Displaying 20 results from an estimated 500 matches similar to: "Unwanted conference with snom320 and asterisk 1.07bristuffed"
2006 May 04
0
Unwanted conference with snom320 and asterisk 1.07 bristuffed
I have 13 Snom 320 with asterisk 1.07 bristuffed. The problem is that
sometimes on random basis, when one customer is placed on hold and another
call arrives, the customers are put in conference with each other. This look
very strange to me, but I've disabled the confernce button on the snom
phones to prevent the human errors, but it still occurs.
Investigating I've discovered that a
2006 Apr 27
1
Snom 320 HOLD and TRANSFER not detected
I have a preoblem with my snom 320 phones. I have 5 snom phones installed
and all of them have 5.2 firmware. All have same settings in the advanced
panel. On 2 phones when I press the hold or transfer key nothing happens and
* does not start the musiconhold. In the The hold and transfer keys are set
as F_R and F_TRANSFER correctly as the others. Other snoms and gxp-2000 work
ok.
Any ideas?
2009 Aug 01
1
SNOM Phones Displays NR Frequently
Hi,
I am using SNOM phones with Asterisks for few years. They used go occasionaly NR, and I would reregister them from the phone web interface. But it started doing frequently for last few months.
Here are SNOM Phone and the firmware version;
snom190-SIP - Version-Code: snom190-SIP 3.56m
snom320-SIP - snom320 jffs2 v3.36
snom300-SIP - snom300-SIP 6.5.2
Asterisk version - Asterisk
2010 Mar 09
3
Snom Provisioning
Hello all,
I've to deploy about 200 snom320 phones on a instalation.
Do you know any knid of tool to help me with this amount of phones?
I'm thinking in a provisioning tool which I use for setting up the
phones.
Any clue would be welcomed.
Thanks.
Voip-Crazy
2008 May 21
1
using gtalk received instant messages in the dialplan
I have been doing some reading about gtalk and asterisk. Most of it is
pointed to enable using gtalk for making phonecalls. Would it be
possible to use gtalk instant messaging input (just some text send to
the gtalk account configured on an asterisk box) into the dialplan.
This way you could use gtalk im to trigger all kind of events like
sending an sms, adding sip entries to the system,
2008 Jan 23
5
Snom 320 Lost Settings
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
Has anyone ever seen an Snom320 lose settings?
It's been working fine for months and then I got a call this morning
saying that it was asking for country, timezone etc.
I logged in remotely, and it had lost the server address, username,
password, mailbox and ringtone.
- --
Kind Regards,
Matt Riddell
Director
2013 Jan 22
2
Blind transfer behavior - Asterisk 1.8 and 10
Hi,
I want to check the status of a blind transfer (only sip endpoint)
between various phones. Transfer is working perfectly, using ## from
features.conf or using transfer key from phone, here SNOM320.
My problem is that if party to transfer to is busy, the transfer fail
and the call is ended. What I want to do is to return the call to the
party who originate the transfer.
I checked
2006 Apr 20
1
SPA-3000 Bug? Dropped calls while registering.
Hello All!
I am in the process of assembling an asterisk-based phone system for my office - using SPA-3000s to connect the network to the PSTN. I am wondering if anybody else can get (or has already seen) the same behaviour out of their 3000.
The short version: Send multiple Calls to the SPA's FXO port at the same time it is re-registering with Asterisk.
SPA HTTP Configuration:
2006 Jan 06
2
controlling SIP subscriptions from SNOM phones
We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one
phone setup as the receptionist phone, using hints to show busy office
lines. This all works as expected.
This is a new installation, and people are just starting to setup their
phones. For those of you not familiar with SNOM phones, there is a row of
keys on the right side of the phone which SNOM calls function keys. In
2006 Jun 14
1
SPA-941 Disable call waiting or Disable Call waiting via asterisk
I'm trying to disable call waiting for Linksys SPA-941, but
unfortunately as far as I have seen, there are no parameters on the web
interface regarding this feature. I just want callers to hear the busy
tone when the called party is at the phone. Probably I can accomplish
this by using the "disable call waiting" in asterisk as well, but I have
not been able to find any
2007 Apr 23
1
Asterisk+mISDN drops calls after 3-4 secs
Hi,
I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers.
