Displaying 20 results from an estimated 10000 matches similar to: "Searching on how to keep local calls... local"
2010 Apr 13
2
SNOM M9 base station A to base station B
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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<small><font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
I have a question concerning SNOM M9 base station.<br>
<br>
If my customer places a SNOM M9 base
2009 Apr 13
10
Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
These are my settings :
sip.conf :
[root at asterisk asterisk]# cat
2010 Mar 01
2
Is answer() necessary ?
Hello list,
is it necessary to properly answer() an incoming call ?
I don't want to answer a call because the caller has to pay even if the
attached SIP-phones do not answer the phone call. Because I answer() the
incoming call, the caller has to pay for 60 seconds of 'ringtone'.
On the other hand, sometimes an incoming call is send to a macro where
the caller is given the
2009 Sep 07
2
features.conf : feature map ==> getting feature to work
Hi there,
I need some help with a 'custom' feature.
I have following feature defined in features.conf :
[applicationmap]
opnemencallee =>
#3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
In my dialplan :
[from-HostAst]
exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee)
exten => s,n,Dial(SIP/grandstream,30)
I want the callee to be able to press #3 to be able
2010 Mar 01
1
rtcachefriends & qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify
is incompatible with dynamic uncached realtime. Please either turn
rtcachefriends on or turn qualify off on peer 'gerrie'
Am I correct that when I turn on rtcachefriends in sip.conf,
database-changes in my MySQL-DB will not be reflected untill a reload ??
Am I correct that when I turn off qualify in my realtime
2009 May 12
2
Hangup()-command does not hang up the line
When I call my Asterisk-server from my cell phone on one of the
PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card,
and in the dialplan the end of a context is reached and Asterisk needs
to execute the Hangup()-command, I notice the following :
- Asterisk tells me that the conversation was hung up (the log files
tell me the command was executed)
- On my cell phone I hear
2009 May 08
2
Not receiving voicemail message in mailbox
It should be as simple as editing voicemail.conf :
; Voicemail Configuration
;
[general]
; Formats for writing Voicemail. Note that when using IMAP storage for
; voicemail, only the first format specified will be used.
format=wav49|wav|gsm
; Who the e-mail notification should appear to come from
serveremail=asterisk-voicemail
; Should the email contain the voicemail as an attachment
attach=yes
;
2009 May 19
5
OT: SIP hardphone with multi-color BLF
Hi,
Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors
(or more) ?
This could be very useful to support extended presence, for instance.
Regards
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2009 Oct 08
2
How to keep difference between 2 SIP-accounts/trunks from same server ??
Hey list,
I have a problem when I host 2 SIP-accounts on the same Asterisk-server.
Asterisk picks out the SIP-account on alphabetic order A --> Z.
In my sip.conf :
register => user1:passwd1 at server/user1
register => user2:passwd2 at server/user2
[YOCAN-3starsnet]
type=peer
host=server
username=user1
secret=passwd1
fromuser=user1
accountcode=user1_in
[ITCENTER-3starsnet]
type=peer
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
2009 Oct 14
2
FXS to SIP gateway
Hello list !
I don't have the money to test out all the products and reading the
manuals is not always that enlightening...
Does someone here know a good gateway-product that lets analogue
telephones communicate with an Asterisk-server.
I have found the Grandstream GXW-400x to be able to add SIP-accounts to
analogue telephone devices that are connected to the FXS-ports. Moreover
this
2009 Aug 29
1
GoToIfTime : how to define sep 25th till oct 10th ?
Hi list,
quick question :
With GoToIfTime, how to define a period of holiday that starts at the
end of the month and ends at the beginning of the next month ??
Like September 25th till October 10th when incoming calls need to go to
the voicemail...
Greetingz,
Jonas.
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2009 Aug 18
2
You do not appear to have the sources for the 2.6.20-prep kernel installed
I want to install Dahdi and Dahdi-tools on a CentOS 5.3 Xen host and I
receive the following error :
"You do not appear to have the sources for the 2.6.20-prep kernel
installed."
I have installed :
- kernel-headers-2.6.18-128.4.1.el5.x86_64
- kernel-devel-2.6.18-128.4.1.el5.x86_64
- kernel-xen-devel-2.6.18-128.4.1.el5.x86_64
bash-3.2# uname -r
2.6.20-prep
bash-3.2# ls -l
2009 Aug 30
2
MySQL syntax error : I really don't see where...
Hi list,
I'm stuck for the moment @ the following :
My Query (in a macro) :
exten => s,n,MYSQL(Query resultid ${connid} SELECT\ vakantie_set\
vakantie_data1\ vakantie_data2\ FROM\ AstDB\ where\
SIPACCOUNT="${ARG1}")
Asterisk CLI :
[Aug 30 14:07:42] -- Executing [s at macro-vakantie:2]
MYSQL("IAX2/zoiper-9238", "Query resultid 1 SELECT vakantie_set
2009 Apr 27
1
Packet2packet bridging while in sip.conf canreinvite=no
I have put canreinvite=no for all my internal SIP-clients in sip.conf
because I want Asterisk to be in the middle of the RTP-stream so he can
provide MusiconHold and so...
Now, what the Asterisk CLI tells me when I make a call from my one
internal SIP-phone to another internal SIP-phone is :
Verbosity is at least 25
== Spawn extension (intern, 51, 1) exited non-zero on
2014 Jan 08
2
Call duration limit ? Calls end after 15 minutes...
Hello,
I see the strange behaviour that outgoing calls end after 15 minutes.
I didn't knew there is some kind of call duration limit that can be set ?
Is there ?
Using Asterisk 1.8.12.2
Kind regards,
Jonas.
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2009 May 04
1
Can someone help me with my IAX-registration
Thanks for the feedback !
I know the IP-address of my Asterisk-server.
The WAN-interface of my Asterisk-box is set manually (ifcfg-eth1).
I have port 4569 forwarded on my NAT/firewall.
Strangely I have the same 'notice' when being attached directly to the internet (so no firewall in between).
And set my WAN-interface to the IP I get from my ISP or even when obtained by DHCP.
Doesn't
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
2010 May 31
6
Voicemail : mail attachment to multiple mail-addresses
Hello list,
google returns a discussion on the dev-list when I search for how to
mail a voicemail to multiple mail addresses.
Is there yet a seperator that actually works to define multiple mail
addresses ?
Kind regards,
Jonas.
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2011 Mar 09
6
SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten => s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
/INVITE sip:0473 at sip.domain.be SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97
From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2
To: <sip:0473 at sip.domain.be>