Displaying 20 results from an estimated 11000 matches similar to: "Sending SIP NOTIFY / How to get remote SIP port?"
2015 Jun 12
1
Voice mail and caller ID
On Fri, 12 Jun 2015 11:49:05 -0700
John Kiniston <johnkiniston at gmail.com> wrote:
> Try this for CHAN_SIP:
>
> same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
> same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the
> mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we
> have a mailbox defined log into it
Perfect.
2016 Jul 31
3
Removing mailbox and password prompt for voicemail
I tried your extension definition as suggested:
exten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail)
same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s)
same => n,Hangup
But there was no change in the prompts asked, ie. the voice first asked for
'mailbox', and then 'password' as before. The prompts are not removed.
Please clarify what you mean by the
2015 Jun 12
2
Voice mail and caller ID
I have this in my sip.conf:
exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}")
same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s)
same => n,Hangup
However, my extensions are set up so that they always show the external
number, not the extension:
[foobar2](client-phone)
secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx
callerid=Candace <5555551212>
2009 Aug 20
8
mysql sip realtime
Hi
I have some question about mysql realtime.
1) Anyone know exactly if there is a specific order to declare sip table
column for realtime ? In which file can I find that order ?
2) In my extconfig.conf, [settings] are :
sipusers => mysql,general,siptable
sippeers => mysql,general,siptable
so means that I use realtime dynamic exactly ?
Is it normal if some parameters from sip.conf still
2014 Nov 22
2
High resident memory with 11.14.0 ?
>
> Its up to 5.8G of resident memory with 28321 calls processed.
> The OOM killer is going to kill this soon at this rate (8GB RAM machine).
> This seems like a pretty serious problem.
> It looks like I'll need to restart asterisk every night....
Hi the number of cpu cores that you see with top times 512Mbyte is the
level of ram that's needed
e.g. a hp-gen8 with 2 octo
2003 Nov 12
7
SoftFax question
Hi,
I am looking at using the softfax that Steve Underwood has developed.
It's very straight forward when you assign an extension for the fax.
A function that several pbx's has is that they listen for the 'faxtone'
for 5 seconds
after 'answer' in the menu where you can enter your local extension number,
it's normally done in parallel with the DTMF detection. I think
2008 Nov 04
1
Is SIPPEER curcalls working for you ?
Hi,
In this thread
http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html ,
I wondered whether SIPPEER curcalls was working.
I could test this anew today. Here are my findings :
Alice, Bob and Carol ar all using SIP Phones.
Whenever Alice is calling Bob,
- if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0
- if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1
2023 Jul 22
1
Parallel dialoog with different Alert-Info headers
We have a couple of parallel ring settings (and this has worked well for eons).
Either in the form of
same => n,Dial(SIP/1001 & SIP/1002 & SIP/1003 …..)
Or via a subroutine (below) that has a bit of extra logic:
FOO = 1010 & 1019 & 1017 & 1033
...
same => n,gosub(sub-callout,s,1,(${FOO},”Ringing all class FOO telefons"))
Now I have two types of phones
2016 Aug 05
2
Toll free pattern matching
I have this in my config:
exten => _800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})
same => n,Dial(SIP/tollfree/1${EXTEN})
exten => _1800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})
same => n,Dial(SIP/tollfree/${EXTEN})
exten => _NXXNXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})
same => n,Dial(SIP/trunk/1${EXTEN})
exten =>
2010 Feb 16
1
CODECS: Best practice question: Avoid transcode when calling out?
What is the current best practice to avoid transcoding on an outgoing call
to a
party whose codec preference is not known in advance?
In other words, incoming calls are easy since codecs are negotiated from
least-known (the remote party) to most-known (my endpoint) and my codecs can
simply be preferred accordingly to match the remote.
Outbound calls seem harder. Our endpoints always negotiate
2007 Jan 19
1
Red: Sip Phone CID
Here is what I have in my extensions.conf file now. Trustrcid and
sendrcid are turned to "yes" in the conf file.
I'm not fully sure how the SIPCalledRPID works though. The example I
found seems to try and provide the stuff automatically (id and name),
but does the SIPPEER stuff even exist?
I think this is probably the right track though. Any insight would be
much appreciated.
2010 Jan 20
1
Using SIPPEER status with CUT function? SOLVED
On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote:
> Hi All,
>
> I'm using Asterisk 1.4 branch and checking the status of some SIP
> Peers with the functions ${SIPPEER(101:status)} and the result is "OK
> (48 ms)". ?Seems to work fine.
>
> Now I would like to use the function CUT to set a variable with the
> 'OK'
2011 Jul 11
1
${HASH(SIP_CAUSE, ...)} and peer name
Hello,
I'm trying to figure out what was the return code of SIP for a call.
The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to
retrieve the peer name using ${CHANNEL(peername)}, I have an error message
that CHANNEL does not have peername or it is not available to be used.
I tried to print it with NOOP on a live channel, and also after hangup, both
with the same error
2012 Feb 11
1
Asterisk perl AGI confusing variables
Hello all,
I'm struck with a very strange problem today. I've an AGI with some code
subroutine snippet as follows:
sub enable_sbc($) {
my $carrier = shift;
my $tmp = substr($carrier,1);
my $jkh = $tmp;
$server_port = $ast_agi->get_variable("SIPPEER($jkh,port)");
$ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");
2003 Nov 13
1
RE: Aculab SS7/ISUP (new subject)
>Freddi Hansen wrote:
>> with boards from Aculab, we are replacing Aculab boards with Digium
>> boards BUT we would need more
>> Digium boards IF we could use both Digium and Aculab cards in the same
>> server. The reason being that
>> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards
>> in the servers that must support
>>
2006 Dec 15
2
Trying to forward calls by using the Callee's context as the forward dial context
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally be allowed to dial within their own context. (I
don't want a local call only person forwarding to a long dist number,
for example.) I'm able to
2006 Feb 14
5
Multiple AGI Issues
I've got several issues with AGI/FastAGI
1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk block and not return a result until the command is complete? Specifically, the dial command. If I send a Dial command to Asterisk, I don't get a return result until AFTER the call is HUNG UP. Not when it's ringing, not when the call is connected, but when it's
2006 Apr 03
2
SIP Responsecodes
Hi,
It seems as 'the google' has left me today so I am trying the list.
How do I get access to SIP responsecodes from dialplan/agi. Yes I know
that I should stay with 'DIALSTATUS' but there are cases where I need
the responsecode like '484 adress incomplete' and not just the 'NO
ANSWER' DIALSTATUS.
Is there a channel variable/function that skipped over by
2007 Feb 11
0
realtime and save ip server in database
Hello
I change this from chan_sip.conf (see ipsvr):
static void realtime_update_peer(const char *peername, struct
sockaddr_in *sin, const char *username, const char *fullcontact, int
expirey)
{
char port[10];
char ipaddr[20];
char regseconds[20];
time_t nowtime;
-> char ipsvr[20];
time(&nowtime);
nowtime += expirey;
2015 Jun 12
0
Voice mail and caller ID
Try this for CHAN_SIP:
same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox
same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a
mailbox defined log into it
If you are using PJSIP it's more complex
same => n,Set(EndPoint=${CHANNEL(endpoint)}) ; Get the peer
same =>