Displaying 20 results from an estimated 500 matches similar to: "--- Problem with queues.conf and extensions.conf ---"
2006 Jan 10
2
Problem with Action:Originate with ASterisk Manager
Hi Asterisk-users,
I am working with Aterisk Manager API's.
I can login successfuly with the following.
char buff[256];
strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n");
send(msock, buff, 255);
Now I want to try Action: Originate, therefore I tried the following
char buff1[256];
strcpy(buff1, "Action: Originate\r\nChannel:
2016 Feb 22
4
Windstream SIP Trunk settings
Does anyone on this list use Windstream as a SIP trunk provider?
If so, would you mind sharing your peer settings?
I'm using asterisk 13.7.2 and can't seem to get the inbound working
correctly (using registration). Outbound is fine, but they are seeing an
authentication error on their end.
Here are my inbound peer settings:
username=<accountnumber>
secret=<secret>
2005 Mar 21
1
Net2Phone / Vonage
I can cut and paste the log file from a reload right now, and provide
you with the other information when I get home after work:
tmp*CLI> sip debug
SIP Debugging Enabled
tmp*CLI> reload
Mar 21 14:52:42 NOTICE[23231]: indications.c:397
ast_unregister_indication_country: Removed default indication country 'us'
11 headers, 0 lines
Reliably Transmitting:
REGISTER
2005 Sep 23
10
Problem setting up TDM22B card
Hi All,
I have the problem setting up TDM22B card.
Steps what I have followed are:
[1] compiled zaptel-1.0.9.2 & installed the same.
[2] modprobe wcfxo
/lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including invalid IO or IRQ parameters.
You may find more information in syslog or the
output from dmesg
2005 Jul 11
0
Calls dropped upon 'native bridging' after IAX2 transfer
Skipped content of type multipart/alternative-------------- next part --------------
############
# amd BOX #
############
## Step 1
## Bob(ext. 6202) place a remote IAX2 call to the operator (ext. 6302)
## Reminder : _62XX are register on 'amd' and _63XX on 'dell'
-- Executing SetGroup("SIP/6202-d193", "IAX") in new stack
-- Executing
2004 Apr 18
0
AGI Module
Hey all,
I'm sorry to bother you with something so trivial, but I seem to
be having an issue with the Asterisk::AGI module. I am a relative
newbie with Perl so it could be a stupid syntax mistake that I missed.
It seems when I try to execute either the stream_file or the get_data
subs nothing is actually done. It doesn't seem to stream the files, but
on the console it says it played the
2005 Sep 27
1
failed make install on Solaris 10
I finally got Solaris to successfully make asterisk, using these
instructions:
http://sunfreeware.com/programlistsparc10.html#gcc33
Now though, when I issue the make install, I get this error:
mkdir -p /var/opt/asterisk/spool/system
mkdir -p /var/opt/asterisk/spool/tmp
mkdir -p /var/opt/asterisk/spool/meetme
install -m 755 asterisk /opt/asterisk/usr/sbin/
install: asterisk was not found
2006 Feb 02
1
Configuring Meeting Room from Asterisk Manager API
Hi All,
I want to do a three-party conferencing using manager api.
But I found out from the asterisk-users list that I *MUST* use
the meeting room concept.
I wanted to know wheather meeting room can be configured dynamically?
on the fly? Otherwise, configuring meeting room statically is not scalable.
Thanks
Regards,
Somesh S. Shanbhag
---------------------------------
Bring
2006 Feb 07
1
MeetMe - Party's are not exchanging Audio - Is this BUG?
Hi All,
I observed the following in my try towards Multiparty Conferencing.
I am establishing the Multiparty Conferencing through Asterisk Manager API.
I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader.
Following commands are used -
Action: Originate
Channel: SIP/111
Application: MeetMe
Data: |edwx
ActionID: ffe4563
When I use the above, Incoming call will
2005 Sep 27
1
R: Problem setting up TDM22B card
Hi,
I didn't get any solution in the mailing list.
[http://asterisk.linkx.net/asteriskusers/200409/msg01167]
What should be the next step?
Changing the machine???????
Is it machine dependent?...
Regards,
Somesh S. Shanbhag
--- Tzafrir Cohen <tzafrir@cohens.org.il> wrote:
> On Tue, Sep 27, 2005 at 12:13:21AM -0700, somesh s
> wrote:
> > Hi,
> >
> > I did
2005 Oct 18
3
CAPI - displaying individual MSN
Hi,
I'm currently using chan_capi-cm-0.6, with the following capi.conf:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de
[ISDN1]
msn=8304490
incomingmsn=8304490
isdnmode=msn
group=1
controller=1
softdtmf=1
context=demo
echosquelch=1
echocancel=yes
echotail=64
callgroup=1
devices=2
Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
so
2005 May 18
2
Call forwarding...
Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based -
2005 Sep 21
7
add 0 (zero) to incoming callerID - how?
I have an asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.
Simple, right?
This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial
2004 Sep 02
5
Any way to _always_ execute certain commands in a dialplan context?
I've got a need to do something like the following:
[foo-context]
exten => _.,1,SetCIDNum(123)
exten => _.,2,SetCIDName(XYZ)
include => local
include => tollfree
But of course, this example won't work. The goal here is this: if a call
ends up being handled by the "local" or "tollfree" contexts, I want
those SetCID*** commands executed. Otherwise, I
2005 Feb 25
1
SetCIDNum using SIP?
I am experimenting with my * server to use SIP with my long-distance
providers instead of IAX, so that the media path is from the end user
straight to the provider's gateway (hopefully reducing my bandwidth
consumption). I have it working with VoicePulse Connect but SetCIDNum
doesn't appear to work. Is this something with VoicePulse Connect only
or is it generally difficult to set the
2007 Feb 26
2
SetCIDNum is not available on 1.4svn
Hello,
I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it
on 1.4svn 56126 and it does not recognise this application. Any idea?...
Thanks! __Yehavi:
2005 Feb 01
2
Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
I tried to get callerid working the normal way but the cid is never passed
to the phone.
It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in
extensions.conf
which I found in the wiki:
http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc
Is this intended behaviour, or still a bug?
It does work but it only shows one zero even though I have
nationalprefix = 0
2004 Feb 03
1
Mediatrix sip fxo gateway workaround?
Possible Mediatrix 1204 fxo sip gateway workaround
Need some feedback from experienced * users relative to this workaround
please please please.
Problem: The mediatrix 4-port fxo gateway does not provide any mechanism
for * to select which "port" an outbound pstn call will use. (See lots
of previous posts over the past four days for more detail if needed.)
Our reseller has been
2004 Jan 16
1
Caller id and callback
Is there a way to append a 91 in front of all incoming caller id
numbers? What I am interested in is this - when a call comes in and the
caller ID comes across, it is in the format 9165551212. That would be
fine, but if I want to call them right back, and I choose that call and
'dial' on my phone, it fails because it hasn't dialed a 9 first. I
assume also that if there was a 9, it
2004 Jul 27
2
Enum
You can play also with www.enum2go.com <http://www.enum2go.com/> or
wap.enum2go.com
Regards
Alex
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040727/6c42c39d/attachment.htm