I am using Asterisk and
IPKall.http://lists.digium.com/pipermail/asterisk-users/2008-January/203607.html
I tried exactly below mentioned configuration
.http://lists.digium.com/pipermail/asterisk-users/2008-January/203607.html
On Thursday, January 15, 2009, David fire wrote:
hey it is preatty easy now i understand the problem
is simple
hangup in new location
dial steal code for asterisk is just an extension and it should start an
AGI
the system search for the call in the same group
bridge the
Why don't you simply Dial() the call to a separate box keeping Asterisk out
of the audio path?
l.
2009/1/16 Paul bulkm...@monafamily.com
Can anyone tell me how I can completely move an established call off of
one Asterisk server to another?
In our case we have a server with our IVR.
Emmanuel Pascal Bruno wrote:
Has anyone been able to configure portech's mv-378 gateway with asterisk?
I did the configuration as per the manual but it does not work.
My server sees the portech gateway, but when the gateway is trying to
register to my server it fails. It says peer is
Johansson Olle E schrieb:
15 jan 2009 kl. 12.42 skrev Klaus Darilion:
Johansson Olle E schrieb:
14 jan 2009 kl. 18.57 skrev Philipp Kempgen:
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside
the
Are they identical to the ISDN hangup causes?
http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php
klaus
Johansson Olle E schrieb:
15 jan 2009 kl. 13.02 skrev John covici:
That is very nice, but where are the HANGUPCAUSE values documented?
That's the issue...
I do not know cordiip thus I do not know how these 3 different accounts
are signaled to you, but some tips:
A SIP peer is always identified by host:port - thus there is at peer
level no way to differ them. But in the register command you specify the
contact to be called, e.g. 1646H25.
Hi Francesco,
You were correct. I pulled the cable out before everyone got in this morning
and it was a cross over. I've now connected a proper straight-through ISDN
cable (don't know what the Nortel was using before) and L1 is now up on
Asterisk:
BEGIN STACK_LIST:
* Port 1 Type NT Prot.
do you program in any language? if yes just read the chapters about
agi in the asterisk book you can find it in support section in
www.asterisk.org
if you can't program send me an email I think this agi will be easy.
I will program it for you (if you can't program)
2009/1/16, Geoff Lane
How to add SipAddHeader in outgoing call file.
I am implementing a Callback scenario, in which a user makes a call to
Local Access Number. The system have to callback to the user. During
callback a call file is generated. All I want, is to add
SipAddHeader(pchargingvector,val) in outgoing Invite.
On Friday, January 16, 2009, ddf...@gmail.com wrote:
do you program in any language? if yes just read the chapters about
agi in the asterisk book you can find it in support section in
www.asterisk.org
I'm a reasonable PHP and VBScript programmer and have dabbled since
the 1980s in a wide
Use a Local/ channel in the Originate command, which can punt the
outbound leg through dial plan logic that can call SipAddHeader() and
tack on the header.
Mian M Asif wrote:
How to add SipAddHeader in outgoing call file.
I am implementing a Callback scenario, in which a user makes a call to
2009/1/16 Geoff Lane ge...@gjctech.co.uk
On Friday, January 16, 2009, ddf...@gmail.com wrote:
do you program in any language? if yes just read the chapters about
agi in the asterisk book you can find it in support section in
www.asterisk.org
I'm a reasonable PHP and VBScript programmer
Laurent a écrit :
Those terms would be ISDN-related. VN4 is Version Number 4, and
ETSI is the European standards-adopting organization for telecoms.
So you might want to check for E1 support (ISDN in Europe,
basically) if you want to connect a PRI-capable equipment - I
assume that's what you
How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again
in steps please.
From: torinti...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 16 Jan 2009 03:25:33 +0200
Subject: [asterisk-users] Asterisk Upgrade
I was Using freepbx-2.1.1, Asterisk 1.2.29
Hello All,I'm very new in asterisk.Please help - how I can write conf files
(or some example) for to delete one ext. and to add another, it means for
example:
I need to call from one asterisk to another by trunk to trunk and my dialing
(for ex.) 100#...@1.2.1.2
when the the trunk of first asterisk
1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk
2. Install 1.2.29.
Torintino T wrote:
How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully
again in steps please.
