Displaying 20 results from an estimated 20000 matches similar to: "Set origin CallerID when forwarding calls to mobile phone"
2007 Jul 17
0
help with sip configuration for sipgate.de on asterisk 1.4
hi there,
i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main
computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de,
there i can receive call and make them, i can hear the other end but they can not hear me, this is
only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help
2008 Jan 02
2
Trixbox and mail2fax
Hi there,
is there any howto how do i configure a asterisk/trixbox for mail2fax?
The fax must be send over sipgate or other SIP peers. (i dont have
any "normal" telephones connected).
What i wanne do is somethink like this:
Subject: +49691234567
Attache: *.pdf
The attched pdf have to be send ;)
--
Mit freundlichen Gr??en
Daniel
mailto:daniel at listmails.de
2010 Nov 19
1
callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
Hi all,
I've got 4 actors on my stage:
Alice calling from outside
Bob transferring incoming calls to Charlie
Charlie who has a mobile phone
My PBX which is connected to my ISDN line.
I want Charlie to see Alice's Callerid after Bob has transferred the
call as if Charlie is receiving the call from Alice, transparently.
Tried to set the callerid but Charlie sees my telco line number,
2010 Feb 18
0
ISDN phone not ringing. ISDN PBX not answering?!
Hi,
I've set up an Asterisk as voip gatway:
VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx.
Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset.
I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode.
The msn is set at the dect phone/base station
2006 Mar 13
0
How to forward inbound sipgate calls to different users in my entreprise, (
Hi everybody, I would like to know if it is possible to add a suffixe
number to my official sipgate number?
For example my official sipgate number is 123456
My local users numbers are 0001 0002
Could it be possible to compose from internet 1234560001 or 1234560002 ,
then the message could be forwarded to user 0001 or 0002.
Actually I can only forward my sipgate number to only one user.
So I am
2005 Aug 06
0
SIP rejecting calls?
Hi,
I have researched more into the problem of my Asterisk set-up not answering
calls.
The following error was shown on the CLI, can anyone explain what the
problem causing Asterisk to not answer the SIP calls be?
Information: I have an Asterisk box on a home LAN, behind a D-Link
router/firewall connected to a cable modem. The 82.x.x.x is the IP for my
cable modem. 192.168.0.101 is my
2020 Aug 10
0
With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
Hi Dan,
i did it wrong, sorry:
curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST "
http://localhost:8088/ari/channels/newChannelId"
<http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world>
--data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)":
"Alice" ,
2013 Sep 18
2
sipgate outgoing calls
Hello
i am trying to setup sipgate gateway
i can get incoming calls fine, but when i dial in and then try to dial
out i get this in asterisk command line
-- Called 01179248615 at sipgate
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on INVITE to
'"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1'
--
2004 Dec 23
2
Incoming calls from Sipgate go through the wrong peer
Hi,
I have a few accounts with sipgate.co.uk to get some different DiD
numbers. However, when an incoming call comes in, it seems to pick the
wrong peer from sip.conf which sends the call into the wrong context and
it fails because there is no extension in that context to match the
register.
Using the config's below, if I dial the DiD on account 2222222, it works
fine - picks peer 2222222
2020 Jun 18
3
CallerID fail with Voicetrading operator
Hello,
does some people here use https://voicetrading.com which is a Dellmont
service from Netherlands. At the high begining they were also known as
Finarea (CH and DE mixed Co)
Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our
callerID is no more seen by them. We use
Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or
equal to CALLERID(num). We tried
2005 Aug 20
0
Help needed receiving incoming calls.
2020 Aug 10
0
With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
Hi Jöran,
Would it be possible to see an example using curl of how you are passing the PAI Header through ARI create?
Dan
From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jöran Vinzens
Sent: Friday, August 7, 2020 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] With
2009 Nov 28
2
can't hear anything at incoming calls
Hi out there,
I think i've everything set up properly, outbound calls are working fine, but
at incoming calls I can't hear anything, but the other one is able to hear me
perfectly.
I'm using an asterisk 1.6.1.10 in my internal network in a NAT, connected to
my sip-provider using a trunk.
Firewall settings on the router are:
forward UDP port 5060,5004,10000-20000 to asterisk server
2004 May 11
2
Sipgate to regular phones
I could call a regular phone through sipgate.
Now i can not:
Failed to authenticate on INVITE to '"xyz"
<sip:4xxxxxx@sipgate.net>;tag=as4ddd4a6f'
A call from outside to my sip-phone through sipgate is OK.
Can anyone verify ?
Is it a sipgate problem ?
greetings nicolas
2008 Nov 17
1
Deny FOP originated calls
Hi,
I just want to deny FOP originated calls in TRIXBOX. All remaining
operations (hanging up, transferring, etc) should go. Where is that
option in TRIXBOX (already googled, nothing, checked config files but
cant find that option)?
Thanks a lot.
--
Rodolfo Alcazar
Responsable red y datos
Deutsche Gesellschaft f?r
Technische Zusammenarbeit (GTZ) GmbH
Programa de Apoyo a la Gesti?n P?blica
2009 Aug 04
5
Anyone actively using RLT for mobile phone forwarding?
Hello,
We currently have a scenario where a large percentage of inbound
calls on a telco PRI are intended for professors who aren't currently
in their offices/at their desks.
My question is, is anyone actively using the Asterisk "RLT" (Release
Link Trunking) feature to bounce these sorts of calls back to the
telco? The idea being to forward the call to their mobile phone without
2020 Aug 10
2
With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
Hi Dan,
i would do something like this (it is not a copy of what we are doing but
an example of how i would do it)
Important here is the "--data" and "-H" Option as well as the "variables"
section within the Body. I added the callerid function here as well as it
is the sample in the asterisk wiki.
curl -v -H "Content-Type: application/json" -u
2005 Mar 20
0
rejected calls
Hi,
Using a couple of sip phones and using asterisk to connect them to a
single sipgate.de account.
if I call a mobile I have no problem makeing conversions. If the mobile
rejects the call (by pressing hangup while it rings), something strange
happens:
the following is seen in the logfile, everytime a rejected mobile call
happens:
-----------------
Mar 20 22:52:29 WARNING[4682]: Forbidden
2008 Nov 29
0
asterisk-users Digest, Vol 52, Issue 81
I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work.
-----Original Message-----
From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com>
To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com>
Sent: 11/29/2008 1:13 PM
Subject: asterisk-users Digest, Vol 52, Issue 81
2004 Dec 01
0
sipgate x asterisk: problems to receive PSTN calls?
I noticed that I'm no longer able to receive calls from PSTN to my
SipGate DID number.
I changed the sip.conf and extension.conf as per
http://www.voip-info.org/tiki-index.php?page=Sipgate but the problem
remains...
However, I can receive calls from another sipgate user. The problem is
only affecting calls from the PSTN (DID). Anyone with the same problem?