On Fri, 2010-03-19 at 20:14 +0100, Christian Victor wrote:
2010/3/19 tjoen tj...@dds.nl:
register = tjoen:mypas...@sip_proxy/1234
[sip_proxy]
type=peer
host=ekiga.net
I guess you need to register to the actual hostname, not the peers name.
register = tjoen:mypas...@ekiga.net/1234
Hi Everyone,
I have an annoying problem,
When get a call from outside to an internal extension, The caller hears
continous ring. It should be ring for 5 sec and wait , and goes like this.
Iwhere should be the problem?
Thanks
--
Hey everyone
The question that i am putting up will sound a bit odd as i am a
newbie to asterisk. I have downloaded and install the asterisk and had
a look at some of the configuration files like sip.conf, users.conf
and extensions.conf. Now my question is that I want to do voip with
another pc
Use x-lite softphone. There is no need for any hardware. First you'll setup
an extension in sip.conf and then create a dialplan context in
extensions.conf so that this extension actually does something. The best way
to learn how to do this is by reading chapters 4, 5 and 6 of freely
available book
Hi,
I try to get an 8 Port Junghanns BRI card working under dahdi. The card
works with zaptel but I have no success under dahdi.
I load the module with modprobe wcb4xxp. I dont get any errors but I
dont see the spans in /proc/dahdi. The output from dmesg remains empty.
I use the following dahdi
On Sat, 20 Mar 2010, Loic Didelot wrote:
Hi,
I try to get an 8 Port Junghanns BRI card working under dahdi. The card
works with zaptel but I have no success under dahdi.
I load the module with modprobe wcb4xxp. I dont get any errors but I
dont see the spans in /proc/dahdi. The output from
Hi,
I run dahdi_genconf but I think this step can only be successful if the
module has been loaded correctly and the spans are visible
in /proc/dahdi but this is not the case.
Loic.
On Sat, 2010-03-20 at 13:21 +, Jeff LaCoursiere wrote:
On Sat, 20 Mar 2010, Loic Didelot wrote:
Hi,
I
Hi Guys,
I have a need to alter the general timeout in Asterisk. I am wondering if
this is something that is hard coded into Asterisk code or if there is a
parameter that can be set somewhere.
For outbound, I am using x. and hence unless I append a # sign, I would have
to wait maybe 5 seconds or
Hi Everyone,
I have a provider who is asking me to send SIP signals through
111.111.111.111 and then media through Media 1: 222.222.22.222 and Media 2:
244.244.244.244. This provider authenticates by IP and I think is using
Sonus gear and hence they have some load balancer or something...
I have
bruce bruce wrote:
For outbound, I am using x. and hence unless I append a # sign, I
would have to wait maybe 5 seconds or so for the call to go through.
Is there
You really do need to give us a snippet of the outbound code.
Doug
--
Ben Franklin quote:
Those who would give up Essential
As soon as the dialed number matches one of the dial patterns defined in
extensions.conf or its included files, asterisk starts dialing it. The wait
you have is probably from the trunk provider's side because by default
asterisk doesn't start playing the ring tone unless it gets acknoledgement
das sandesh wrote:
We are using Yealink T28 phones. Asterisk version: 1.4.21.2, dahdi: 2.2.0.2
This sounds like a few bugs which were opened (and recently closed) related to
call transfers. I'm not sure when those bugs were introduced, but upgrading to
a
newer version of Asterisk may resolve
CHEN XUEQIN wrote:
I have a similar problem when using AGI for call control. Also
udp port leak for some incomplete call. I wonder if the problem
is related to issue 16774.
Only way to know would be to reproduce on a development machine, and then try
testing the patch on 16774 to see if the
cool dude wrote:
hi leif,
thx for replying. can u plz ellabroate how to use 'o' optioan in Dial so
that callerid should work.
There isn't much to elaborate on. Just enable the 'o' option in Dial like any
other option, and that should pass through the callerID based on the
description
you
Thanks Zeeshan,
I'll try that and will revert back to you.