From:
Hello All,I'm very new in asterisk.Please help - how I can write conf files
(or some example) for to delete one ext. and to add another, it means for
example:
I need to call from one asterisk to another by trunk to trunk and my dialing
(for ex.) 100#...@1.2.1.2
when the the trunk of first asterisk
Benoit a écrit :
Laurent a écrit :
Those terms would be ISDN-related. VN4 is Version Number 4, and
ETSI is the European standards-adopting organization for telecoms.
So you might want to check for E1 support (ISDN in Europe,
basically) if you want to connect a PRI-capable equipment - I
Hi,
I am using Asterisk 1.4.12.1 version.
Scenario for the call is as below:
UAC Asterisk
UAC Transfered to
|--Invite-|
| |
|
|---Invite| |
|
Thanks.
Date: Fri, 16 Jan 2009 07:15:29 -0500
From: abalas...@evaristesys.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Upgrade
1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk
2. Install 1.2.29.
Torintino T wrote:
How can i erase
On Fri, 16 Jan 2009, Alex Balashov wrote:
1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk
I'd suggest not removing /etc/asterisk if that's the only source of your
config files... If you (re)generate them from elsewhere, it's probably OK.
and the important one, I'd have thought
Thanks to you.
Date: Fri, 16 Jan 2009 13:24:16 +
From: gordon+aster...@drogon.net
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Upgrade
On Fri, 16 Jan 2009, Alex Balashov wrote:
1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk
I'd
Thanks for your reply!
Can you tell me what you have in your Portech configuration settings (Mobile
to Lan Settings; Sip Proxy settings etc...) My sip.conf file is pretty
similar to yours but still cant register.
On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini marcota...@libero.itwrote:
Dear Alex Balashov and All others,
can anyone give me the example how i can add local/channel with out
call file which used for Callback, Below is my dialplan for Callback.
Need to know where i can add SipAddHeader() in below dialplan. I want
to add in call leg one.
exten = _X.,1,wait(1)
exten =
David @ULC schrieb:
Jan 15 17:18:43 ERROR[2636]: chan_sip.c:11355 handle_request: Missing Cseq.
Dropping this SIP message, it's incomplete.
This is the important part.
One of your phones or FWD or IPCall does not speak SIP properly.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock /
Klaus Darilion schrieb:
One of the problems with hangupcause is, that is might get changed from
one Asterisk to another - e.g. Hangup(3) generates a SIP 404 response
which gets translated to hangupcause 1. So, a mechanism to signal
Asterisk hangupcauses from one Asterisk to another Asterisk
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
18 (Ring Begin)...
[Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
One of the problems with hangupcause is, that is might get changed from
one Asterisk to another - e.g. Hangup(3) generates a SIP 404 response
which gets translated to hangupcause 1. So, a mechanism to signal
Asterisk hangupcauses from one
Klaus Darilion schrieb:
Are they identical to the ISDN hangup causes?
http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php
Yes. What you pass to Hangup() are Q.931 ISDN cause codes.
See causes.h and hangup_cause2sip() in chan_sip.c for a list.
Philipp Kempgen
--
AMOOCON
Can anyone who has used both comment on the pros and cons ? Need to buy
about 30 of these, for a small company with limited IT support.
Julian
__
This email has been scanned by the MessageLabs Email Security System.
For more
Pascal Bruno wrote:
Thanks for your reply!
Can you tell me what you have in your Portech configuration settings
(Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file
is pretty similar to yours but still cant register.
On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini
Hi,
I've never used Snom phones, but have used the Grandstreams.
I think you will find that they just feel 'cheap.' We had a half dozen
of them, and the functionality is there, and they work great. But they
just feel rough and cheap when using them.
If you are planning on using different
On Fri, 16 Jan 2009, Julian Lyndon-Smith wrote:
Can anyone who has used both comment on the pros and cons ? Need to buy
about 30 of these, for a small company with limited IT support.
You get more phone for your money with the Grandstream, but ...
You'll hear a lot of people here who've had
Hi all,
Suposing that 2 SIP phone register at a remote (internet) asterisk, what
is the best way, if any, to make the RTP traffic go phone to phone, whithout
using the internet conection (asterisk)?
Thanks,
Gabriel
___
-- Bandwidth and Colocation
canreinvite=yes.
