Hi,
Inspired by a recent rant about one particular provider, I am getting
very curious about something I've never mastered. I'd like someone to
explain this here or at least post a link or two that can educate me
and probably countless others who have no knowledge in this area. I'm
sure there are
Im a newbie in Zaptel or Asterisk.
What schould I do know?
_
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Danny
Nicholas
Gesendet: Montag, 12. Januar 2009 18:10
An: 'Asterisk Users Mailing List - Non-Commercial
Yesterday, a low-duty production server that I maintain core-dumped. At
the time there were only around 2 calls going through it.
The strace on the screen made it look like it was caused by Dahdi.
The machine is running
asterisk-1.6.0.3
dahdi-linux-2.1.0.3
dahdi-tools-2.1.0.2
Hi Randulo,
I think this topic is probably more appropriate for asterisk-biz, as was
the aforementioned rant about one particular DID provider. But,
whatever - it is what it is.
I assume that by DID providers you are referring to origination -
that is, picking up calls on PSTN numbers and
Personnaly, i had recently encountered a global machine check exception
with
two cards (TE220p and B410) and many kernel panic with mISDN (mostly if
i tried to unload it).
Dahdi still hasn't failed me (directly)
Thomas Kenyon a écrit :
Yesterday, a low-duty production server that I maintain
Philipp Kempgen schrieb:
=== Amooma ===
* http://www.amooma.de/asterisk/sprachbausteine/#prompts-tts
These files are generated by our web-based text-to-speech engine.
Pros: If you need additional custom prompts, just go to
http://www.amooma.de/tts/ and generate them and the voice will
http://www.voip-info.org
Read all that.
On 13 Jan 2009, at 08:51, fidibus83 wrote:
I’m a newbie in Zaptel or Asterisk.
What schould I do know?
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
] Im Auftrag von Danny Nicholas
Gesendet: Montag,
On Tue, Jan 13, 2009 at 10:49 AM, Alex Balashov
abalas...@evaristesys.com wrote:
I think this topic is probably more appropriate for asterisk-biz, as was
the aforementioned rant about one particular DID provider. But,
whatever - it is what it is.
Alex, thanks for the excellent explanations!
I will add that the model of the VoIP ITSP (be it a DID provider or a
termination provider) is inherently a squeezed one.
The original niche of these providers was to simply provide VoIP in the
first place; established carriers were used to dealing with TDM (hard,
synchronous circuits using
On Tue, Jan 13, 2009 at 11:22 AM, Alex Balashov
abalas...@evaristesys.com wrote:
The original niche of these providers was to simply provide VoIP in the
first place; established carriers were used to dealing with TDM (hard,
synchronous circuits using ISDN or SS7) and had a lot invested in
On Tue, Jan 13, 2009 at 10:18:30AM +, Steve Howes wrote:
http://www.voip-info.org
Read all that.
Great RTFM answer. Let me be more specific: what does RECOVERING
mean?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
randulo wrote:
Alex, thanks for the excellent explanations! Exactly what I was hoping
for.
Good! Glad to help.
Yes, in fact I should have said origination and termination.
Ah. Termination is a somewhat different game than origination. From a
technical point of view, most of what I said
RED means the cable is unpluged or misconfigured.
you also need to configure zapata.conf in /etc/asterisk
David
2009/1/12 fidibus83 fidibu...@aol.com
Hello,
I have a problem with zaptel. I hope you can help me.
I installed and configure zaptel.
ZAPTEL.CONF
randulo wrote:
When you think about the learning curve of the average asterisk
beginner, the picture painted for them is Become your own telco! and
we all know that's not exactly accurate. For the small asterisk
install it's much more accurate to say Get an enterprise-class pbx
free (if you
On Tue, 13 Jan 2009, randulo wrote:
Hi,
Inspired by a recent rant about one particular provider, I am getting
very curious about something I've never mastered. I'd like someone to
explain this here or at least post a link or two that can educate me
and probably countless others who have no
The interesting thing about the UK is that you folks did local loop
deregulation right, unlike in the US, where it's vicious and toxic and
is mostly a story about the bewildering multitude of ways in which the
incumbents screw competitive CLECs.
