Hi,
Polycom phones configured on asterisk pbx and are using contact directory on
phones. To modify entries xml file for each phone needs to be modified and
have to reboot all phones to accept updated file.
Is there any way via asterisk, that we can use central database and on
modification
I use the mini-web browser built into the phone and have a custom
button (directory) that accesses the directory, which is hosted on a
web server.
It isn't perfect, but it's better than the XML files IMHO. That said,
there's an enterprise license for these phones which enables directory
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk
I'm migrating an application running on a fairly old 1.4 (or 1.2?)
version of Asterisk to some boxes running 1.6.0.27
The application takes an inbound INVITE like:
mumble-fratz-sip%3afoo%40bar@asteriskbox.abc.com:5062
The older version of asterisk replies with a 200 OK and a Contact:
header
Hi,
I'm using an Asterisk box with zap channel as a gateway between PSTN and
an alarm receiver system. The alarm system uses Contact ID protocol.
My problem is that the negotiation fails and I think that the problem is
that kissoff tone is cut and the transmitter doesn't recognize it.
Maybe the
On Thu, Feb 5, 2009 at 7:22 AM, Geoff Lane ge...@gjctech.co.uk wrote:
The nice thing about that is that if I use MySQL I can run the
management application on another machine, and so don't need to run a
web server on the Asterisk box. However, I wonder whether the overhead
necessary to run
On Wednesday, February 4, 2009, D Tucny wrote:
I use a slight variant of this...
exten =
s,n,Set(CALLERID(name)=${IF(${ISNULL(${DB(cidname/${CALLERID(num)})})}?Unknown:${DB(cidname/${CALLERID(num)})})})
exten = s,n,NoOp(Caller ID name mapped to ${CALLERID(name)})
Basically the same as
For a simple (but flexible) case I would consider ODBC + func_odbc.
Here is the idea (in case you aren't aware of how it goes...)
- Make a DB available (your choice as long as it is accessible via ODBC)
- Create table in it with your contacts (say columns number and
name, maybe more)
-
On Wednesday, February 4, 2009, Ex Vito wrote:
For a simple (but flexible) case I would consider ODBC +
func_odbc. Here is the idea (in case you aren't aware of how it
goes...)
[... snip ...]
It may be a bit more work than using the Ast DB or other means, but it
has the advantage
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or anything else maintained externally, just a
shows as IM.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane
Sent: Tuesday, February 03, 2009 10:05 AM
To: Asterisk Users
Subject: [asterisk-users] Contact lookup
Hi All,
Asterisk 1.4.12 on CentOS 5
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or
: February 3, 2009 11:51 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Contact lookup
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass
2009/2/4 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the
Hi,
I am trying to SUBSCRIBE for message waiting indications to asterisk,
it sends 200 OK but contact header is missing(it is mandatory since
subscribe is dialog establishing method), due to which parsing fails, any
body knows about this issue...?
Regards,
Subramanya
Hi,
Hi,
I am trying to SUBSCRIBE for message waiting indications to asterisk,
it sends 200 OK but contact header is missing(it is mandatory since
subscribe is dialog establishing method), due to which parsing fails and
also expires is 0 in the 200 OK any body knows about these issue...?
Greetings,
I have a problem getting Asterisk registered as a UAC against the
MetaSwitch call agent, because the customer insists on running it on a
NAT'd box. Thus, the Contact: field in the REGISTER request betrays
the private IP address of the Asterisk box, but the source IP of the
message
Got this figured out. externip= does work if the other NAT-related
options are also enabled, plus it appears that Trixbox (which is what
the end-user was using, it seems) handles this well in its config file
structure regardless.
--
Alex Balashov
Evariste Systems
Web:
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage---
Hello,
Here is my config :
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage---
Remarque : message transféré en pièce jointe.
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur
Hello,
Here is my config :
Asterisk as registrar server :public ip:5050
Ser as outbound proxy server :public ip 5060
I wish ser to handle the packets between Nat box
(netfilter) and Asterisk However contact field in
the sip HF don't change from nat box to asterisk which
don't allow to keep
Hello,
Here is my config :
Asterisk as registrar server :public ip:5050
Ser as outbound proxy server :public ip 5060
I wish ser to handle the packets between Nat box
(netfilter) and Asterisk However contact field in
the sip HF don't change from nat box to asterisk which
don't allow to keep
On Thu, 2005-09-01 at 15:59 -0400, Jeremy Melanson wrote:
Hi Jesse.
A couple questions..
What firmware version are you using?
Bootrom 2.6.2.20032
Sip 1.5.2.0054
How does your phone get it's config (FTP, TFTP, Manual config)?
Initially it got the config from TFTP w/ the new boot rom.
I'm testing out some IP501 phones and I ran into an issue. WHen I try
to add a new contact into the directory, I am not able to. A window
blinks really fast but the entry isn't saved. When you exit the Contact
Directory system you get a 'Busy! Please try again' window.
What the heck could be
Hi Jesse.
A couple questions..
What firmware version are you using?
How does your phone get it's config (FTP, TFTP, Manual config)?
-
Jeremy
On Thu, 2005-09-01 at 12:51 -0700, Jesse Keating wrote:
I'm testing out some IP501 phones and I ran into an issue. WHen I try
to add a new contact
On Thu, 2005-09-01 at 13:04 -0700, Jesse Keating wrote:
Bootrom 2.6.2.20032
Sip 1.5.2.0054
I rolled back to Sip 1.4.1.0040 and I can save entries, but the menu
system is all different and not easy to navigate. This is not so good.
--
Jesse Keating
GameHouse -- Systems Engineer
Hello Khurram,
This is adnan from EBS kindly contact me as soon as possible i'll
contact you on your number but its almost busy every time.
Other *'s users kindly forgive me because i have no option right now.
___
Asterisk-Users mailing list
[EMAIL
Sorry to do this to the list, but I have no choice .
Walker,
I've been trying to send you an email off-list for the last couple of weeks,
but one of my mail-hops is failing, do you have alternative address that I
can try ???
Paul
___
Asterisk-Users
Hi,
I have noticed that when I am calling from my Snom-phone to another
Snom-phone through Asterisk, the SIP-message's Contact -header could be
sometimes empty and for example other Snom get no BYE-message.
Here is example of that kind of message:
10 headers, 0 lines
Sending to 192.168.0.32 :
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