Gabriel Ortiz Lour wrote:
Hi all,
Suposing that 2 SIP phone register at a remote (internet) asterisk,
what is the best way, if any, to make the RTP traffic go phone to phone,
whithout using the internet conection (asterisk)?
Thanks,
Gabriel
On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote:
Hi all,
Suposing that 2 SIP phone register at a remote (internet)
asterisk, what is the best way, if any, to make the RTP traffic go
phone to phone, whithout using the internet conection (asterisk)?
Allow reinvite? Assuming both
They will be in the same LAN, probably behind NAT.
Being in the same LAN helps something?
2009/1/16 Jerry Jones jjo...@danrj.com
On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote:
Hi all,
Suposing that 2 SIP phone register at a remote (internet)
asterisk, what is the best way,
Gabriel Ortiz Lour wrote:
Hi all,
Suposing that 2 SIP phone register at a remote (internet) asterisk,
what is the best way, if any, to make the RTP traffic go phone to phone,
whithout using the internet conection (asterisk)?
Thanks,
Gabriel
By default, Asterisk will attempt to
Thank you!, I will try that in a few hours and let you know what happens.
On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini marcota...@libero.itwrote:
Pascal Bruno wrote:
Thanks for your reply!
Can you tell me what you have in your Portech configuration settings
(Mobile to Lan
On Jan 16, 2009, at 7:52 AM, Julian Lyndon-Smith wrote:
Can anyone who has used both comment on the pros and cons ? Need to
buy
about 30 of these, for a small company with limited IT support.
We recently deployed 85 phones to our office. We tested the
Grandstream GXP2000, GXP2020,
Hi,
A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN
trought another E1. When the legacy user dial to the PSTN the call pass
trought Asterisk.
All works OK, the only problem is the delay on the Asterisk server when it
receives the digits from the 1st E1 link. It
I do have it functioning with Dial(). I was looking for a way to
completely move the call from the first box though. When using Dial() media
moves, but the call is still tied to the first box. In looking at captures
when the call is ended, the first box invites out to the ITSP again, then
I have this call:
SIP/protel-525512047 default 90445528885371 1 Ringing
AppDial (Outgoing Line) 90445528885371 264:24:2
(None)
I cannot use the soft hangup commando from the CLI because I do not
know the whole SIP channel string. What other
try to know the whole string ?
core show channels
2009/1/16 Carlos Chavez cur...@telecomabmex.com
I have this call:
SIP/protel-525512047 default 90445528885371 1 Ringing
AppDial (Outgoing Line) 90445528885371 264:24:2
(None)
I cannot use the
2009/1/17 Carlos Chavez cur...@telecomabmex.com
I have this call:
SIP/protel-525512047 default 90445528885371 1 Ringing
AppDial (Outgoing Line) 90445528885371 264:24:2
(None)
I cannot use the soft hangup commando from the CLI because I do not
If youre using the GUI it will hang it up. Otherwise sip reload might do
it.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grygoriy
Dobrovolskyy
Sent: Friday, January 16, 2009 12:00 PM
To: Asterisk Users Mailing List
Hello all,
I have one Asterisk 1.4.21 system connected to a North American POTS
line. Normally hangup detection works fine, and Asterisk hangs up
properly if you are talking to a caller and they hang up; but
occasionally a call comes in (typically from a US telemarketer) where
the caller hangs
I guess you already tried this?
http://www.voip-info.org/wiki-Asterisk+cmd+Transfer
Thanks
l.
2009/1/16 Paul bulkm...@monafamily.com
I do have it functioning with Dial(). I was looking for a way to
completely move the call from the first box though. When using Dial() media
moves, but
exten = _X,1,Dial(DAHDI/g1/${EXTEN})
or
exten = _9,1,Dial(DAHDI/g1/${EXTEN:1})
the first whan put X to match the amount of digits.
the second one dial if you put a nine before. and many X as digits
David
2009/1/16 Gabriel Ortiz Lour ortiz.ad...@gmail.com
Hi,
A have an
Hello,
When I bridge an incoming and outgoing call (attempting to simulate
call-forwarding) I'm only getting one CDR -- that of the outgoing call.
A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone
on PSTN) and bridges the call.
