Hi List,
is there somebody how is able to help me here? Or at least to get more
details why this occurs?
best regards
Christoph
Am 08.06.2011 18:00, schrieb Christoph Timm:
Hi List,
I'm running into trouble, if I perform a 'yum update' on my AsteriskNOW.
Currently I'm running Asterisk
Doug,
I see that this patch is for 1.6.0.1
But we use version 1.6.2.12.
And if I can see it, this patch is already included in version 1.6.2.12. Or am
I wrong?
Regards,
Arjan Kroon
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
2011/6/19 Claude Hayn chayn...@gmail.com
ITSP failover for PRI
** **
Hello All,
** **
We’re using an Asterisk based SIP-T1 trunking gateway and would like to
implement failover between two ITSPs.
** **
What about incoming calls ?
Do you have a way to have calls that
On 06/20/2011 04:20 AM, Olivier wrote:
What about incoming calls ?
Do you have a way to have calls that normally comes from ITPS1 to
comes from ITSP2 ?
No, there is no BGP for the PSTN.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Two requests, not from me but the community.
1. Don't top post
2. When you find your solution, reply to this thread so others will be
(silver) spoon fed the answers and blindly accept them without trying things
and going through a learning curve and experimentation when they find your
post in
2011/6/20 Alex Balashov abalas...@evaristesys.com
On 06/20/2011 04:20 AM, Olivier wrote:
What about incoming calls ?
Do you have a way to have calls that normally comes from ITPS1 to
comes from ITSP2 ?
No, there is no BGP for the PSTN.
Yes, that's what I thought but you never know ;-)
On 06/20/2011 05:13 AM, Olivier wrote:
Yes, that's what I thought but you never know ;-)
(Maybe SS7 offers such redundancy but I've got no experience of any
king in this domain).
SS7 certainly offers link redundancy, but the issue is that your
numbers can't just be flash-ported to a
Arjan Kroon | Mobillion wrote:
And if I can see it, this patch is already included in version 1.6.2.12. Or am
I wrong?
That I can't answer. I'm still using 1.4.x and am experimenting with
1.8.x. I recall reading that it wasn't supported directly until 1.8
without patches.
Doug
--
Ben
Oke,
But is there a patch from version 1.6.2.12?
Greeting,
Arjan
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle
Verzonden: 20-06-2011 11:36
Aan: Asterisk Users Mailing List - Non-Commercial
It is not included. It was supposed to be included in 1.6.3, but that verison
of Asterisk was never released, it became 1.8.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Arjan Kroon|Mobillion
Sent:
Dears;
OK, I have two things now:
1) When I do reload from the asterisk CLI, then all the skinny phones are
reset. This is very bad thing, how to avoid this from happening in each reload?
Even if the reload will be done to take sip configuration !!
2) The line tone that is heared (the
But Steve... didn't you just top post?
On Mon, Jun 20, 2011 at 10:52 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Two requests, not from me but the community.
1. Don't top post
2. When you find your solution, reply to this thread so others will be
(silver) spoon fed the answers and
Hi
In asterisk channel ,I so number of variable regarding the Codec ,Can
anyone explain what are those variable variable means.Below are the
variables
1. chan-readformat
2. chan-writeformat
3. chan -rawreadformat
4. chan -rawwriteformat
5. chan-nativeformats
Thanks
Arjan Kroon | Mobillion wrote:
But is there a patch from version 1.6.2.12?
Not that I can see. You could try applying the patches against that
version and see if they apply cleanly.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
On Mon, Jun 20, 2011 at 5:39 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
Oke,
But is there a patch from version 1.6.2.12?
Greeting,
Arjan
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
We try the patch against the version 1.6.12 but we got many reject failures.
(see below)
Unfortunately we have to upgrade to 1.8
Tx,
Arjan
File to patch: apps/app_dial.c
patching file apps/app_dial.c
Hunk #1 FAILED at 111.
Hunk #2 FAILED at 123.
Hunk #3 FAILED at 164.
Hunk #4 succeeded at
Ryan,
The problem is not with SIP, but with ISDN.
Or is this patch also applied for ISDN calls?