No, the results of the BT bifurcation isn't
On Tue, 13 Jan 2009, Alex Balashov wrote:
The interesting thing about the UK is that you folks did local loop
deregulation right, unlike in the US, where it's vicious and toxic and
is mostly a story about the bewildering multitude of ways in which the
incumbents screw competitive CLECs.
For
I'm really liking Flowroute right now and have switched most of my
did's to them. As with many of the providers lately, they are pre-
pay... but the site is nice and payments move quickly. What I really
like about Flowroute is the no-minimum, diversity of plans, cost, and
unlimited channel
Gordon Henderson wrote:
On Tue, 13 Jan 2009, Alex Balashov wrote:
The interesting thing about the UK is that you folks did local loop
deregulation right, unlike in the US, where it's vicious and toxic and
is mostly a story about the bewildering multitude of ways in which the
incumbents
Managing financial exposure on postpaid can be hard stuff. Some give up
on it.
Also, despite the fact that it's a well-known and tiresome boondoggle -
as well as a fairly moribund loophole - at this point, there seems to be
no shortage of VoIP outfits blundering their way into the carrier
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
=== Amooma ===
* http://www.amooma.de/asterisk/sprachbausteine/#prompts-tts
These files are generated by our web-based text-to-speech engine.
Pros: If you need additional custom prompts, just go to
http://www.amooma.de/tts/ and generate them
Excellent explanation, Alex.
What's interesting is the number of caveats and mixes even in the CLEC
and ILEC world. I work with a CLEC that is also an ILEC (in certain
areas), since they encompass various areas in Georgia (and own the
state's largest contiguous network, passing through old rural
Steve Murphy m...@digium.com writes:
Which of the two would you see being useful to you?
Leg based, as far as I can see, because that looks like the only way
to bill transfers differently depending on which end did the transfer.
Possibly Simple on the Asterisk systems where we forbid
Hi All,
I have set up realtime configuration of asterisk with mysql,
and it is working fine.
Asterisk version is :1.4.21
I have a issue regarding MOH, i have created musiconhold.conf in database as
per custom configuration.
When we reload moh then it is working fine, but some times the moh get
Hi,
Thanks for your reply.
I have already used this
exten= 1002,1,Dial(SIP/1002|30|rg)
exten= 1002,2,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1002)
but my incoming call is getting hangup, it is not going to second priority.
So is there any configuration we have to do in local channel.
Hi,
If I have multiple kernel sources in /usr/src, e.g.
linux-headers-2.6.26-1-686
linux-headers-2.6.26.custom.1
how does the Zaptel Makefile(?) know which one to pick?
Is it a good approach to compile the kernel first and then compile
Zaptel manually afterwards?
Or should I rather put
Philipp Kempgen schrieb:
If I have multiple kernel sources in /usr/src, e.g.
headers
linux-headers-2.6.26-1-686
linux-headers-2.6.26.custom.1
how does the Zaptel Makefile(?) know which one to pick?
Is it a good approach to compile the kernel first and then compile
Zaptel manually
On Tue, Jan 13, 2009 at 02:31:28PM +0100, Philipp Kempgen wrote:
Hi,
If I have multiple kernel sources in /usr/src, e.g.
linux-headers-2.6.26-1-686
linux-headers-2.6.26.custom.1
how does the Zaptel Makefile(?) know which one to pick?
Is it a good approach to compile the kernel first
Max Alex schrieb:
exten= 1002,1,Dial(SIP/1002|30|rg)
exten= 1002,2,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1002)
but my incoming call is getting hangup, it is not going to second priority.
I hate ExecIf syntax but I don't see anything obvious here.
Could you try to send the
Max Alex wrote:
Hi,
Thanks for your reply.
I have already used this
exten= 1002,1,Dial(SIP/1002|30|rg)
exten= 1002,2,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1002)
This doesn't look correct (Based on looking at gotoif), try:
exten= 1002,2,ExecIf($[${DIALSTATUS} != ANSWER
Tzafrir Cohen schrieb:
On Tue, Jan 13, 2009 at 02:31:28PM +0100, Philipp Kempgen wrote:
If I have multiple kernel sources in /usr/src, e.g.
linux-headers-2.6.26-1-686
linux-headers-2.6.26.custom.1
how does the Zaptel Makefile(?) know which one to pick?