The only CDR created is from B to C. I
Here's also an example snip from the debug log:
[07:42:20] -- Executing [...@mainmenu:15] Dial(Zap/1-1,
SIP/105|18|tKk) in new stack
[07:42:20] -- SIP/105-08571180 is ringing
[07:42:39] -- Nobody picked up in 18000 ms
[07:42:39] -- Executing [...@mainmenu:16] Answer(Zap/1-1, ) in new stack
Yes, this is the first method I tried. The transfer only works if it is
done before a media path is set up to the first box (not answered by the
IVR). If it is answered then transferred, I get a 500 internal server error
back from the ITSP and the call dies. I never see anything hit the second
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
18 (Ring Begin)...
[Jan 16 10:38:57]
Is there any product that's a single port mini-PCI FXS card?
I'm aware of the Openvox A400M
http://www.openvox.com.cn/products.php?genre_id=39, but I really only
wanted one port.
How about a single or dual port PCI or PCI express FXS card?
Basically I wanted to build a small linux router
Adam Moffett schrieb:
How about a single or dual port PCI or PCI express FXS card?
Basically I wanted to build a small linux router with one or two phone
ports.
I'd recommend Sangoma's new B700 FlexBRI hybrid card (4 BRI ports,
2 FXS/FXO)
Thanks Philipp,
This and everything else I see out there is a bit more than I need :)
I'm sure a single or dual port analog FXS card is not something most
people want though, otherwise somebody would be selling it.
Thanks anyway though.
I'd recommend Sangoma's new B700 FlexBRI hybrid card
hi
i want to setup an asterisk in my home i always worked whit digium hardware
but have any one try the chines cards or the openbox or sangoma?
is just for home 2 FXS and 2 FXO i need an aceptable audio and the pc will
do only asterisk and is a dual core 1G RAM
the cheaper chines cards will be ok?
Hi,
I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most
24 ports) with 2 ethernet interfaces for network/switch redundancy.
So far I've only found the Grandstream GXW4008.
I've searched similar brands such as Linksys and higher-end brands such as
Quintum, but they
How many times have you been on a conference call and some other
participant puts their line on hold, leading to their hold music
making conversation impossible for the rest of the group?
This scenario happens to me all the time and it drives me NUTS. I
prefer no hold music, because I am
The Asterisk.org development team has published Asterisk 1.4.23-rc4.
This release candidate is available for download from
http://downloads.digium.com/.
A number of critical issues have been resolved since the last release
candidate for 1.4.23. We hope to have this be the final release
I don't know of any ATA like that except the grandstream.
The service provider grade way to do this would probably be a Cisco (or
similar) with a T1 interface and a channel bank to break the T1 into 24
FXS ports.
Hi,
I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably
I would be more worried about the ATA gateway failing than the switch, as
you have found yourself. How about two gateways and two phones on
everyone's desk :)
j
On Fri, 16 Jan 2009, Adam Moffett wrote:
I don't know of any ATA like that except the grandstream.
The service provider grade
On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote:
My provider migrated from an old EOL softswitch to Trixbox.
I have a number (8159093011) on a different server on a different network.
It appears as though the incoming calls are trying to authenticate against
that
Agg, I felt bad about being pedantic. How about splitting the load and
reducing the single point of failure? Instead of one big ATA how about a
number of smaller ones (two port) split between your switches?
j
On Fri, 16 Jan 2009, Jeff LaCoursiere wrote:
I would be more worried about the
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
18 (Ring Begin)...
[Jan
On Fri, Jan 16, 2009 at 4:10 PM, sean darcy seandar...@gmail.com wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55]
Yes, BUT .. not 100% and discontinued in 1.4.22 on ...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Friday, January 16, 2009 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial
Why not do a zap restart instead of restarting asterisk? You could write
an AGI to do the ZR when the condition occurred and lines where empty.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sounds like an awful hack. What does DAHDI do that Zaptel does not?
Sounds more like a post for the bugs list
On Fri, Jan 16, 2009 at 4:48 PM, Danny Nicholas da...@debsinc.com wrote:
Why not do a zap restart instead of restarting asterisk? You could write
an AGI to do the ZR when the
So Zap restart is a worse hack than restarting *? What DAHDI does that
Zaptel doesn't -
1. Makes Digium lawyers happy
2. Gives Developers headaches
3. Makes Coffee and soft drink manufacturers happy since we need so much
caffine.
4. Make priests happy since we have to go to confession about it.
Both are hacks. Neither are good hacks.