Arjan
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Ryan Wagoner
Verzonden: 20-06-2011 13:51
Aan: Asterisk
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Ryan Wagoner
Sent: Monday, June 20, 2011 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Connected Line
I didn't think it was possible if the endpoints, or Asterisk was behind a NAT.
Someone please correct me if I am wrong.
http://www.voip-info.org/wiki/view/Asterisk+sip+directrtpsetup
From: Sagbo Romaric
Sent: Sun 6/19/2011 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
If you can't ping between the two end points, then you can't do direct RTP.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Terry Brummell
Sent: Monday, June 20, 2011 8:16 AM
To: Asterisk Users Mailing
On 20/06/2011 8:18 AM, Steve Underwood wrote:
On 06/20/2011 03:38 AM, khalid touati wrote:
Hi Guys,
I solved temporarely my issue by kind of tricking Asterisk, I used
the following line instead of the old:
exten = h,n,System('/usr/local/
bin/fax2mail -p -f ${FAXFILENOEXT} --cid-number
Dear,
Can you provide me the firewall rules which help me to address this issue in
the
case of my architecture.
I try some rules without success.
Best,
Romaric SAGBO
De : Eric Wieling ewiel...@nyigc.com
À : Asterisk Users Mailing List - Non-Commercial
On 20/06/11 13:18, Eric Wieling wrote:
If you can't ping between the two end points, then you can't do direct RTP.
precisely. If 10.10.9.1 isn't reachable from the network that 10.10.8.1
is on then 10.10.8.1 isn't going to be able to send RTP to 10.10.9.1.
You need to add routes to the
Hi,
Have anybody integrated OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with
Asterisk?
Thanks,
Gopal
--
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New to Asterisk? Join us for a live introductory
Ok, thanks,
Can you help me to have this kind of rules ?
I try with iptables without success.
Best,
Romaric SAGBO
De : Paul Hayes p...@provu.co.uk
À : asterisk-users@lists.digium.com
Envoyé le : Lun 20 juin 2011, 16h 39min 32s
Objet : Re: [asterisk-users] Re :
Check out this product.
http://www.i6net.com
On Mon, Jun 20, 2011 at 9:40 AM, Gopal krishnan gopalakrishnan...@gmail.com
wrote:
Hi,
Have anybody integrated OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with
Asterisk?
Thanks,
Gopal
--
The only way this will work is to remove NAT from this scenerio.
And it's not Asterisk's fault per se. The phones are built 'that way'
also. That's why other free providers don't use SIP phones, but build
their own client software.
The others are trying to tell you SIP/RTP doesn't work the
You can ask a million more times. The answer will not change.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sagbo Romaric
Sent: Monday, June 20, 2011 11:05 AM
To: Asterisk Users Mailing List -
Now I add route and it's work now.
But, I need to improve it because I need to have direct RTP without to have add
the rules to firewall.
Any client behind his NAT can talk with another behind his NAT.
Best for all of you.
Romaric SAGBO
Ingénieur Réseaux et Télécoms.
BP 613 Porto Novo
Tél:(+229)
On 20 Jun 2011, at 16:33, Sagbo Romaric wrote:
Any client behind his NAT can talk with another behind his NAT.
Still not possible.. The internet doesn't really work like that. SIP even more
so.
S--
_
-- Bandwidth and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Le 20/06/2011 04:40, Gopal krishnan a écrit :
Have anybody integrated
OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with Asterisk?
Voiceglue works for me: http://www.voiceglue.org/
Thanks,
- --
Jean-Denis Girard
SysNux
hello liste
i have create an menu like below
exten = my_number,1,Ringing()
exten = my_number,2,Wait(4)
exten = my_number,3,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,2,Background(${sounds_path}welcome)
exten = #,1,Goto(menu,s,1)
exten =
Put this line in
Exten = s,3,goto(home,s,1)
You are experiencing fall through when no dtmf is pressed and since there
is no handling, the call hangs up.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent:
I missed one important parameter in my setup of BLF for polycom phones (at
least if you want to do one touch directed pickup)
In sip.conf add
notifycid=yes
the notifycid=yes causes asterisk to add a target uri = callID to the XML
of the SIP notify. Without this target uri the Polycom
I have problems using the call pickup under Asterisk 1.8.4.2. I have
another Asterisk with 1.6 - and it is working fine with the same settings.