By default:
Hi
Our sip provider has two servers that sends calls to our asterisk 1.6.
When server 1 sends call everything is working, but when server 2 sends call I
get
[Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call
from '' to extension '0840303390' rejected because extension
Philipp Kempgen schrieb:
Tzafrir Cohen schrieb:
By default: /lib/modules/$KVERS/build
KVERS default to your kernel revision. e.g. `uname -r`, 2.6.26-1-686 .
This link will point to the appripriate linux-headers directory.
If you build your own kernel and install it using the kernel's
Alex Balashov abalas...@evaristesys.com writes:
There are no exceptions to this rule; numbers are assigned to carriers
and are switched and routed by carriers. Where anyone is providing
DIDs, there is a UC (Underlying Carrier) involved that is actually doing
the hauling relative to the
- Original Message -
From: Ralf Träskman
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Tuesday, January 13, 2009 4:04 PM
Subject: [asterisk-users] 404 not found from one ip-adress
Hi
Our sip provider has two servers that sends calls to our
On Tue, Jan 13, 2009 at 03:03:04PM +0100, Philipp Kempgen wrote:
Tzafrir Cohen schrieb:
On Tue, Jan 13, 2009 at 02:31:28PM +0100, Philipp Kempgen wrote:
If I have multiple kernel sources in /usr/src, e.g.
linux-headers-2.6.26-1-686
linux-headers-2.6.26.custom.1
how does the Zaptel
On Tue, Jan 13, 2009 at 3:11 PM, Benny Amorsen benny+use...@amorsen.dk wrote:
PS: Sorry if this seems like an advertisement, but I do believe that
what we are doing is a new and exciting direction for Asterisk. Like
in my other posts, I have avoided mentioning the company I work for.
Not at
Provider 2 is dropping into a new context than Provider 1. The $EXTEN is
probably coming in from P1 as XX and P2 as AXX. Check your incoming
and default sections of extensions.conf.
_
From: asterisk-users-boun...@lists.digium.com
Hi
The provider dont use register, they are running openSER I have this in my
sip.conf
[outgoing]
context=ip-only
disallow=all
allow=alaw,ulaw
canreinvite=yes
dtmfmode=rfc2833
host=sip.hub.ip-only.se
insecure=very
reinvite=yes
type=friend
[incoming]
disallow=all
allow=alaw,ulaw
context=ip-only
Hi
Its the same provider and i use dns name in sip.conf
[outgoing]
context=ip-only
disallow=all
allow=alaw,ulaw
canreinvite=yes
dtmfmode=rfc2833
host=sip.hub.ip-only.se
insecure=very
reinvite=yes
type=friend
[incoming]
disallow=all
allow=alaw,ulaw
context=ip-only
type=user
Regards
/ralf
From:
On Tue, Jan 13, 2009 at 9:59 AM, Ralf Träskman r...@adlibris.com wrote:
Hi
The provider dont use register, they are running openSER I have this in my
sip.conf
[outgoing]
context=ip-only
disallow=all
allow=alaw,ulaw
canreinvite=yes
dtmfmode=rfc2833
host=sip.hub.ip-only.se
Benny--
Thanks for the response! I've inserted comments in the following:
PS. Pardon the HTML format; my email editor splits lines at an
unadjustably
small number of columns, but in HTML, no line length limits, and better
looking examples!
On Tue, 2009-01-13 at 14:16 +0100, Benny Amorsen
Tzafrir Cohen schrieb:
On Tue, Jan 13, 2009 at 03:03:04PM +0100, Philipp Kempgen wrote:
Tzafrir Cohen schrieb:
On Tue, Jan 13, 2009 at 02:31:28PM +0100, Philipp Kempgen wrote:
Is it a good approach to compile the kernel first and then compile
Zaptel manually afterwards?
Or should I
Good Morning Everyone,
It will be great if someone can help me upgrade a Cisco 7971G-GE to SIP. If
so, please email me the detailed instructions to do the upgrade.
I will appreciate it much if you have the latest 8.4(2) firmware (file name:
cmterm-7970_7971-sip.8-4-2.cop) and email it to me or
I have concocted a system for my children's primary school where parents
can dial in and subscribe to an emergency broadcast message so that
they can be automatically contacted in case of a problem like the school
being shut because of snow etc.
