Hi,
I run Asterisk server, ver 3.1 (a bit old version).
I know that the file “/etc/asterisk/sip.conf” contains phone numbers/extensions.
I have a doubts and a question.
the field “callerid=” is where I can type the name of the user. This name will appear on the display of the VoIP phone. Do I have to use quotes or not when I have spacee? Which one of this entry is correct?
A. callerid=“John Doe the anchorman”
B. callerid=John Doe the anchorman
when I apply changes to any .conf file or to a specifi file like sip.conf do I have to reload or reboot asterisk?
Which command is the best one to use?
core restart gracefully
asterisk -r -x “reload”
Is there any other file where I can find the name of the user who corresponds the extension, other than the file sip.conf?
Hi,
I run Asterisk server, ver 3.1 (a bit old version).
Please identify how you believe you are running that version.
There has never yet been a version 3.1 of Asterisk.
I know that the file “/etc/asterisk/sip.conf” contains phone
numbers/extensions.
I have a doubts and a question.
the field “callerid=” is where I can type the name of the user. This
name will appear on the display of the VoIP phone. Do I have to use quotes
or not when I have spacee? Which one of this entry is correct?
A. callerid=“John Doe the anchorman”
B. callerid=John Doe the anchorman
The examples given in the default sip.conf file are:
callerid=John Doe <1234>
callerid=“Jane Smith” <5678>
Note that this is not only the Caller ID name, but also the number.
when I apply changes to any .conf file or to a specifi file like
sip.conf do I have to reload or reboot asterisk? Which command is the best
one to use?
core restart gracefully
asterisk -r -x “reload”
This depends on what you are changing.
Some things like sip.conf can be reloaded using “asterisk -Rx ‘sip reload’”
Almost everything other than asterisk.conf can, as far as I know, be achieved
with “asterisk -Rx ‘reload’”
If you really do want to restart the entire system, then “core restart
gracefully” is a good choice.
Is there any other file where I can find the name of the user who
corresponds the extension, other than the file sip.conf?
Unlikely, but possible.
It sounds as though you are not using any database (“realtime”) features of
Asterisk, so everything is going to be configured using flat (“text”) files.
However, the dialplan itself (extensions.conf and anything it includes) can
adjust the Caller ID name and/or number for any call (inbound or outbound) as
part of processing a call, so if you are experiencing strange behaviour, you
might need to look there.
Maybe it would be good to tell us specifically why you are asking this
question, so we might be able to guide you better.
Are you perhaps inheriting an Asterisk system from someone else, and you’re
not entirely sure how it has been set up?
If you are still using chan_sip, you should use callerid_name, as callerid is relying on various heuristics to determine what is name and what is number. It looks like the current SIP handler relies on those heuristics, although, as I haven’t followed the code through completely, it could also go wrong if there is no explicit number.
I would suggest using "s, if you don’t provide a number as well, as the parsing rules will treat something that looks like it is just a number as being the number, not the name, unless it is quoted.
The full format approximates the rules for email addresses (although most email programs don’t allow those to work to best effect). Email addresses are designed so that, as long as you don’t include characters that cause confusion, they look like the name as it would appear in a paper memo heading (no quotes) with the technical reference, the email address, in angle brackets).
In my experience, it is not normal to give just a name, even though it looks like the, current, code supports this.
Just to add to what has already been said, restart is normally too drastic and core reload is generally also too drastic. However, there are some limited cases where it is necessary to restart, because the the change involves something fundamental. I don’t think chan_sip has any such cases. chan_dahdi and app_queue do have some parameters that cannot be changed without a restart. I’m not sure about chan_pjsip. Generally you will get a warning message if you specify something that cannot be changed without a restart.
What goes in sip.conf are not extensions (except to the FreePBX people, who conflate peers with extension and generally use the same name for both. (An Asterisk server is also the hardware, not the software; asterisk acts as both a SIP client and a SIP server.)
Other channel drivers have their equivalent of callerid, but only for their own end points. There is a users.conf file, that can be used as an alternative to channel driver files, but I think if you use it, you will tend to confuse people trying to support you.
The dialplan (etc.) can change CALLERID(name) on the fly.
If you don’t explicitly set it, any name used by the phone will be used.
I think that could be some commercial product from Digium. This forum can only really support the underlying open source Asterisk, and, as noted, there never has been a version 3.
First of all thank you for your reply.
My apologies for an incorrect info. I just checked, via the CLI, current Asterisk version.
I have “Asterisk 11.8.1”
As for the question "Maybe it would be good to tell us specifically why you are asking this
question, so we might be able to guide you better" this is my situation:
in sip.conf I have declared callerid=John Doe without quotations but when I the user calls me on the voip phone display it appears another name. So, I am questioning whether there is another place, other than sip.com where I can find this Caller ID name.
I already checked in extensions.conf but did not find anything.
As for my “apply changes to any .conf file” question I was wondering if is there a way not to restart the entire Asterisk but only tell Asterisk to update info changed in sip.conf.
Last question: when I make a change in sip.conf (like I add a new phone device, or I modify Caller ID name do I always have to core restart gracefully? Is there a different way where Asterisk does not go down? I do not want all other users not to be able to continue using their phones while I only must apply a change/update.
It might be coming from the calling phone, if the parameter is being ignored, possibly because of the null caller number.
The funny version makes me think you have an Asterisk appliance, which might well do a database lookup in the dialplan, possibly hidden in an AGI script.
If you are using Asterisk Realtime Architecture, some of sip.conf may actually be coming from a database.
sip reload
module reload chan_sip
There should also be a command that allows you to tell it to reload specific files.
That’s what I meant, as callerid=12345 would not be interpreted as being the same as callerid-name=12345, so it is best to always make the distinction.
However, you are making yourself impossible to support by using an ancient version, and you are asking questions about configuration loading that you should be able to answer for yourself, from the documentation, so I’m probably not going to continue on this thread.
Hi david551,
I am sorry for your disappointment. It kind of offended me but no problem, I can understand your frustrations. Too bad I was only looking for help considering that I do not have knowledge and experience with Asterisk and that I still have unresolved issues to deal with being left alone. I was hoping to find some answers and solutions but perhaps I did not post my questions the way you would expect, or I should have done. My apologies for this.
I will try to open new threads with direct and simple/short questions. Perhaps, this will make it easier on people like you who are devoting you free time in assisting users, like me, who need help.
