Hi
An Asterisk queue uses field names / config variables such as:
announce-holdtime
However, documentation regarding realtime is very unclear.
voip-info.org suggests to use announce_holdtime.
Is this correct?
What about monitor-type? Should it be underscored too (monitor_type)?
Thanks,
--- On Fri, 10/5/12, Vieri rentor...@yahoo.com wrote:
An Asterisk queue uses field names / config variables such
as:
announce-holdtime
However, documentation regarding realtime is very unclear.
voip-info.org suggests to use announce_holdtime.
Is this correct?
What about
On Fri, 2012-10-05 at 05:21 -0700, Vieri wrote:
--- On Fri, 10/5/12, Vieri rentor...@yahoo.com wrote:
An Asterisk queue uses field names / config variables such
as:
announce-holdtime
However, documentation regarding realtime is very unclear.
voip-info.org suggests to use
Hi
Charles Solar a écrit :
Hi guys, I am new here but I have a quick question.
I have an incoming trunk that sends me calls from various usernames I have
with them. Only trouble is they send invites as s...@my.ip.addr, not as the
username I have with them. So I cannot match extensions like
Hi!
I have an incoming trunk that sends me calls from various usernames I
have with them. Only trouble is they send invites as s...@my.ip.addr, not
as the username I have with them
You need to adjust your register = statement with them: Add
/username
to the end of it, then calls won't
Ah that is brilliant, thanks a lot.
Charles
On Mon, Jun 1, 2009 at 9:35 AM, Administrator TOOTAI ad...@tootai.netwrote:
Hi
Charles Solar a écrit :
Hi guys, I am new here but I have a quick question.
I have an incoming trunk that sends me calls from various usernames I
have
with
Again, another brilliant solution that I was unaware of :D
Thanks so much
On Mon, Jun 1, 2009 at 10:24 AM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
I have an incoming trunk that sends me calls from various usernames I
have with them. Only trouble is they
Do be aware that routing on the To URI specifically breaks RFC rules.
Routing should be done based only on Request URI, the user part of which
Asterisk sees as an extension.
Tell your provider to pass you the DNIS in the RURI, or get another
provider. Most should be able to override your
Hi guys, I am new here but I have a quick question.
I have an incoming trunk that sends me calls from various usernames I have
with them. Only trouble is they send invites as s...@my.ip.addr, not as the
username I have with them. So I cannot match extensions like I would want
to.
Here is a
Hi,
I am using my asterisk server like a gateway and one provider ask me to pass
an extra field with the IP of the peer that is using the connection,
probably to have more control on the authentication. I was wondering how I
can implement this.
Thank you
Hi,
Could you please explain what your provider is expecting?
You should only have to provide your public IP address.
On 4/4/07, Il Neofita [EMAIL PROTECTED] wrote:
Hi,
I am using my asterisk server like a gateway and one provider ask me to
pass an extra field with the IP of the peer that is
How is the UserUserInfo field in the Q931 messagetranslated to SIP message.
Does Asterisk have RFC 3398 support?
Thanks
--Ray
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Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
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Hello,
Here is my config :
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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
Hi there
I don't know if this utility is available anywhere at the moment but I
thought id ask you guys if you know of one
What I would like is a way of adding a field to my cdr records (either the
Master.csv or a destination mysql table) for cost ! based on some sort of
config file (or table)
Hi I am trying to configure a SIP registration to Primus. They said I
have to remove the opaque= field to get this to work. Does anybody
know how to do this?
Thanks,
-ls
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Title: Queue_log field definitions
Can anyone tell me the field definitions for the queue_log file in the Asterisk log directory?
1080593958|1080593892.0|salesq|NONE|ABANDON|1|1|50
MIS wrote:
Can anyone tell me the field definitions for the queue_log file in the
Asterisk log directory?
1080593958|1080593892.0|salesq|NONE|ABANDON|1|1|50
fprintf(qlog, %ld|%s|%s|%s|%s|, (long)time(NULL), callid,
queuename, agent, event);
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Hi I wonder if anyone can throw some light on the * console message. This
only occurs when I register a phone on the end of a BT ADSL line, with a Draytec router/modem.
The phone registers okay but cannot dial out.
Console message:
Notice[1125329600]: File chan_sip.c, Line 1759
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