Displaying 20 results from an estimated 2000 matches similar to: "Request for best practices"
2010 Dec 10
1
1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra
Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i.
On 9133i and 57i:
#<extension># works for a blind transfer.
Xfer<extension>Xfer doesn't!
All this worked on 1.6.2.14.
Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an
outside call, and tries to transfer it to 145 using the Xfer button:
-- SIP/169-0000009c answered
2006 Jan 26
4
extension to extension dialing
Sorry for all the newbie questions. I really appreciate everyone's help
today.
Okay I've got outgoing and incoming calls working with no echo. yay! Now
I'm having an issue with SIP extension to extension calling. Any time I
dial another extension it goes right into voice mail. My
extensions.conf is pretty small and rough but, here's what I have right
now. Most of it was taken
2006 Feb 27
7
TDM400P digium card
Okay everyone -
I'm moving away from using sipura 841 phones. I'm starting to test with
Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but,
for now we have a digium tdm400P with 4 analog lines coming into it. So
my question is will upgrading the IP phones with my existing digium
tdm400 card be enough to satisfy my users ? or is it really a combo
deal needing to
2006 Apr 05
2
What causes deadlock?
Hi
What causes deadlock?
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x82acb10', 10 retries!
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x8298160', 10 retries!
Here is the portion of the log:
Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)...
Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing
2004 Dec 13
1
Repost: Cisco 7960 and Asterisk...not working....
Anyone help me here? I am a newbie so be gentle ;-)......
It worked once and then I played with the configs.
I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22
I have the 7690 with a SIP iamge (Whatever latest is )
I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here
2009 Feb 26
2
Problems with Outbound Calls
Hi everyone!
I'm quite a newbie at this Asterisk stuff so please bear with me.
We've recently decided to start training in Asterisk via AsteriskNow!
Asterisk version is 1.4.18.1 through AsteriskNow! 1.02
The box we have is paired with a Digium TE110P and we've managed to get
it to the point where incoming calls via a single DID (from NTT Japan)
can be received and answered
2005 Jan 27
1
Stumped by BroadVoice SIP
Hello guys.
I am a fairly new user to Asterisk, and I'm just having a tough time.
My goal is to set up a VOIP PBX. I have signed up with a BroadVoice
number, and I have three systems with SIP phones.
The PBX and the SIP phones are all behind a Cisco PIX running NAT.
I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with
little luck.
The SIP phones are two X-Lites on
2005 Jul 15
2
[Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i
don't know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
context = default ;x changed from default to sip
localnet = 192.168.100.0/24
2004 Apr 07
2
error 488 - Not Acceptable Here
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
are registered by the below information
*CLI> sip show peers
Name/username Host Mask Port Status
2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored
2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored
2000/2000 192.168.22.198 (D)
2005 Aug 27
2
Problems with registration
My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP
phone.
Here is my sip.conf file:
;
; SIP Configuration
;
[general]
context=default ; Default context for incoming calls
port=5060 ;added
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ;
2005 Mar 15
2
Setting up Security Groups
I appologize for the long, new-ish question, but after a few days of trying to work a solution by reading through the list archives and WIKI and coming up with what I thought would work, I think I'm just not getting a fine detail.
I titled this thread "Setting up Security Groups" because I'm trying to set up some sip user groups with certain calling rights, e.g., one group of
2003 Oct 11
2
"context confusion" internal context 2 context only?
I'm trying to create several contexts for extentions with
different levels of access to features and I'm wondering
how the heck do I include all the contexts so that you
can call internal to any extention in another context without
giving the features of the higher level context to the lower
level context?
ie.....
[admin]
include => local
include => longdistance
include =>
2003 Nov 12
3
Dial Plan Sequencing
I have an interesting dilemma with sequencing in the dialplan. Up to
now, I have assumed that the extensions in the dial plan were tested in
the order that they appear in extensions.conf. In other words, I have
the following fragment which was designed to dial toll free on the PSTN
and all other long distance on VoIP:
[longdistance]
include => local
2004 May 24
2
Newbie extensions.conf I need to include [SMS] context.
I want to include a new context in my exensions.conf
I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan
and I can sort of follow it?!
I have a context [local] that I know zapata.conf points to, I have edited
extensions.conf and put in my phone, sip and iax extensions. I want to add
an sms context.
I understand that all calls go through my [local] context and I have
2009 May 27
3
1.6.0.9: Now "Unable to create ... 'DAHDI'"
Still trying to upgrade to 1.6.0.9 for 1.4.
It worked - it worked all day yesterday, but this morning:
-- Executing [646xxxyyyy at longdistance:1]
Answer("SIP/172-08276a60", "") in new stack
..........
-- Executing [646xxxyyy at longdistance:6] Dial("SIP/172-08276a60",
""DAHDI/g2"/1646xxxyyyy") in new stack
May 27 09:56:57]
2008 Jan 25
1
Problem with FollowMe
I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed
the WIKI page on setting it up but I can't seem to get it to work.
Here is my Asterisk version:
pbx1*CLI> core show version
Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on
2008-01-10
12:08:48 UTC
Here is a log of when the FollowMe is being called:
NOTE: I've tried to use the AstDB as
2003 Oct 24
4
Context restrictions
Can someone please explain what I am doing wrong here? I only want the
extensions listed in long-users to be able to access the longdistance
context.
If I do this, I get a congestion tone no matter what I dial. If I add a
[default] context and include => longdistance, then the local callers
can call the long distance number fine, which is not what I want, but I
still want long-users to be
2005 Mar 17
1
Include/Macro not working right...
Hey guys. Thanks for the help on the Pattern matching, I got that
working pretty nicely.
the next problem I have is that I'm using an include file, but its not
really working...
In my extensions.conf:
[incoming]
exten => _NXXNXXXXXX,1,SetCallerID("Unknown Called Number")
#include "numbers.conf"
exten => _NXXNXXXXXX,3,Macro(Number,1000,${EXTEN})
[macro-Brand]
2003 Jul 16
8
Call Pickup
Hi,
I have been trying to workout how to use the call pickup.
So Far, I have the following in zapata.conf
[channels]
signalling => fxo_ks
context => local
pickupgroup=1
callgroup=1
channel => 1-3
When I dial *8# all I hear is busy tone.
What have I missed?
thanks
Jay.
2005 Mar 10
2
Cisco and Asterisk
Hey all,
I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get
a bit of help here.
First I'll explain my setup, and then my problem.
Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO
ports. I have an analog phone line plugged into the first port
(voice-port 1/0/0). I've got it setup so that calls coming into that
analog line are