone boots up to get a
Dear all,
I am still not able to store voicemail into mysql and I am hoping someone
can help me out.
Here is my voicemail.cof:
[general]
format = wav
attach = yes
dbuser=ast
dbpass=sqlpass
dbhost=localhost
dbname=asterisk
odbcstorage=asterisk
odbctable=voicemessages
[default]
; Syntax for new
I have to chime in here to say that we have had an IAXy for four years
and it has given flawless service. Yes, it has no features like DNS
but we haven't required this. It's small and easily hidden in a home
or soho scenario and I've also used on the same network as the
asterisk box or on a
Hello List!
We're having trouble making call deflection on ISDN PRI. We would like to
transfer a call to an external extension but keeping the callerid of the caller
so it can be presented to the receiver of the transferred call.
At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
What you are looking for is something like this piece of code. Adapt it for
your scenario:
[default]
exten = _.,1,NoOp(incomming call from ${CALLERID} to [EMAIL PROTECTED])
exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten =
Dear all,
When I make a call using my PRI line, all goes well, but suddently the
call hangs up.
I searched the asterisk logs, and I found that.
Write to 55 failed: Unknown error 500
Short write: 0/15 (Unknown error 500)
What does this mean?
Why this occurs?
How could I solve that?
Someone
Hi,
does anybody know about the setvar option in asterisk's sip.conf. I am
trying to define it for a peer that's used when making calls using the
originate ami call, but it seems to not have any effect.
Marcus
--
Marcus Hunger - [EMAIL PROTECTED]
Telefon: +49 (0)211-63 55 55-61
Telefax: +49
Thanks for the reply Recardo..
I was indeed looking at something like this.
Also I was also looking at Asterisk's SRV lookups. Is there anyway I can
know that a SRV lookup has failed?
Regards,
Aadil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
i am getting Registration Refused error when i debug on console.
please tell me how can i registration every user without any username
and password and these user can make calls between each other.
i am very thankful if any body help me in this regards,
advcomm6*CLIiax2 debug
Tx-Frame Retry[-01]
On Wed, 2008-03-26 at 20:45 -0700, Vu AnhTuan wrote:
I'm having problem with voice quality on my trixbox using TDM2400B.The
trixbox is connected via 20 FXO ports on a TDM2400 with the hardware
echo cancel module. Echo cancel almost works, but the users hear what
they describe as a 'background
On Fri, 2008-03-28 at 02:12 +, John Meksavan wrote:
I got the sidecar to subscribed to an extension on the Asterisk
server, but the LED state on the SPA-932 never changes even when I am
a call with that extension on another VOIP phone- SPA-941. I got the
speed dial function to work, but
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like this:
[myphone]
secret=verysecretpassword
You can test manually any SRV DNS record using dig, like this:
dig -t SRV _sip._udp.fwd.pulver.com
At the asterisk CLI you can also verify that SRV lookup has been succeeded.
It shows something like this when it does:
parse_srv: SRV mapped to host fwd.pulver.com, port 5060
In your dialplan you
28 mar 2008 kl. 13.42 skrev Jared Smith:
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in asterisk's sip.conf.
Sure! This is one of my favorite features.
Let's say I have a definition for my phone in sip.conf, and it looks
something like
So, wouldn't it be great to enable setvar for outgoing calls too?
On Fri, Mar 28, 2008 at 1:55 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
28 mar 2008 kl. 13.42 skrev Jared Smith:
On Fri, 2008-03-28 at 12:30 +0100, Marcus Hunger wrote:
does anybody know about the setvar option in
On Fri, 2008-03-28 at 13:55 +0100, Johansson Olle E wrote:
Well, Jared, but that's the reverse. You stripped out this important
part:
am trying to define it for a peer that's used when making calls
using the originate ami call, but it seems to not have any effect.
The important thing
Without call-limit defined, when a sip extension calls
another sip extension then show hints shows that
both are InUse (as expected). When one of them hangs
up, both hints status become Idle (as expected).
With call-limit=1 for each SIP extension:
the caller is always Idle while the callee is
Hi list,
I am faced by a situation where I am trying to make a softphone and
a Siemens C450IP talk to each other. Both are hooked up directly to
the same asterisk, in the same IP net.
