Displaying 17 results from an estimated 17 matches for "sip_custom".
2008 Dec 01
2
Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"
...tcode=5260477782
amaflags=billing
context=Incoming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes
Asterisk 2 sip.conf
GNU nano 1.3.12 File: sip_custom.conf
[VoipDirect777821]
type=friend
host=141.122.139
username=VoipDirect777821
secret=wsPiOov8830
accountcode=5260477782
amaflags=billing
context=Incomming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpse...
2008 Apr 04
1
Friday April 4th @ 12 Noon EDT: VoIP Users Conference (Asterisk!)
Yes, we mostly talk about asterisk, hardware, software, phones,
people, events and things asterisk-related. Asterisk is a registered
trademark of Digium. Asterix is a registered trademark of Ren?
Goscinny. We never discuss that, though.
Here's the short URL for sending out to others that might be
interested: http://x2z.eu
It lands here: http://www.VoipUsersConference.org
Once again, I
2009 Aug 14
2
no ring tone
...n't ring.
Edit sip_nat.conf for proper NAT:
localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here)
externrefresh=10
fromdomain=DOMAIN.com (Set your external domain name here)
nat=yes
qualify=yes
canreinvite=no
Add extra codecs to /etc/asterisk/sip_custom.conf
allow=gsm allow=h261
allow=h263
allow=h263p
videosupport=yes
_________________________________________________________________
Windows Live?: Keep your life in sync.
http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009
-------------- next part --------------
An...
2005 Aug 16
1
problems with eyebeam - video phone
...dress to bind to (all addresses on machine)
disallow=all
allow=h263
allow=gsm
allow=ulaw
allow=alaw
; H.263 is our video codec
; allow=h263p
; H.263p is the enhanced video codec
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
And I left only H.263 basic in codec's configuration in Video Phone.
No chance to get the communication in H.263 protocol.
I saw that to use H.263+ protocol I need Asterisk CVS.
I am not using asterisk CVS
I am using asterisk 1.0.9 (last stable version a coup...
2010 Feb 02
0
Issue when reloading
.../etc/asterisk/sip_general_custom.conf': == Found
== Parsing '/etc/asterisk/sip_nat.conf': == Found
== Parsing '/etc/asterisk/sip_registrations_custom.conf': == Found
== Parsing '/etc/asterisk/sip_registrations.conf': == Found
== Parsing '/etc/asterisk/sip_custom.conf': == Found
== Parsing '/etc/asterisk/sip_additional.conf': == Found
== Parsing '/etc/asterisk/sip_custom_post.conf': == Found
== Parsing '/etc/asterisk/users.conf': == Found
== Parsing '/etc/asterisk/phoneprov.conf': == Found
-- Reloadi...
2011 Jul 13
1
Connect Avaya to Asterisk PBX
...t=1720
callerID="ALT Asterisk PBX"
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
context=internal
[Avaya]
type=friend
context=internal
host=10.1.129.247
port=1720
canreinvite=no
disallow=all
allow=alaw
dtmfmode=inband
*Here is also the content of sip_custom.conf:*
[general]
context=internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=10.1.129.231
srvlookup=yes
conreinvitte=no
[1000]
type=friend
secret=malvin123
host=dynamic
dtmfmode=inband
disallow=all
allow=all
nat=yes
Thanks & regards,
Malvin
-------------- next part ---------...
2005 Jun 01
0
Segmentation Fautl / Core Dump with G.729
...t = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=alaw
allow=g729
allow=g723
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
language=es
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
----------------------------------------------------------------------------
---
SIP_ADDITIONAL.CONF
----------------------------------------------------------------------------
---
[as5300]
type=peer
qualify=yes
host=xxx.xxx.xxx.xxx (AS5300 box)
----------------...
2010 Apr 30
0
Problems with t38modem and bitrate sent to t38-termination service
...9;t know if this is an Asterisk issue or a t38modem issue.