I would like to provide an 0800 number service
Good Evening,
The company I work for is attempting to connect an Cisco ISDN Router to an
OpenVOX B200P BRI Card so that we can get it to dial out across an existing
ISDN PRI Line also installed in the Asterisk PBX
Everything has compiled successfully and the BRI card has been detected when
why 0800? the parents will subscribe to the system only once
you have a lot of flat fee services on-line to call land lines/mobiles in
UK.
David
2009/1/13 Julian Lyndon-Smith aster...@dotr.com
I have concocted a system for my children's primary school where parents
can dial in and subscribe
The number will also be used as a information line where a more
detailed message can be played.
The scenario is:
1) School is closed because the boiler has broken down.
2) The Head (or any authorised person) calls the service and leaves a
detailed message of the reason for closure
3) The
2009/1/13 Lee Wilson leef...@yahoo.co.uk
Good Evening,
The company I work for is attempting to connect an Cisco ISDN Router to an
OpenVOX B200P BRI Card so that we can get it to dial out across an existing
ISDN PRI Line also installed in the Asterisk PBX
Hello,
Is your setup like this ?
I recently upgraded a server to Asterisk 1.4.22 with OpenR2.
Previously I was using 1.4.18. It seems that 1.4.22 has a big bug using
chan_alsa.so for overhead paging. After rebooting the server it would
work once or twice and then I just got an error on the CLI:
[Jan 7 10:35:14]
On Tue, 13 Jan 2009, Ayman Boules (Live.COM) wrote:
It will be great if someone can help me upgrade a Cisco 7971G-GE to SIP.
If so, please email me the detailed instructions to do the upgrade.
Where's that link to http://letmegogglethatforyou.com?;
I will appreciate it much if you have the
On Tue, 13 Jan 2009, Julian Lyndon-Smith wrote:
I have concocted a system for my children's primary school where parents
can dial in and subscribe to an emergency broadcast message so that
they can be automatically contacted in case of a problem like the school
being shut because of snow etc.
I wrote a really long email, but it hinged on one thing I need
clarified...
tir, 13 01 2009 kl. 09:05 -0700, skrev Steve Murphy:
CDR1: A - B start: e1a ans: e2 end: e4 Party: B disp:
ANSW linkedID: abc9
CDR2: A start: e1 ans: e1 end: e6 Party: A disp:
ANSW
Hi Gordon,
Gordon Henderson wrote:
On Tue, 13 Jan 2009, Julian Lyndon-Smith wrote:
I have concocted a system for my children's primary school where parents
can dial in and subscribe to an emergency broadcast message so that
they can be automatically contacted in case of a problem like
On Tue, Jan 13, 2009 at 6:35 PM, Olivier wrote:
2009/1/13 Lee Wilson
Good Evening,
The company I work for is attempting to connect an Cisco ISDN Router to an
OpenVOX B200P BRI Card so that we can get it to dial out across an existing
ISDN PRI Line also installed in the Asterisk PBX
test
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asterisk-users mailing list
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I have an account with FWD and I have configured my SIP.conf with
[fwd]
type=friend
secret=password
username=901835
host=fwd.pulver.com
But when I am trying to dial out my own DID , I dont see any call landing in
asterisk.
In extension.conf (vicidial) file I have
exten = 2062036895
I also tried but cant see any call landing up in asterisk.
Btw, how to find out whether a call is landing in Asterisk or not ?
[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
Julian Lyndon-Smith wrote:
The only fly in the ointment is that my server is in 01702, but I need
a local number (01376) for political reasons
That's hardly a problem, (If the call is to be presented using VoIP)
more or less any provider will give you a local number from another area.
I
Thomas Kenyon wrote:
Julian Lyndon-Smith wrote:
The only fly in the ointment is that my server is in 01702, but I
need a local number (01376) for political reasons
That's hardly a problem, (If the call is to be presented using VoIP)
more or less any provider will give you a local number
ngrep port 5060
or tcpdum port 5060
By default asterisk runs on port 5060, that way you can see if your getting
the signal or not.