- a softphone runs on 192.168.14.3
- the C450IP is at 192.168.14.30
- asterisk runs on the machine known
--- Atis Lezdins [EMAIL PROTECTED] wrote:
On Thu, Mar 27, 2008 at 6:32 PM, Vieri
[EMAIL PROTECTED] wrote:
I have a queue I configured as strict and a cron
script I use to QueueAdd and QueueRemove agents
according to my company's requirements. Usually I
have
2 or 3 agents at a time
Hi,
I am on amd64 Linux and not really too happy with twinkle, linphone
and ekiga. Unfortunately, X-Lite and Zoiper, even though they
provide Linux versions (w00t!) have only x86 versions for download.
Do you guys know of amd64 versions of those, or can you recommend
other softphones that will
28 mar 2008 kl. 14.00 skrev Marcus Hunger:
So, wouldn't it be great to enable setvar for outgoing calls too?
Well, maybe in the outbound channel then. But that won't help much.
mixing the caller's and callee's variables in the INCOMING channel
would be messy and only cause issues.
But
Remember that if you enable call-limit=1 with a type=friend, you will
actually have one inbound call (on the user)
and one outbound call (on the peer).
Groupcount in the dialplan is propably a better solution to enforce
call limits than anything in the SIP channel.
It works with all channel
Particularly, I want to set the SIPADDHEADER variable dynamicly for peers
with rt-engine. Working around it might be possible, but having the thing
working transparently for Dial and Originate would be great.
On Fri, Mar 28, 2008 at 2:47 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
28 mar
--- Johansson Olle E [EMAIL PROTECTED] wrote:
Remember that if you enable call-limit=1 with a
type=friend, you will
actually have one inbound call (on the user)
and one outbound call (on the peer).
Groupcount in the dialplan is propably a better
solution to enforce
call limits than
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
28 mar 2008 kl. 14.56 skrev Marcus Hunger:
Particularly, I want to set the SIPADDHEADER variable dynamicly for
peers with rt-engine. Working around it might be possible, but
having the thing working transparently for Dial and Originate would
be great.
That should work today with the
John Meksavan wrote:
Asterisk Users,
I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B
wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for
our receptionist. After reading many forum postings on how to
configure the side car, I uprgraded the SPA-962
Vu AnhTuan wrote:
hi you,
I'm having problem with voice quality on my trixbox using TDM2400B.The
trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo
cancel module. Echo cancel almost works, but the users hear what they
describe as a 'background crackle/buzz'
Contact me at [EMAIL PROTECTED] and ask for a beta for the 64 bit build
of zoiper
Cheers,
Zoa
martin f krafft wrote:
Hi,
I am on amd64 Linux and not really too happy with twinkle, linphone
and ekiga. Unfortunately, X-Lite and Zoiper, even though they
provide Linux versions (w00t!) have
Create a User and a Peer on both the machines for each other.
e.g IAX.conf on PCa
[pca2pcb]
type=peer
host=[IP OF pcb]
username=pca2pcb
serect=pca2pcb12345
qualify=yes
[pcb2pca]
type=user
context=default
auth=md5
secret=pcb2pca12345
deny=0.0.0.0/0.0.0.0
permit=[IP of pcb]
qualify=yes
ON PCb
Hi All
I am developing a client that uses libjingle to do xmpp stuff with
ejabberd. I can also make audio calls between those clients. What I am
trying to archive now is to send calls to pstn using jingle. I was
told in the jingle-dev community that asterisk can do that.
Is there any way to send
Mojo with Horan Company, LLC wrote:
Sean Dennis wrote:
bilal ghayyad wrote:
Hi All;
I have been chocked just when I saw some posts talking
about how much the IAXy is bad :) -
So I would like to ask, did any one try it later and
wether it is good or not? I am asking this because I
Hanna Wallin wrote:
Hello List!
We're having trouble making call deflection on ISDN PRI. We would like to
transfer a call to an external extension but keeping the callerid of the
caller so it can be presented to the receiver of the transferred call.
At the time we're using Zaptel
On the Asterisk CLI show hints
Registered Asterisk Dial Plan Hints =-
[EMAIL PROTECTED]: SIP/211
State:IdleWatchers 1
- 1 hints registered
On the Asterisk CLI sip show subcriptions
Peer UserCall
On Fri, Mar 28, 2008 at 12:05 AM, Paul Hales [EMAIL PROTECTED] wrote:
Can't you just use the same bootrom for all your polycom phones?
