Displaying 20 results from an estimated 3000 matches similar to: "Incoming SIP callerid"
2014 Oct 10
1
howto cancel simultaneous calls - dial(sip/phone1&sip/phone2)
hi.
i have dialplan with 2 simultaneous calls - dial(sip/phone1&sip/phone2).
when i cancel call on phone1 (push "reject" button), the call is still
ringing on phone2
can i cancel call on both phones from one place(one phone)?
thanks
--
---------------------------------------
Marek Cervenka
=======================================
2004 May 09
2
Help with initial setup
Hi,
I've have followed through the help docs in trying to get an initial setup
going with two phones and the asterisk server. Firstly, all I'm trying to
do is get the two phones actually talking to one another VIA asterisk..
I've added this to sip.conf:
[phone1]
type=friend
host=dynamic
defaultip=192.168.1.106
;username=blah
;secret=blah
dtmfmode=rfc2833 ; Choices are inband,
2005 Aug 27
0
Newbie :SIP ETXTN to SIP EXTN calls
I am new to asterisk and need to dig up some info on how to set it all
up. It looks a bit daunting especially all the options available in the
.conf files.
I have 2 SIP phones, GXP2000 and a budgettone 100.
phone1 - 192.168.0.160/24 extension 1000
phone2 - 192.168.0.161/24 extension 1001
Server - 192.168.0.57
I get the following all the time, but can make calls between the 2
extensions,
2018 Dec 07
4
how to use a database
On 12/07/2018 03:36 PM, Administrator TOOTAI wrote:
> Le 07/12/2018 à 14:32, hw a écrit :
>
> [...]
>>
>> Queues seem to be the only way to have several phones ring at once, or
>> are there other ways?
>
> Dial(SIP/Phone1&SIP/Phone2&...&SIP/Phonex,,)
>
Good to know, thanks!
What are the entries needed in the queue_members table when using
2005 Feb 01
0
TBM400 no callerid on incoming calls?
I have installed my TDM11B according to the docs at the Digium page but I
do not get incoming caller id. My telco confirmed that callerid should be
passed but I do not see it coming in. I am in The Netherlands with a KPN
line. The number is not even visible in console mode on * running stable
1.0.5. Ideas anyone?
-- Starting simple switch on 'Zap/4-1'
Feb 1 19:19:40 NOTICE[16582]:
2004 Dec 01
0
extension and PSTN connection
I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B
(603, 604). I have two lines on the TDM22B.
I cannot figure out some of the problems:
1. 601 dials via ZAP/3-1 to local phone number at PSTN:
ringing
pickup on PSTN (empty)
still ringing in the phone set 601
2. call from PSTN back:
601 picks up ... everything works !!!
No caller id shows up
3. For testing I have only one
2005 Mar 01
2
Park Craches asterisk
I've just installed asterisk on a Debian Linux (apt-get it)
And i have placed two sip phones in sip.conf and i'm testing parking
with them
I have phone1-SIP/1000 and phone2-SIP/1007
The following happens if i park from calling party and everything is OK
1. Pickup Phone2 and call to Phone1
2. Talk
3. Phone2 dials #700 and parks the call (it is placed in 701)
4. Phone2 is hangup
5. Pickup
2006 Nov 04
1
Pass through
Hi!
I want to tell asterisk to simply pass-through any codecs that my phones
support. I have to use codecs that are not popular and implemented by a
third-party, asterisk has nothing to do with them.
I've made a test with g722 (that asterisk doesn't support), i've set all my
two snom 300 phones to support only g722 and asterisk declined the sip
invitation. That is bad for me. Is it
2006 Nov 03
0
Pass-through any codecs
Hi!
Maybe you can help me.
I want to tell asterisk to simply pass-through any codecs that my phones
support. I have to use codecs that are not popular and implemented by a
third-party, asterisk has nothing to do with them.
I've made a test with g722, i've set all my two snom 300 phones to support
only g722 and asterisk declined the sip invitation. That is bad for me. Is
it possible that
2003 Nov 06
3
Grandstream problem
Hi,
I installed Asterisk an all works fine exept for Grandstream.