I have setup the same callgroup and pickupgroup for all extensions in
sip.conf - just to make things simple for testing. The sequence *8 seems
to be
Dears,
i am using sipp to test asterisk(1.6.22) performance ,but when i limit the
calls to 150 ,only 100 active calls on asterisk found ?why
sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
On 06/20/2011 01:09 PM, Khaled W. Chehab wrote:
Dears,
i am using sipp to test asterisk(1.6.22) performance ,but when i limit the
calls to 150 ,only 100 active calls on asterisk found ?why
sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150
You did not provide any log output, or anything that
On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Dears;
snip
Have you thought about perhaps just flashing the phones to use the SIP
firmware?
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
--
On Mon, Jun 20, 2011 at 3:52 AM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:
Two requests, not from me but the community.
1. Don't top post
*cough*
2. When you find your solution, reply to this thread so others will be
(silver) spoon fed the answers and blindly accept them without
I'm using the sip firmware.. It's alright.. I feel like I'm not receiving
all the features I should.. But MWI works and multiple call appearance..
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, June 20,
On Mon, Jun 20, 2011 at 7:44 AM, Larry Moore lmo...@starwon.com.au wrote:
snip
I personally have considered this behaviour to possibly be a bug.
Once a fax is sent, the sending fax machine typically hangs up the call -
sending the call to the h extension. It's the same as if you are on an
On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,2,Background(${sounds_path}welcome)
exten = #,1,Goto(menu,s,1)
exten = i,1,Playback(${sounds_path}error-key)
exten =
Hello list,
how can I get the second character/cipher of an extension ?
If I have : exten = 12345,n,NoOP()
How can I get 2 ?
If I have : exten = 787,n,NoOP()
How can I get 8 ?
Thanks !
Kind regards,
Jonas.
--
_
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jonas Kellens
Sent: Monday, June 20, 2011 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Get second cipher
how can I get the second character/cipher of an extension ?
If I have : exten = 12345,n,NoOP()
How can I get 2 ?
${EXTEN:1:1}
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
On Mon, Jun 20, 2011 at 2:09 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
**
Hello list,
how can I get the second character/cipher of an extension ?
snip
I vaguely recall that to get a substring out of an extension variable, you
would use it in the format ${EXTEN:offset:length}, so for
Replying to my own post:
I have done some more digging, disabling parts of configuration files
one at a time - since there is nothing useful in the console for this
problem. Turns out that if I enable the following lines in features.conf:
parkext = 700
parkpos = 701-720
context = parkedcalls
From: Warren Selby wcse...@selbytech.com
Sent: Monday, June 20, 2011 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problem with ReceiveFAX app from FFA
On Mon, Jun 20,
It could be your OS limit try ulimit command.
--
Sent from my iPhone
On Jun 20, 2011, at 2:21 PM, Kevin P. Fleming kpflem...@digium.com
wrote:
On 06/20/2011 01:09 PM, Khaled W. Chehab wrote:
Dears,
i am using sipp to test asterisk(1.6.22) performance ,but when i
limit the
calls to
On Mon, Jun 20, 2011 at 2:43 PM, Bryant Zimmerman brya...@zktech.comwrote:
I concur we use the h extension to log inbound faxes to a database and
then we process them outside the asterisk platform. Our biggest issue with
ReceiveFAX is about a 20% t.38 negotiation fail ratio. We then force
I tried the ulimit
[root@localhost ~]# ulimit
Unlimited
Then
sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150
SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice)
SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice)
SIP/127.0.0.1:5061-0 s@from-trunk:4
Oh! Wait you set ulimit for running shellYou should set ulimit on
asterisk. Also you can set ulimit command on asterisk startup file /
etc/init.d/asterisk and restart asterisk also you can set in
limit.conf file
I had this issue before and I solved that way.
--
Sent from my iPhone
On
Can you please specify more
1-how to set the ulimit on
[root@localhost ~]# ulimit
unlimited
[root@localhost ~]# ulimit --help
-bash: ulimit: --: invalid option
ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit]
-
How to set
Sorry, to not answer before!
Thanks a lot, as sun as i am able i will test this setup!
