Displaying 20 results from an estimated 200 matches similar to: "domain name display issue in linux pc"
2005 Oct 09
1
Problem setting SIP incoming/outgoing
Hi
I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my configuration in sip.conf
[general]
register =>
2008 Jul 22
1
issue with high latency
Hi,
Is there a specific latency that asterisk accepts? I encountered a
problem wherein when the latency was unusually high,my xlite's (i have 2
xlite) cannot register. but when the link suddenly went stable, the
x-lite just registered. what i forgot to look at is if the registration
packet is reaching my asterisks.
------ when xlite cannot register ---------------
Pinging
2009 Nov 07
1
Trouble registering Cisco 7942
I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960
and 7912 currently connected and functioning. I'm trying to use the
recommendations from here:
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
I have created a "XMLDefault.cnf.xml" and it took the latest image but
the phone states it's unprovisioned? Any
2004 Nov 29
1
Polycom Reboot Script PRI errors!!
Kevin wrote:
> There is a reboot script posted on the wiki to reboot Polycom
> telephones. When I execute this script, I get the following messages.
> I am concerned as this is causing issues with asterisk and the PRI.
> Does anyone have any ideas why this would be happening?
>
>
>
> asterisk console:
>
> -- Remote UNIX connection
> -- Remote UNIX
2004 Jan 22
2
Polycom Reboot Script - Please wiki-size me
With my thanks to Brian West and his offering in the thread,
"Subject: Re: [Asterisk-Users] Remote reload Cisco 7960"
I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones
PolyReboot.pl takes an IP address as a single argument and reboots the
phone.
You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg - all
lower case. Further,
you need to use ftp for
2005 Jan 13
3
SER vs Asterisk for SIP
Why is SER considered a better SIPserver than asterisk , why is it that SER
can handle more clients than asterisk can. And if this is just cause of say
poor SIP handling code in asterisk then is there anything being done to fix
it. Just wanted to know why SER claims to be better than asterisk as a SIP
server. ?
--
regards
Vikram (http://www.vicramresearch.com)
2010 Jun 17
1
Asterisk no audio on calls problem.
Hi there,
I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also.
I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP.
The configuration is a follows
Asterisk PBX 10.202.17.217/24 ------>|
2004 Jun 16
2
embedded Asterisk
Hi All,
I have a thin cliente here that i want to run asterisk:
- National Semicondudor Geode GX1 266MHz Geode 266MHz single chip
- NS Cx5530a Southbridge National Semiconductors SC2200
- NS PC97317 in chipset
- 32MB Compact Flash
- 64MB Ram
- 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816
Some tip?
I have a ide>flash
2008 Dec 22
1
Web-driven SIP call thru Asterisk IPBX
Hi,
I think that the web-driven SIP Phone (free) doddle (beta version) can be useful with your Asterisk applications.
You can pre-fill it with your sip settings (Asterisk host name or IP /?realm / sip user), you just need to setup the HTML link as that: (Attached is the HTML page example)
?
/**************************/
simple HTML code example:
/*************************/
<html>
<head>
2004 Jan 16
1
ERROR[8192]
Hi all!
I get this error when trying to start asterisk:
ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk
What can be the problem?
Thank you!
Miklos
iPFONE Telefonia IP
Rua Caio Graco 735 S?o Paulo SP
iPBX +55 11 3801-3702
UK +44 870 - 3403539
FWD 64662
sip:ipfone@sipserver.com.br
www.ipfone.com.br
info@ipfone.com.br
-------------- next part
2008 Feb 23
1
Reg. VNC server and Windows and Centos interworking
Hi all,
I have a Windows XP PC and a Centos PC (Centos 4.3 release, kernel- 2.6.9-34.EL) which are in the same LAN. Currently, I am able to use VNC server on the centos to view the Windows PC, but not able to view the Centos PC from the Windows PC. This applies for even file sharing also. The samba server is running and I can view the windows network from Centos PC but in the Windows network, not
2007 Aug 31
0
chan_sip.c:5495 sip_reg_timeout: ERROR
Hello,
I?ve been using Asterisk 1.2.18 for a while, and today, with no apparent
changes, I started receiving these messages:
Aug 31 13:26:57 NOTICE[27528]: chan_sip.c:5495 sip_reg_timeout: --
Registration for 'user at sipserver' timed out, trying again (Attempt #19)
All trunks and extensions went to:
sipserver:5060 user 120 Request Sent
011
2008 Feb 19
1
SIP Request: OPTIONS
Hi,
I have register a sip user to sip server. I can see after registration * is
sending periodic "SIP Request: OPTIONS" messages to server. but it's not
getting back any response that should be SIP 200/OK as the documents say.
3130.299707 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip:
sipserver.net
3131.299513 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip:
2004 Dec 18
1
Setting up asterisk for one user in private ip NAT.
Hi.
I've just bought SIP telephony service from a Swedish telco.
I've managed to make and receive calls with kphone.
Now I want to set up asterisk to be able to add fancy features like
voice mail and recording conversations. But first I
have to get the basic setup right. I'm running asterisk and kphone on
the same machine, behind at NAT-router.
When I make a call (from my regular
2008 Jun 12
2
Reg. setting Domain name on Cento 5 pc
Hi all,
I am running centos 5.1 and I wish to change the domain name and
dnsdomainname of my PC. currently the settings are--
$ hostname
sipx.com
$ hostname --fqdn
sipx.com
$ domainname
(none)
$ dnsdomainname
com
I have searched in the net for tips but everywhere only the hostname change
is provided. I need to change/set the domain name and the dnsdomain name on
my pc to sipx.com and this
2005 Sep 23
0
Problem with outbound calls
Hi everybody,
I have some problems making calls from a sip user (HT286) to the pstn trough
Digium Wildcard TE110P, i allways have an error : SIP 403
INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd
From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>
2008 Feb 08
1
Reg. PPPoE server installation
Hi all,
I am trying to install the PPPoe server rp-pppoe-3.8 from Roaring penguin on centos PC (Centos 4.3, Kernel release 2.6.9-34.EL) but not able to start it. When I start it it gives the error--
/etc/rc.d/init.d/pppoe status
pppoe-status: Link is down (can't read pppd PID file /var/run/pppoe.conf-pppoe.pid.pppd)
ror -
/etc/rc.d/init.d/pppoe start
Bringing up PPPoE
2008 Feb 07
1
Reg. Kernel switching
Hi all,
I am a newbie to linux and this is my first mail to this group. We
have a PC running Centos version 2.6.9-42.0.2.EL. I needed
to load a router software called Nistnet and it would not run
properly. It is looking for RTC (Real Time clock) support as a
module.
I 've gone through the LKM Loadable Kernel module support under
Documentation pages of Linux.org and did the following:
2003 Aug 04
14
Mysql CDR
hello all,
I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record.
Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault.
the original version of cdr_mysql.so works fine but I need the start time and end
2003 Mar 14
3
SIP registrations
Can asterisk act as a SIP registrar or location server?
I would like to be able for a user agent(client) to register with
whatever client they are using as "username@domain-name.com". Rather
than the entry/username/password that is setup in the sip.conf file.
That way a user could log into any SIP enable client and their calls
would follow them around.
I have read the sip.conf man pages