in setting up 79xx on sccp, with sccp-b library, and tftp
server, which part is the main problem for you?
best
On Wed, Feb 16, 2011 at 3:10 PM, Andrew Latham lath...@gmail.com wrote:
On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote:
Hi,
Anyone who has deployed Cisco 7945G
Hi,
Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your
experience.
/ag
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lolz, I agree. Better to spend more and use it for some time. It is not a
big installation about 4-5 extension so can spare the budget for it easily.
/Khurram
On Sun, Feb 13, 2011 at 5:45 PM, Michael Graves mgra...@mstvp.com wrote:
On Sun, 13 Feb 2011 09:28:19 -0700, Joel Maslak wrote:
My
Thanks Gordon, Grandstream is in my purchase list : )
/ag
On Sun, Feb 13, 2011 at 10:58 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Sat, 12 Feb 2011, ast guy wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking
Probably I will go with cisco 7945g I hope its support is good with
asterisk. Have you used it ? is it simple in configuration?
/Khurram
On Sat, Feb 12, 2011 at 1:47 PM, Andrew Latham lath...@gmail.com wrote:
On Sat, Feb 12, 2011 at 9:31 AM, ast guy ast...@gmail.com wrote:
Hi,
I have
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone that works well with
asterisk server.
Pls suggest.
cheers
/ag
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Thanks for the comments, I will go through the detail and price and then
will buy accordingly,
cheers
/ag
On Sat, Feb 12, 2011 at 2:11 PM, Terry Brummell te...@brummell.net wrote:
Yes, I use provisioning for my Polycom's. And unfortunately, as far as I
know, the Mitel's do not support
Hi,
I have made a fresh install of asterisk-1.6.2.10 and when I register my
soft phone it gives following error. Rest are default configurations.
32.454370 MY_IP - ASTERISK_SERVER_IP SIP Request: REGISTER
sip:ASTERISK_SERVER_IP
32.454505 67.19.43.202 - MY_IP SIP Status: 401 Unauthorized(0
Hi,
I am facing terrible issue regarding no audio/voice on both sides. I am
using g729 codec on two machines and carrier also supports g729 codec. I can
see the RTP traffic flowing but there is no audio.
Call is going from Server 1 to Server 2. I can see the established SIP
channels on Server but
Hi,
I am using codec g729 on two asterisk machines, but when call is forwarded
from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs
following error and there is no audio. Also the IVRs being played have
choppy voice.
Insufficient information for SDP (m = 'audio RTP/AVP 18
On Tue, Nov 24, 2009 at 10:49 PM, Miguel Molina mmol...@millenium.com.cowrote:
ast guy escribió:
Hi,
I am using codec g729 on two asterisk machines, but when call is
forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1
outputs following error and there is no audio
This one is manual way of doing it.
You can get more details at page
http://www.voip-info.org/wiki/view/Asterisk+Starting+and+Stopping
It has provided the init.d scripts where you can automate the process.
/ag
On Sat, Nov 14, 2009 at 7:44 AM, Yawar Hadi yawarh...@gmail.com wrote:
cli stop
Hi,
I am trying to install asterisk-1.2.34 but facing following issue. I have
gone through it and found that there are files in /usr/lib
libpq.a libpq.so libpq.so.4libpq.so.4.1
make[1]: Entering directory `/usr/src/asterisk-1.2.34/cdr'
gcc -shared -Xlinker -x -o cdr_pgsql.so
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:
why don't you write an AGI which talks to asterisk manager API 5038 port and
executes the asterisk commands. You execute asterisk command via agi not
using system command
-ag
On Feb 11, 2008 11:24 AM, Rob Hillis [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi guys,
Hi,
I have a scenario that * Server A ( behaving as client) has sip peers, P1,
P2, P3 with different contexts. Peers register to another * or any other SIP
server. Using realtime * I am able to create a peer entry in sip buddies
table and a register statement in sip.conf on client side and it's
? It's unnecessary.