My configurations are:
/etc/asterisk/udptl.conf:
[general]
udptlstart=4000
udptlend=4999
udptlchecksums=no
T38FaxUdpEC = t38UDPRedundancy
T38FaxMaxDatagram = 400
udptlfecentries = 3
udptlfecspan = 3
use_even_ports=no
/etc/asterisk/sip_custom.conf:
[t38modem-options](!)
type = friend
host = 127.0.0.1
context = fax-out
canreinvite = no
disallow = all
allow = ulaw
t38pt_udptl = yes
t38pt_rtp=no
t38pt_tcp=no
dtmfmode = rfc2833
nat = no
qualify = yes
[T38modem0](t38modem-options)
port = 6060
[T38modem1](t38modem-options)
port = 6061
[T3...
2005 Jul 06
1
SIP/2.0 403 Forbidden
...roblems.
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
[1000]
username=1000
secret=abc123
context=mytest
host=dynamic
-----------------
/etc/asterisk/extensions.conf :
[general]
static=yes
writeprotect=yes
;Suport phones
;SUPPORTPHONES=SIP/2205&SIP/2206&SIP/2207&SIP/2208&SIP/2209
[globals]
XLITE=SIP...
2007 May 21
3
Aastra MWI
I need to setup MWI on a few Aastra 9112's. I've tried doing so in the
web interface by setting "Explicit MWI Subscription" to true, but no
lights, no stutter tone.
Firmware: 1.4.0.1048
Thanks!
--
Warm Regards,
Lee
2005 Aug 20
0
Help needed receiving incoming calls.
2010 Mar 09
0
DUNDI Sip authentication failure
...nounsolicited,nocomunsolicit,nopartial
;priv => dundi-priv-via-pstn,400,SIP,192.168.199.21/${NUMBER},nopartial
[00:40:48:B2:78:6B]
model = symmetric
host = 192.168.199.23
inkey = 192.168.199.23
outkey = 192.168.199.21
include = priv
permit = priv
qualify = yes
order = primary
*/etc/asterisk/sip_custom.conf
language=fr
nat=never
;Subscribecontext=ext-local
[priv]
type=friend
dbsecret=dundi/secret
context=dundi-priv-local
host=192.168.199.23
qualify=yes*
/etc/asterisk/extensions_custom.conf
[ext-local-custom]
;for Direct IVR dialing if IVR is installed on the PBX B
exten => _36X,1,Macro(dund...
2005 Jun 24
4
UTStarcom F1000 WiFi IP Phone Review
I bought a UTStarcom F1000 WiFi IP Phone from
http://www.luxoncomm.com and tested it with Asterisk.
This is a my first impression of the device.
The F1000 supports SIP. It looks and operates like
a cell phone, and connects to the Internet through
WiFi, so you can use it at any WiFi hotspot. I set up
a 802.11b wi-fi network with a Linksys BEFW11S4
Wireless-B broadband router with no security
2006 Apr 26
2
Unable to accept incoming PSTN calls
...e:
[general]
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)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
#include additional_a2billing_sip.conf
extensions.conf:
zapata.conf file:
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Wh...
2006 Jun 09
3
FXO registration and VegaStream
I am trying to configure a VegaStream 50 FXO to work with asterisk. The
problem that I am having is that the VegaStream does not support incoming
registration from asterisk. VegaStream only allows outbound registration.
My question is does asterisk allow incoming registration from an FXO? If yes
how? Or better yet, has anybody been able to make the VegaStream FXO work
with asterisk? According
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
...closest proxy server
host=proxy.mia.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=3055741428
secret=MYSECRETPASS
context=from-broadvoice
;Disable canreinvite if you are behind a NAT
canreinvite=no
;Don't try to authenticate on incoming calls
insecure=very
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++
EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++
EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++
EXTENSIONS+++++ EXTENSIONS+++++ EX...
2012 Jul 05
7
FreePBX: using context other than the default context and the generation for the configuration
Hi All;
If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal).
Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I