Displaying 5 results from an estimated 5 matches for "siphost".
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2008 Jan 31
1
Dropped calls
...31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for
channel '0x82042e8'
[Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for
channel '0x82042e8'
[Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID:
D1B9-141D-46684820168D9512F870-009 at SipHost Their Tag c136d668-768786 Our
tag: as0bc591fc
[Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID:
D1B9-141D-46684820F9EEEBF1F8F2-008 at SipHost Their Tag 2b4f6f33-768786 Our
tag: as496fd97d
[Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID:
D1B9-141D-4668482079ECFA6...
2020 Oct 25
2
chan_sip doesn't authenticate on INVITE from a Dial() command
...is an optional [!dnid] argument you can append to alter the
* To: header.
(Note: I don't think I have ever seen that optional "!dnid" argument
documented anywhere...?)
So, the version with the username and password looks to me like what I want...
Dial(SIP/${SIPuser}:${SIPpass}@${SIPhost}) or else
Dial(SIP/${SIPuser}:${SIPpass}@${SIPhost}!${SIPdial})
would seem to be what I need (I need to authenticate to SIPhost with the
credentials SIPuser and SIPpass and I want to dial on to SIPdial).
However, doing this results in the NOTICE message:
chan_sip.c:23862 handle_response_invite...
2020 Oct 25
0
chan_sip doesn't authenticate on INVITE from a Dial() command
...end to alter the
> * To: header.
>
> (Note: I don't think I have ever seen that optional "!dnid" argument
> documented anywhere...?)
>
> So, the version with the username and password looks to me like what I
> want...
>
> Dial(SIP/${SIPuser}:${SIPpass}@${SIPhost}) or else
> Dial(SIP/${SIPuser}:${SIPpass}@${SIPhost}!${SIPdial})
>
> would seem to be what I need (I need to authenticate to SIPhost with the
> credentials SIPuser and SIPpass and I want to dial on to SIPdial).
It turns out that the username needs to be included twice (!?), as in:...
2004 Jan 30
2
Can Asterisk act like a normal sip phone?
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello everyone,
I'm relatively new to the subject - so pleace don't punish me for
idiotic questions. ;-)
Can Asterisk act like a normal Sip phone and e.g. connect to another
sip-gateway? Background: There is a new german company at:
http://www.sipgate.de (sorry German only page)
They offer a a gateway between a real telephone number and
2006 May 23
3
AGI ?
...mple of what I have in my script.
#!/usr/bin/perl
use strict;
use Asterisk::AGI;
my $callerId = "MyNumber";
my $log = "/var/log/asterisk/dialout.log";
my $gIaxUser = "MyIAXPROVIDER";
my $gIaxPass = "MYIAXPASS";
my $gIaxHost = "myIAXHost";
my $gSipHost = "MYSIPHost";
########### SYSTEM VARIABLES #####################
my $AGI = new Asterisk::AGI;
my $gDialedNo = shift;
my $gNumber = "";
my $gPrefix = "";
my $gAreaCode = "";
### Sets Outbound CallerID ###
$AGI->set_callerid($callerId);
open(LOG, "&...