On Fri, Jan 16, 2009 at 5:04 PM, Danny Nicholas da...@debsinc.com wrote:
So Zap restart is a worse hack than restarting *? What DAHDI does that
Zaptel doesn't -
1. Makes Digium lawyers happy
2. Gives Developers headaches
3. Makes Coffee and soft
On Fri, 2009-01-16 at 15:52 +, Julian Lyndon-Smith wrote:
Can anyone who has used both comment on the pros and cons ? Need to buy
about 30 of these, for a small company with limited IT support.
For evaluation a got a couple of different phones.
gxp2000, cheap, but it works, some people
Hello list,
Can someone please point me out why would a stream like the following
only write ONE line (the first) on the given file?
Action: login
Username: test
Secret: 123456
Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename: voicemail2.conf
Action-00: Append
Cat-00:
On Fri, Jan 16, 2009 at 6:12 PM, Örn Arnarson o...@arnarson.net wrote:
Am I missing something? Any ideas appreciated.
No you are not missing anything. The Asterisk CDR implementation has a
number of issues and one CDR per bridge is one of them. There is
currently a re-design discussion going on
Danny Nicholas wrote:
Why not do a zap restart instead of restarting asterisk? You could write
an AGI to do the ZR when the condition occurred and lines where empty.
Yes, a cron job to restart zaptel would cut off any call then existing.
But how would I test for it? I can imagine:
On Friday 16 January 2009 16:47:40 Jose P. Espinal wrote:
Can someone please point me out why would a stream like the following
only write ONE line (the first) on the given file?
Action: login
Username: test
Secret: 123456
Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename:
On Friday 16 January 2009 17:43:21 sean darcy wrote:
Danny Nicholas wrote:
Why not do a zap restart instead of restarting asterisk? You could
write an AGI to do the ZR when the condition occurred and lines where
empty.
Yes, a cron job to restart zaptel would cut off any call then
Tilghman Lesher wrote:
On Friday 16 January 2009 17:43:21 sean darcy wrote:
Danny Nicholas wrote:
Why not do a zap restart instead of restarting asterisk? You could
write an AGI to do the ZR when the condition occurred and lines where
empty.
Yes, a cron job to restart zaptel would cut off
On Sat, Jan 17, 2009 at 3:08 AM, Steve Murphy m...@digium.com wrote:
Greyman--
I've been thinking of Benny Amorsen's comments on Simple CDRs...
This is tricky... I need to create these CDR's for the billing system:
src: A start: e1 ans: e2 end: e6 dst: B disp: ANSW
src: A start: e4
Thank you very much, Tilghman,
I did not respond before because for some reason the IP of the list server was
blacklisted on spamhaus.org and I was not getting the messages
see:
2009-01-16 22:47:49 H=(lists.digium.com) [216.207.245.1]
F=asterisk-users-boun...@lists.digium.com rejected
Marco,
The configs work fine for me. I can receive calls with no problem. Now,
were you able to dial using the sim card? I cant figure out how I can do it
since asterisk doesnt have a channel to place call through the portech
gateway.
On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno
I want to dial out using the sim card. What I did, I have used the SIP
channel ex:
Channel: SIP/thenum...@mv378
It shows the called is being made in the dialplan, but the number I have
entered does not dial, it just goes straight to the specified dialplan
extensions.
Then what I did, in the
Sorry for bothering you, but I got it, I just had to put # in callnum!
On Sat, Jan 17, 2009 at 1:44 AM, Pascal Bruno tipas...@gmail.com wrote:
I want to dial out using the sim card. What I did, I have used the SIP
channel ex:
Channel: SIP/thenum...@mv378
It shows the called is being
The EdgePBX FX08 has up to eight ports (using Digium/compatible
modules), a couple of ethernet interfaces, and runs Astfin2 (Asterisk
1.4.21 and uClinux 2.6.22 on a Blackfin DSP).
http://www.edgepbx.cn/shop/index.php?controller=productproduct_id=6
or you might prefer its baby brother, the
On Fri, Jan 16, 2009 at 01:24:16PM +, Gordon Henderson wrote:
On Fri, 16 Jan 2009, Alex Balashov wrote:
1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk
I'd suggest not removing /etc/asterisk if that's the only source of your
config files... If you (re)generate them from
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