PaulH
On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote:
I have a question about DHCP and boot server supporting more than 1
model of
I may be missing something here... but won't a 32bit binary run just fine on a
64bit platform? Would you even see a performance increase or advantage to a
64bit soft phone versus a 32bit version?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: zoa [EMAIL
On Fri, 2008-03-28 at 15:37 +, John Meksavan wrote:
So on my SPA-962 + SPA-932, the LED state remains GREEN, because
Asterisk thinks it is in Idle state, which extension 211 is clearly
not.
Why is that?
Do you have the call-limit setting in sip.conf for SIP/211? At the
Asterisk
Ok,
Now I have a friday afternoon patch for the community.
In the branch
http://svn.digium.com/view/asterisk/team/oej/peer-chanvars/
there's an addition to the SIPPEER() dialplan function where you can
retrieve a setvar= channel variable defined in sip.conf for the peer.
The branch is based
also sprach Tim Nelson [EMAIL PROTECTED] [2008.03.28.1637 +0100]:
I may be missing something here... but won't a 32bit binary run
just fine on a 64bit platform? Would you even see a performance
increase or advantage to a 64bit soft phone versus a 32bit
version?
Not if all the libraries have
I am trying to get BLF working on Grandstream phones with 1.2.27. I
actually have it working, but I found a very strange issue and I am
wondering if anybody knows what the problem is.
Here is the scenario. If I have 3 phones, A, B and C. A monitors
presence of B and C. Right now, if I call
Hi,
Iam getting calls from an POTS system on an NT port. Multiport BRI card
running bristuff 0.3.
From time to time the recognized number is incomplete and dial failed.
Is there any way to increase timeout waiting for called numbers?
Because dialed numbers can be from 3 to 13 digits there is no
Dear Asterisk-User friends,
After realtime queues are defined, how does it work with the agents? There
seems to be no db table for agents.
If I can't define agents for the realtime queues in the db, how can agent
login/logoff be done?
Thanks alot for your help.
Thanks,
Mark
Thanks for you guys help. The status LED on the sidecar takes an awfully look
time to change from GREEN to RED and vice versa. Some times, it would reguire
up to 15-20 minutes at beginning or ending the call on the extension.
What would cause the delay? Is it my network?
Best Regards,
mark morreny wrote:
Dear Asterisk-User friends,
After realtime queues are defined, how does it work with the agents?
There seems to be no db table for agents.
If I can't define agents for the realtime queues in the db, how can
agent login/logoff be done?
Thanks alot for your help.
Hi,
I have a wav file recording that i want to use on my voicemail, how
can i set this up?
thanks!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi,
Sorry to resend the same question. This mail is just to add a few bits of
details:
When I tried to join the support queue, I get
L RealTime: Retrieve SQL: SELECT * FROM queue_member_table WHERE interface
LIKE '%' AND queue_name = 'Support' ORDER BY interface
[Mar 29 10:01:52] WARNING[6203]:
Dear Mark,
I did also populate members to the queue_member_table. The output of show
queue also tells me that Asterisk read the member info too. When I tried
to access the queue, it saidUnable to join queue 'Support' What do you
think may have gone wrong? Also, how would I be able to add a
Dear Mark,
Here is my queue_member table, is this how it should look?
mysql SELECT * FROM queue_member_table WHERE interface LIKE '%' AND
queue_name = 'Support' ORDER BY interface
- ;
+--+++---+-++
| uniqueid | membername | queue_name |
Actually, its just the opposite... The call is okay for a few seconds,
then the odd echo kicks in. When the training isn't turned on, it takes
20 seconds to so to kick the echo. With the training on, it works great
except for this bug. Several of the people using the same * system but
different
Paul Hales wrote:
Can't you just use the same bootrom for all your polycom phones?
To elaborate in case it isn't obvious from above: Even if you needed
different config files or even SIP applications by phone, you don't have
to go to separate DHCP entries by phone. The MACADDESS.cfg file points
Matthew Gibson wrote:
What are you trying to do? I run a 7970 here with SIP.
Get it to work ;)
I can get the phone to register but something via way of NAT (I'm not
using it) is getting in the way. I was hoping to find an example
SEPxxx.xml file from someone using the 7971.