When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok
When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so)
It's the same when I
2011 Jan 21
0
Queues with ringinuse=yes
I'm setting up a queue for two independent operator phones that are capable of answering multiple calls at once. It's currently working with the following settings and Asterisk 1.4:
queues.conf:
[telefonistas]
strategy=roundrobin
;strategy=leastrecent
music=default
timeout=60
retry=0
maxlen=0
wrapuptime=0
ringinuse=yes
autofill=yes
joinempty=yes
member => SIP/8899
member =>
2003 Oct 29
1
Host unspecified ??
Dear,
When I start asterisk -vvvvvvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field.
Name = phone1 and phone2
Host=unspecified
mask 255.255.255.255
port = 0
status = unmonitored
I can ping the two phone's and get a reply (also from the laptop)
phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14)
2007 Jun 07
1
DUNDi and reinvites...
I don't know if this is possible, and I can't quite get my head around
how to do it...
If I am using DUNDi for redundancy in a cluster, when Phone1 makes a
call to Phone2, both Asterisk A and B will be in the RTP stream:
+---+ +---+
| A |-----| B |
/+---+ +---+\
/ \
Phone1 Phone2
Is there a way configure re-invites
2008 Oct 31
1
Monitor group calls (recording calls)
Hello there,
I appreciate any help about this problem that I can't figure out...
I need to record all my calls: this is pretty easy using Monitor() before
the Dial().
eg:
exten =>
425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb)
exten => 425,n,Dial(${PHONE1},10)
Now, I want to create a call group: I mean, I want a number (eg 800) that
makes
2005 Mar 04
2
Broadvoice + incoming call works only for ~2 minutes
Hi, all.
The asterisk setup is working fine, receiving calls via broadvoice "initially". ?
When call comes in via broadvoice number, asterisk picks it up and routes
correctly, as long as the call came in within ~2 min from the previous one.
In other words, as long as a call comes in within ~2 min since the previous one,
asterisk will answer the call. However, if the call comes in
2005 Aug 02
0
strange asterisk issue
I have the following asterisk configuration (sip.conf) :
[General]
externip=82.79.81.3
localnet=192.168.10.0
localmask=255.255.255.0
[Phone1]
type=friend
host=dynamic
nat=yes
qualify=yes
context=sip
callerid="Phone1" <1>
disallow=all
allow=gsm
[Phone2]
type=friend
host=dynamic
qualify=yes
context=sip
callerid="Phone2" <2>
disallow=all
allow=gsm
[Phone3]
2004 Dec 19
1
sip phones in different private networks have one way audio
Hello
I have one phone (phone1) in one network, the other (phone2) in public
network. both can call the other side; phone1 can be heard by phone2, phone2
can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER.
Is NAT still necessary to be set on both phones?
Thank you!
steven
2005 Feb 28
1
call from two sip phones registered in different asterisk server
Hi all
I have registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones.
i configured the extensions.conf file in both the server.
the extensions.conf file on server 192.168.0.9 is
exten=>301,1,Dial(SIP/301@192.168.0.6,20,tr)
exten=>401,1,Dial(SIP/phone1,20,tr)
301 is the extension number for phone 2 in asterisk server
2011 Dec 29
0
Help_In Voicemail , vedio play but voice is not here out.
Hi all,
I am using to Xlite to save video voice mail.
when i retreive it, then only video show , no voice is here out.
Plz tell me where ,i am wrong , and how i can able to see video plus here audio in voice mail box.
I did following configuration
In Sip.conf
videosupport=yes
[phone1]
type=friend
host=dynamic
context= employees
mailbox=101 at default
2008 Dec 05
0
Bug in schema.rb generation during db:migrate
I am thinking that I have found a bug in Rails migrations. My app is
using UUIDtools to generate guids for primary keys. To do this I
pass :id=>false and then create my own id column as shown below. Next
I leverage "execute" to create an index. It seems to work fine. The
table in MySql is perfect. However the ID column and primary key on
the ID column are not in the schema.rb file
I