I believe that Dial(SIP/gs102/1234) will achieve what you want.
ast guy wrote:
Hi,
I'm trying to call a SIP server while providing the SIP server
username/password in dial string but it's not working ...
Dial(SIP/gs102:[EMAIL PROTECTED]);
User on sip server
Hi,
I have installed asterisk real time and sip buddies information is being
stored in DB. Now I have a question,
Asterisk Realtime Server -A
Third party SIP server-B
Question: Is there any configuration in * RT that it can register with
defined sip user on Server-B
I was only able to find sip
Hi,
I'm trying to call a SIP server while providing the SIP server
username/password in dial string but it's not working ...
Dial(SIP/gs102:[EMAIL PROTECTED]);
User on sip server (192.168.2.81):
[gs102]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=gs102
secret=test
host=dynamic
Hi,
Let me explain what I'm looking for a solution using asterisk.
I have one third party SIP based server (A) and on Asterisk server (B).
1. Extension-1 -- Server A calls Server B.
2. Server B does some processing and calls/sends back to Server A ---
Extension-2
3. SIP session has been
there is no /proc/zap folder .. can you tell how can I create /dev
nodes. I have tested the same configurations on FC5 and these device
links were created ...
drwxr-xr-x 2 root root 160 Jan 17 10:59 .
drwxr-xr-x 13 root root 3640 Jan 17 11:00 ..
crw--- 1 root root 196, 1 Jan 17
*Walter Willis,
*Thanks a lot, got the commands from zap Makefile and it worked, now can
create conference room, my question still stands why it didn't create
itself. Will go through make file to get an answer to that.
Anyone else facing the issue can resolve by running following commands
Hi,
I'm using zaptel-1.2.22.1 with asterisk-1.2.10 and following steps to
make zaptel working...
OS is gentoo linux 2006.1
Hardware:
-
:05:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
Subsystem: Unknown device 8085:0003
Flags:
Hi,
Can any one please guide how do I handle the SIP phone-context URI
parameter. I got following traces..
9.191690 IP_A - IP_B SIP/SDP Request: INVITE
sip:12599;[EMAIL PROTECTED]:5060;user=phone, with session
description
9.191942 IP_B - IP_A SIP Status: 404 Not Found
9.195656 IP_A -
Hi,
Just need to confirm whether dial() command provided options
h: Allow the callee to hang up by dialing *
H: Allow the caller to hang up by dialing *
work for SIP channels as well ?
-ag
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Hi,
It's really weired issue,I'm facing with asterisk-1.2.10 version. I
see SIP call sessions stuck in asterisk for hours and then somehow get
released. There happens to be an issue with BYE/CANCEL release msgs
b/w sip entities. Has anyone faced this issue before also rtptimeout
option given in
Hi,
I would like to know what is Asterisk Dynamic Loader. Let me
explain what I'm about to ask.
I have three Asterisk servers running my in-house built app_xyz.so
application. Now what I do to save time is compile application on one
server and scp app_xyz.so on rest of servers. All servers have
Hi,
I'm running Asterisk 1.2.10 on gentoo linux and facing strange kind of issue.
1. chan_sip.so takes about 10 secs to load up when asterisk starts.
2. When I dialout using SIP it takes 20 secs to output -- Called SIP
[EMAIL PROTECTED] and get ring back from B party...
Is there any config that
Hi,
I have downloaded iaxcomm version iaxcomm-lin-1.0rc3, when I try to
execute it it gives following error.
# ./iaxcomm
Error wxWindows Fatal Error : Couldn't Initialize IAX Client .
any idea what's going wrong ?
-ag
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I was going through Zap channel page on voip-info.org
Zap/channel-instance
channel is the channel number and instance is a number from 1 to 3
representing which of up to 3 logical channels associated with a
single physical channel this is.
for what purpose logical channels are used?
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