On Sat, Mar 20, 2010 at 4:10 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Use x-lite softphone. There is no need for any hardware. First you'll setup
an extension in sip.conf and then create a dialplan context in
extensions.conf so
Thanks for the input. I am using A2Billing and it takes long time to
authenticate PIN number and to dial destination number. If # sign is used
then it's a different story and it goes through quick.
-Bruce
On Sat, Mar 20, 2010 at 11:16 AM, Zeeshan Zakaria zisha...@gmail.comwrote:
As soon as
Seems like pattern matching needs to be fixed in some config file. Can you
give example of a number you dial?
--
Zeeshan A Zakaria
On 2010-03-20 12:15 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the input. I am using A2Billing and it takes long time to
authenticate PIN number and to
i belong to india. i am making pbx using sangoma
fxo card. i want that when ever call comes to my PSTN line i should see
the no from where call is coming. so i have to configures
chan_dahdi.conf according to my region. i checked dahdi.conf and in
that they have mentioned for india
Does your regular phone shows callerid on this line. If the service provider
is sending the callerid, asterisk doesn't have to do anything special to
retrieve it.
--
Zeeshan
On 2010-03-20 1:25 PM, cool dude cool_dudeof...@yahoo.co.in wrote:
i belong to india. i am making pbx using sangoma fxo
Hi Thanks very much for reply it and helping me out.
This is the out put
-bash-3.2# ls -l /lib/modules/2.6.18-164.6.1.el5xen/
total 1280
lrwxrwxrwx 1 root root 54 Nov 6 23:31 build -
../../../usr/src/kernels/2.6.18-164.6.1.el5-xen-x86_64
drwxr-xr-x 2 root root 4096 Nov 3 17:31 extra
Sorry but i did not understand how did you built it?
Can you please break up for me?
Thanks
Dani
On 19 Mar 2010, at 12:47 PM, tjoen wrote:
On Fri, 2010-03-19 at 01:26 +0200, Tzafrir Cohen wrote:
On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote:
On Thu, Mar 18, 2010 at 6:56
Hello
I have a 4 span PRI board with Zaptel, and Im using it for a long time.
In the last days I noticed that the result of zap show status show a
ZTDUMMY but I never installed it:
o*CLI zap show status
Description Alarms IRQ
bpviol CRC4
T4XXP
You can comment ztdummy out in /etc/sysconfig/zaptel. As for affecting icmo
On 2010-03-20 5:16 PM, Joao Gomes Pereira gomespere...@startel.pt wrote:
Hello
I have a 4 span PRI board with Zaptel, and Im using it for a long time.
In the last days I noticed that the result of zap show status show a
Ztdummy enabled should not affect incoming calls.
On 2010-03-20 5:22 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
You can comment ztdummy out in /etc/sysconfig/zaptel. As for affecting icmo
On 2010-03-20 5:16 PM, Joao Gomes Pereira gomespere...@startel.pt
wrote:
Hello
I have...
--
Using spandsp-0.0.6-pre17, SendFax on 1.6.1.18 and ReceiveFax on
1.6.2.8. Sip.conf on both sides has t38pt_udptl = yes.
-- Executing [...@fax-tx-test:3] SendFAX(SIP/side-sip-0009,
/var/spool/asterisk/fax/20091113_1455.tif) in new stack
[Mar 20 17:05:34] WARNING[6433]: app_fax.c:178
Hi,
Edit your logger.conf, set messages in debug mode, make test incoming
and outgoing calls. Copy the log in message dirz3* and post.
Goke
On 3/20/10, Zeeshan Zakaria zisha...@gmail.com wrote:
Does your regular phone shows callerid on this line. If the service provider
is sending the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm testing with a Grandstream BT200 telephone and, according to I read,
it has a LED that blinks if for that extension messages were left.
In Voice Mail UserID, under the ACCOUNT tab, I put *100 that is the
extension in which my Asterisk
Hello,
if have a problem since I switched to asterisk-1.6:
When making an outgoing call through chan_dahdi, I
cannot hear anymore early audio, the asterisk generated
sound (as defined in indications.conf) is played.
Thus, I cannot hear announcements by the operator,
and when the line is busy,
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