I installed the new driver (0.3.1-rc30) on our pbx but since no voice
was passing I decided to go back to old version (0.3.1-rc23).
Last friday everything seemed to work fine but now every incoming
call drops after 3-4 seconds while Asterisk console is showing these
messages:
Apr 23 12:42:39 DEBUG[7625]:
2007 Jul 12
0
No subject
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to A
realm=192.168.0.2
context = default ;Default for incoming calls
[5549]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
type=friend ;(inbound and outbound calls accepted)
secret=localphone ; obvious password for testing
host=dynamic
callerid=Jason White <5549>
dtmfmode=auto
mailbox=5549 ;(Asterisk VM-system's
2006 Jan 26
3
snom 320 echo problems
Hi there -
I'm having some echo problems on my snom 320 phones. Anybody experience
this before ? I don't have any issues with the sipura 841s I have
though.
Any help is greatly appreciated.
Thanks !
Nora Lavelle
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2009 Jun 08
2
Snom, Asterisk and Patton SN1400 - sending bye instead of hold
Hey Everyone,
i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM
300,320,360 Devices.
In the combination with asterisk and the patton, there are occuring some
strange behaviour, due to the calling and answering everything works
good, clear voice, great availability.
All the devices have to use ulaw, alaw and slinear is available but
never the first choice since i use my
2010 Mar 02
0
1.4 chan_sip use internal IP for dialog-info+xml SUBSCRIBE, why?
Asterisk 1.4.29
BLF-SUBSCRIBE go to internal IP (ngrep output):
U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 -> 62.134.xxx.xxx:5060
SUBSCRIBE sip:12 at 62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP
212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From:
<sip:K922002626 at 62.134.xxx.xxx>;tag=vyx8c0trgx..To:
<sip:12 at 62.134.xxx.xxx>;tag=as13e7cb7c..Call-ID:
2003 May 30
1
A Major Problem!
hi,
we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know.
our set up is, an Asterisk server equipped with, 4 port station interface card ,single port fxo card and several soft sip phones
we have found problems with the following scenarios,
outside caller (calling through fxo interface) <------------------------------>
2006 Feb 14
9
Asterisk and Snom 360
Is anyone using the SNOM 360 as a reception console with Asterisk? We
are trying to have the ability to view whether an extension is on or off
hook, or ringing with the Snom, which seems to work fine. The issue is
that we are having difficulty picking up calls and transferring.
Anyone have experience / insight?
Darrell S. Long
Director of Technology
BestWeb Corporation
Phone 877-777-2932
2010 Mar 16
1
Asterisk + Sip Phone + BLF
Hi,
I used Grandstream (gxp2000, gxp2020) and Snom (370) SIP Phones, but
with 2 extensions BLF status does not work correctly.
have someone ever tested a Sip Phone with more then 60 BLF without problems?
Can someone suggest me model and brand?
Thanks, bye.
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2010 Feb 13
2
Call Pickup with 1.6.2.1 and Snom
Hi,
I've used various patches with asterisk 1.4 to have support for call
pickup and notification with good results.
Now I'm trying vanilla 1.6.2 with its official support for "dialog-info
+xml" notifications with no success. This is what i'm doing:
- Phone A has a key configured as type "extension" pointing to Phone B.
- In sip.conf I added
2013 May 25
0
Asterisk 1.8 wrong Def. Username
Hi,
We face a strange behavior with Asterisk 1.8.15 and SIP defaultuser
definition.
in sip.conf
[blabla0](natted-phone,ulaw-phone,callgroup1,snom-320)
defaultuser=tel-221
mailbox=221
callerid="My CID"
dtmfmode=auto
;defaultip=10.0.12.21
CLI sip show peer blabla0
Addr->IP : 10.0.12.21:2067
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP