I’m trying to set the callerid on a SIP call:
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I’m trying to set the callerid on a SIP call:
On 2012-05-25 16:47, jose carlos de souza wrote:
> I need help.
> Sorry for my bad english, i’m from Brasil. And just starting to leran
> Linux because of Asterisk.
> I want to configure Asterisk 1.8 with ipv6 in my Lan network at home.
> I’ve done a clean install of it and SIP clients it’s working well in my
> home with Ipv4 (execept of the bad quality of the voice, that it’s
> coming out, but that’s not important, what i relly want to see it’s ipv6
> I’m using Debian 6.0.4 on a Virtual machine (VMware), Linphone 3.5.2 as
> my SIP client and a Router DIR-600 from D-LINK as DHCP sever to my LAN
> networking. (With a DHCP IP Address Range of: 192.168.0.100 to
> What i change in configurations to work with ipv6 was:
> change my network interfaces in /etc/network/interfaces , i add this:
> iface eth0 inet6 static
> addres fe80::c0a8:6e
> netmask 64
> gateway fe80::c0a8:1
> Change sip.conf
> UDPbindaddr = ::
> TCPbindaddr = ::
> On Linphone check the box “use IPv6 insted of IPv4″
> Your SIP Identity: sip:9005@fe80::c0a8:6e>
>I alredy put the brackets. Like: 9006@[fe80::c0a8:6e] and still dosent work.
> I’m calling from a windows 7(the one with the virtual machine on it) to
> another windows 7 both with dual-stack ip protocol, and both are in my
> LAN network.
> The call’s only works when i use the IPv4. for example:
> when my linphone its configure to sip:firstname.lastname@example.org and Proxy
> addres: 192.168.0.10.
> I have created on Asterisk 10 SIP extensions, 9001 to 9010 on.
> I whant to know what im doing wrong.
> Or what files do i have to changes and where are they located? Wicth
> changes should i made?
> Thanks for the attention and help.
Some time ago I’m using Asterisk (currently 184.108.40.206) at home to manage
the calls. Nothing yet very complex, just something compiled by me using
the source code from the official site of the project and configuring
the files manually to both Asterisk and DAHDI. For now I’m not using any
GUI, but when I have more time, I plan to try something in the future,
for example, to make a statistic of the calls.
But, thinking about the statistics of the calls, in the last days I was
taking a look at the /var/log/asterisk/cdr-csv/Master.csv file, which I
understand is where the calls are registered. But all seem to have a
“ANSWERED” state, even those receiving a busy tone. This happens with
both internal calls between SIP extension and from SIP to PSTN.
A test I did is putting a Grandstream BT200 on DND mode (Do Not Disturb)
and call it from a softphone. While the softphone receives the message
that the extension is busy, the CDR registered the call as “ANSWERED”.
Not sure if it’s something usually due to the way it is configured the
dialplan or any other configuration issue.
Thanks in advance for your reply.
I have two asterisk servers connected via iax.
home_server < => IAX2 < => clinic_server
I’m just testing, calling from “home_server” via “clinic_server” but I’m getting an error message:
Call gets through to “clinic_server” but will not call back.
Dial(“IAX2/home_server-957″, “IAX2/home_server/218,30,rw”) in new stack
[Apr 14 16:59:45] WARNING: app_dial.c:2218 dial_exec_full: Unable to create channel of type ‘IAX2′ (cause 20 – Unknown)
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Hi, I have a question regarding NAT – I have two Asterisk setups, and a couple of softphones on my laptop to test them. In the first Asterisk I’ve got nat=yes for all SIP phones. The second setup is identical as far as software is concerned, but the server is running on a VPS with one of the larger VPS hosting services.
On this second setup I was able to phone out from my XLite softphone but when I tried phoning in nothing happened, basically because the phone was always UNREACHABLE. It would register fine with Asterisk but then disappear. After nosing around for quite a bit I found a suggestion that I try setting nat=route for the SIP phone, and suddenly it worked both ways.
Another SIP phone on the same laptop connects to the first Asterisk server setup (which runs on a dedicated box with fixed IP, not in VPS) but I notice that every minute or so Asterisk tells me that the phone is unreachable, then a few seconds later it becomes reachable again.
My laptop is currently sitting at home with typical home-Internet configuration (ADSL, Nat, no fixed IP).
I did see something somewhere about the big VPS providers using some form of “hidden” NAT but I don’t know what that could mean.
My question is: Does this difference in behaviour have something to do with the second server running on a VPS – and are there any drawbacks to using nat=route on all client SIP phones?
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Am 26.01.2012 18:43 schrieb “Steve Edwards”
> On Thu, 26 Jan 2012, eherr wrote:
> It is accessible from HTTP.
>> However, the access list only allows access from my home and the password
>> is strong.
> Can you configure it to ‘syslog’ accesses where you can monitor it.
> Maybe your access lists are invalid, misunderstood or not being honored.
> Thanks in advance,
> Steve Edwards email@example.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
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On Fri, 6 Jan 2012, Dale Noll wrote:
> I found the following lines to be helpful.
I think a ‘better practice’ would be to put the ‘stuff likely to change’
into the environment variables of the Asterisk process so they will
‘trickle down’ to sub-processes like AGIs.
This way, when you upgrade Oracle, you don’t have to track down and change
all affected AGIs.
Something like this snippet from my Asterisk start up script:
I like to ‘ignore’ the environment of the process executing the script
that starts Asterisk and add in only what is needed — I’m a ‘parts left
out don’t get broken’ kind of guy
Can you give this a try and report back?
Just installed asterisknow 1.6. I can access freepbx. I need to test
system on my LAN. Which softphone is best to use? I’m running ubuntu
on Dell optiplex G260 desktop at home. I’m hoping to setup basic IP PBX
for incoming/outgoing calls. No video.