Displaying 20 results from an estimated 1000 matches similar to: "sip_header=value?"
2007 May 17
1
Multiple lines on Linksys/Sipura phones
I'm going to be deploying around 30 IP phones with Asterisk in the near
future. I've tentatively settled on the Linksys/Sipura SPA9xx family.
I am unclear on the notion of "lines" in the context of SIP phones like
these. The SPA942 model has a 2-line-to-4-line upgrade available, but I
don't know why I'd need to purchase it.
I have tested a SPA942 with Asterisk, and
2007 Apr 07
2
Different devices for asterisk!!!
Hi all,
Im trying dial a user according to the device s/he uses. i mean if the user
is using asterisk as a peer, then i have to pass the extension in the dial
application like this:
Dial(SIP/${EXTEN}@user) ;so that s/he can perform routing according to the
DNID.
and if the user is using sipura, linksys or grandstream i dial the user like
this,
Dial(SIP/user)
so is there a way to know what kind
2007 Oct 25
3
Getting SIP Response Code from HANGUPCAUSE
I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead?
Doug.
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
-------------- next part --------------
An HTML
2010 Nov 23
2
Function SIP_Header not registered
Hello,
I'm trying to use SIP_HEADER function on my dialplan but I receive this
message (on the console):
pbx.c:3367 ast_func_read: Function SIP_Header not registered
Why?
Thank's
- Bakko
2007 Dec 01
1
REFER mesage extraction using SIP_HEADER
Hi * users,
I would like to extract the information present in the SIP REFER
message that comes to asterisk. Would SIP_HEADER() allow me to do that
? I have used SIP_HEADER() for extracting the to and from SIP headers
previously.
Thanks
Regards
--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University
Tel: 1-646-387-5998
2008 Mar 13
3
How to find out the IP of the calling party?
Hi All,
I'm trying to achieve the following:
- If <sip/iax user> logs in from home, they can dial internal extensions
only (this is to avoid employees going wild on local/mobile calls from home)
- If <sip/iax user> logs in from the office, they can call anyone they want.
Since I have my users defined in an LDAP tree, I'd like to stick to
one-account-per-user (each account is
2006 Mar 29
5
Problem with setting ringtones on Cisco 7960 phone.
Hi All,
I am running into a problem setting the ringtones via _ALERT_INFO on the
Cisco 7960 phone.
I am using * 1.2.1 and have tried setting the variable to several
values. I have also tried setting the phone's software to both 7.5 and
8.2 thinking that it might be a version issue, but with no success.
I have examined the packets and do see the ALERT_INFO header being sent,
but the
2007 May 30
12
False ring problem
Hi all,
when a user dials any number, asterisk automatically generates ringing which
caller can hear, and after 2 - 3 rings asterisk detects that the called user
is busy, then caller hears busy tone. for example user hears---
tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
at the start so that user hears only beep beep beep if the called user is
busy. I have used the R
2008 Aug 06
2
shared mysql connection in dialplan
hi all,
i just finished developing some incoming call features in a macro. that
macro gets executed everytime an incoming call is received and a new mysql
connection is made using the MYSQL cmd in dialplan. i want to use a single
mysql connection for every incoming call.
my idea of doing it is like this, i want to get a mysql connection in a
global variable, just to share the connection with
2007 Aug 17
4
Call Limits
Hi all,
Some of my asterisk users have used their maximum call limit for incoming
calls (peers). There incoming call limit should automatically reset to zero
after hangup but its not happening and they no longer can recieve any calls
as their allowed limit is already full. So is there any way to reset the
call limit on peers by commands or do i have to restart my asterisk server?
--
Best Regards
2007 Apr 19
1
CDR(dst) != CALLERID(dnid)
Hi guys,
i just came to know that CDR(dst) field is set to current extension instead
of the dialed no. i need to set it to DNID because our every user has 5 dids
and i want to show the caller at the end of the month which numbers he
dialed for every call, along with other cdr info. Our rating depends on the
dialed number also. here is my extensions.conf
exten=> 1212,1,Dial(SIP/rizwan)
2007 Oct 24
2
Remote provisioning for ATA's
Hi all,
I need a fully developed web based remote provisioning system. I cant find
anything reliable on the internet. Have already checked ataconfig.com and
voxilla-ays.com. have tried to contact them but got no response. So if
anybody knows a good provisioning system then plz tell me about it.
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
-------------- next part
2007 Sep 11
3
Prevent multiple sip registrations
Hi all,
Is there anyway i can prevent multiple sip registrations from different IPs
using single username in asterisk. Does asterisk provide any aid in this
respect? As far as my knowledge is concerned i dont think there is any
support for this in asterisk, so i think i'll have to makeup a script which
sniffs sip packets coming for asterisk and detect for multiple register
requests coming from
2007 May 09
3
The 'h' extension problem
Hi all,
There is a problem with my dialplan. here is the dialplan:
exten=> 123,1,Dial(SIP/U1,,Ttg)
exten=> 123,2,Hangup
exten=> h,1,AGI(onhangup.pl)
The problem is whenever U1 is called or calls someone, if U1 hangsup the
call then the h extension is NOT executed. but if the other person hangsup
the call, then the h extension is executed (assuming that the other person
is calling
2008 Jan 25
1
Need sample configuration files for sipura/linksys ata
Hi all,
i need sample xml configuration files for linksys pap2, linksys pap-2t,
sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are
linksys/sipura products. So if anyone has these sample files then plz share.
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2008 Nov 17
2
two sip listening ports for single asterisk
Hi all,
We are planning to shift our sip users from one platform to another.
(basically from one asterisk server to another). the problem we are facing
is that both asterisk servers are using different ports to listen for sip.
and both have live customers on them. provisioning their ata's is not a
good option for us coz of our settup. we cant just ask the customers to
change their ports for
2007 Mar 14
4
what happened to asterisk wiki???
Hi
im trying access the www.voip-info.org website since yesterday but i cant
open it. google search diaplay correct search results but it doesnt open
when i click the link. it displays a message about tcp error which says
-->"There was a problem communicating with the server". I dont know what the
problem is. I just want to ask whether their server is down or not and is
everybody
2007 Oct 29
2
XML file for spa devices
Hi all,
i need an XML file format which is used in remote provisioning of different
spa devices. Please somebody tell me the format or tell me where can i find
it on the internet. I also need a list of parameters which are configured
using auto-provisioning.
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
-------------- next part --------------
An HTML attachment was
2006 Oct 23
0
SIP_HEADER function; what names are available?
Minor update - use the following:
> if (strcasecmp(data,
> "x-Asterisk-Request-URI-pseudo-header")==0)
> {
> ast_copy_string(buf, p->initreq.rlPart2, len);
> -----Original Message-----
> From: Steve Langstaff
> Sent: 23 October 2006 09:58
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users]
2007 Aug 01
3
How to use stun server?
Hi all,
This is the first time i am using stun with asterisk for nat problems. I
have read the rfc which describes how stun works. i didnt have any problems
understanding it. I have also intalled the stun server called stund which i
downloaded from sourceforge. I have seen on the list that most people use
stund here. I have started the stun server and its running silently. Now i
dont know what to