No, you can keep dialing and make your call if you wish, or you can
answer the call.
--
Scott Plante, CTO
Insight Systems, Inc.
(+1) 404 873 0058 x104
[EMAIL PROTECTED]
http://zyross.com
Brent Torrenga wrote:
List,
Question about the Polycom 650: when dialing the digits for a phone
Dear Sanjay,
Sorry sanjay i miss to explain completely. My PC2PC mean is
Dialer2Dialer i want to allow call between Dialer with out any
registry and authentication through IAX.
so i need to setup Asterisk accept calls from any user and users can
call to each other without any password and
There is a sip.cfg entry divert.dnd.x.contact that is supposed to be
where the call goes if DND is enabled. You could presumably set that to
* plus the extention to go to the extension's voicemail, or to some
other dialplan to play whatever you want, though I haven't tried it.
Lee, John
What is your extensions.conf setup? that has alot to do with it (I
strongly suggest you use macros.) What SIP NNN code does the phone
return when DND?
On Mon, Mar 17, 2008 at 2:00 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
I am using Polycom IP600 phone. If I call a phone which has DND (do
*CLI show application Transfer
-= Info about application 'Transfer' =-
[Synopsis]
Transfer caller to remote extension
[Description]
Transfer([Tech/]dest[|options]): Requests the remote caller be transferred
to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
an incoming
If the problem is specific to certian inspections I would verify the LAN
segments involved in connecting those devices.
Rob Schall wrote:
Actually, its just the opposite... The call is okay for a few seconds,
then the odd echo kicks in. When the training isn't turned on, it
takes 20 seconds
With canreinvite=no you are forcing asterisk to remain in the call
path. As long as Asterisk is in the call path, it is supposed to be
transcoding the calls, so it doesn't care what the compatible codecs are
between then endpoints. Each leg of the call is phone-asterisk so
asterisk
They are all connected directly to the same switch which asterisk also
connects into. Its a small office (6 people).
Rob
Anthony Francis wrote:
If the problem is specific to certian inspections I would verify the LAN
segments involved in connecting those devices.
Rob Schall wrote:
Hello list,
after spending the best part of an afternoon trying to build Asterisk on
an old EPIA VIA C3, I thought that writing a tutorial would make life
easier for future compilers:
http://astrecipes.net/index.php?n=356
I had never compiled Asterisk for a different architecture, and I'm
Hi Kevin, I need I little bit of help again.
I have installed in my PC for testing the last available version of asterisk
for testings. And I am using easyeclipse with cdt plugin to create a C project
and compile the app_skel.c source file from the asterisk-1.4.18.1. (GCC 4.1.3)
I noticed
I have a wav file recording that i want to use on my voicemail, how
can i set this up?
You could play that file before sending the person to your voicemail
and pass the s option to it
Type show application voicemail on asterisk CLI to see the options.
hth
The Asterisk.org development team has released Asterisk versions 1.4.19-rc4 and
1.6.0-beta7.
These releases contain significant bug fixes over the previous pre-releases of
1.4.19 and 1.6.0. We would like to thank everyone for all of the help with
pre-release testing. Unless anything new comes up,
Mark Quitoriano wrote:
Hi,
I have a wav file recording that i want to use on my voicemail, how
can i set this up?
thanks!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
martin f krafft wrote:
What's going on here? From all I can tell, the clients do the right
thing, each selecting the first codec offered by asterisk (which
they support), but asterisk is going a bit lala here, isn't it
I think Brent's on to it there -- as he suggested, get your allow= and
Have you set a call limit for each SIP peer? This is now required as of
version 1.4. It took me a while to figure out all the issues when
migrating to 1.4.
John Meksavan wrote:
Thanks for you guys help. The status LED on the sidecar takes an
awfully look time to change from GREEN to RED
According to http://kb.digium.com/entry/12/
The Iaxy will respond to pings on port . You can ping your
broadcast IP on your network and listen with tcpdump on your
network on port which will show the Iaxy responding and what
IP address it is coming from.
On Sat, Mar 29, 2008 at 7:26 AM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
You could save it to your asterisk voicemail directory, which is often
something like:
/var/spool/asterisk/voicemail/your_context/your_voicemailbox_number
The files used are unavail.*, busy.*, and
70 matches
Mail list logo