Jai Rangi
Buy SIP DID www.didforsale.com
free Trial now purchase required
On Tue, Jan 13, 2009 at 1:13 PM, David @ULC ucoms2...@gmail.com wrote:
I also tried but
[r...@vicidialnow ~]# ngrep port 5060
-bash: ngrep: command not found
[r...@vicidialnow ~]# tcpdum port 5060
-bash: tcpdum: command not found
[r...@vicidialnow ~]#
Also, is my SIP configuration is correct ?
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Sorry for the typo,
tcpdump port 5060
ngrep you can download the rpm (google) easy to install
http://rpm.pbone.net/index.php3/stat/4/idpl/1127130/com/ngrep-1.38-1.i386.rpm.html
rpm -ivh
[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
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David @ULC schrieb:
[r...@vicidialnow ~]# ngrep port 5060
-bash: ngrep: command not found
aptitude install ngrep
[r...@vicidialnow ~]# tcpdum port 5060
-bash: tcpdum: command not found
aptitude install tcpdump
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany -
Anyone using FWD with Asterisk ?
On Wed, Jan 14, 2009 at 2:40 AM, David @ULC ucoms2...@gmail.com wrote:
I have an account with FWD and I have configured my SIP.conf with
[fwd]
type=friend
secret=password
username=901835
host=fwd.pulver.com
But when I am trying to dial out my own DID ,
I'm looking for some info on the Asterisk Appliance.
I understand it has a gui, but can I still do all the dialplan config
that I'm used of doing by hand outside of the gui? If I really wanted
to, could I even ignore that the device has a gui and do all my config
in the files? I guess I'm just
I tried this
http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html
But I am NOT getting call in asterisk.
SIP.conf file :
_
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
externhost=59.160.44.21
localnet=192.168.0.2/255.255.255.0
; register SIP
SIP wrote:
What's interesting is the number of caveats and mixes even in the CLEC
and ILEC world. I work with a CLEC that is also an ILEC (in certain
areas), since they encompass various areas in Georgia (and own the
state's largest contiguous network, passing through old rural ILEC lines
Benny Amorsen wrote:
Alex Balashov abalas...@evaristesys.com writes:
There are no exceptions to this rule; numbers are assigned to carriers
and are switched and routed by carriers. Where anyone is providing
DIDs, there is a UC (Underlying Carrier) involved that is actually doing
the
While I don't know the OpenVOX B200P specifics, some interface cards
need you to change physical jumpers in order to acheive NT vs TE, mode.
Could that be the case ?
--
exvito
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Alex Balashov wrote:
SIP wrote:
What's interesting is the number of caveats and mixes even in the CLEC
and ILEC world. I work with a CLEC that is also an ILEC (in certain
areas), since they encompass various areas in Georgia (and own the
state's largest contiguous network, passing
On Tue, 2009-01-13 at 21:09 +0100, Benny Amorsen wrote:
I wrote a really long email, but it hinged on one thing I need
clarified...
tir, 13 01 2009 kl. 09:05 -0700, skrev Steve Murphy:
CDR1: A - B start: e1a ans: e2 end: e4 Party: B disp:
ANSW linkedID: abc9
If I use below code in my sip.conf ,
[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
how will call understand that where I have to land as we DO NOT provide our
IP in fwd
When I logged in to my IPKall website ,
I see SIP Proxy: as fwd.pulver.com Do I need to change it to my PUBLIC or
STATIC IP ?
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To UNSUBSCRIBE or
SIP wrote:
I don't know that the price of UNE DS1s and DS3s is really all that
exceptional. Sure, it seems impressive that you can get a T1 in LATA
438 for some odd $44, but once you factor in the costs of
interconnection, CO colocation, EELs and interoffice mileage if not
colocated in
Alex,
I must say wow, great explanation. It was a wonderful reading.
Best,
-Jai
On Tue, Jan 13, 2009 at 1:49 AM, Alex Balashov abalas...@evaristesys.comwrote:
Hi Randulo,
I think this topic is probably more appropriate for asterisk-biz, as was
the aforementioned rant about one particular
Thanks Your tip got my on the right track
Regards
/ralf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristian
Kielhofner
Sent: den 13 januari 2009 16:32
To: Asterisk Users Mailing List - Non-Commercial
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