[]'sf.rique
On Fri, Jun 17, 2011 at 4:50 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On Thu, 2011-06-16 at 19:12 -0300, Henrique Fernandes wrote:
It is possible to log queue in mysql without turning on
The problem remains even when
I add to /etc/init.d/asterisk
ulimit -n 65536
[root@localhost ~]# ulimit -a
core file size (blocks, -c) 0
data seg size (kbytes, -d) unlimited
scheduling priority (-e) 0
file size (blocks, -f) unlimited
pending signals
Inviato da iPhone
Il giorno 18/giu/2011, alle ore 06:40, Larry Moore lmo...@starwon.com.au ha
scritto:
On 18/06/2011 5:36 AM, Matteo Campana wrote:
Inviato da iPhone
Il giorno 16/giu/2011, alle ore 16:37, Eric Wielingewiel...@nyigc.com ha
scritto:
We experience the same thing.
Dear Stefan;
First of all, I tried skinny and I faced two major problems (so if I am going
to face same problems in sccp then no need to use sccp, so please advise).
The two problems that I faced them are:
1) When I do reload then the skinny channel is reloaded and that will cause a
restart
If I need to use SIP, from where to get the suitable firmware for these Cisco
IP Phones 7942G?
Where do u download the SIP firmware usually for your Cisco IP Phones?
Your kindly help is highly appreciated.
Regards
Bilal
---
I'm using the sip firmware.. It's alright.. I feel like
I have discovered that if I enable pickupsound = beep in features.conf,
if I try to do a pickup with *8, the calling channel keeps on ringing,
while the phone where I pick-up from shows that the call has been
answered (I don't know where though). Also, it seems to completely
bugger up my
You are supposed to go via cisco and support contract BUT Google is your
friend (JFGI)
Sent from my iPhone
On Jun 20, 2011, at 6:44 PM, bilal ghayyad bilmar...@yahoo.com wrote:
If I need to use SIP, from where to get the suitable firmware for these Cisco
IP Phones 7942G?
Where do u
This has been fixed only last month, see
https://issues.asterisk.org/view.php?id=18654 and try bug18654.diff.txt
That will avoid the deadlock, but it's not the proper fix, there are other
issues that could trip you up,
mainly to do with race conditions with multiple channels picking up the same
I have an asterisk 1.4.26 mte running.
Sometimes inbound caller ID displays asterisk
These calls do not show up on the CLI nor the CDR.
I read somewhere that these are asterisk hack attempts.
Is this true?
What is the best way to defend from this?
I know a secure password
On Mon, Jun 20, 2011 at 6:10 PM, Robert-iPhone rhuddles...@gmail.comwrote:
You are supposed to go via cisco and support contract BUT Google is your
friend (JFGI)
The support contract from Cisco is only US $8.99 on CDW
I really hate to link to my own blog, but I do have a post on there
On Mon, Jun 20, 2011 at 6:33 PM, ERIC HERRON e...@lanline.com wrote:
I have an asterisk 1.4.26 mte running.
** **
Sometimes inbound caller ID displays “asterisk”
** **
These calls do not show up on the CLI nor the CDR.
** **
I read somewhere that these are asterisk hack
On Mon, Jun 20, 2011 at 8:38 PM, Claude Hayn chayn...@gmail.com wrote:
snip
Can someone please make suggestions or point us in the right direction to
resolve this no audio issue?
No audio is usually a NAT issue. Verify you have the proper NAT settings on
your ITSP2 account
Sent from my iPhone
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users
I nominate this for most imaginative use of Asterisk-users of 2011.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Jun 20, 2011, at 8:43 PM, Marcelo
On Mon, Jun 20, 2011 at 11:47 PM, Alex Balashov
abalas...@evaristesys.com wrote:
I nominate this for most imaginative use of Asterisk-users of 2011.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax:
Dear Asterisk Users,
I have a Sipura 2000 device, and since last few days I have been searching
for its latest firmware for upgrade. Googling tells me that Cisco has
stopped the support for this device and I dont have definite idea on where
would I be able to find the firmware to upgrade my
Dear all,
New day has brought me luck :)
I got the solution. Please find the link for the upgrades. I will try it at
my end and if it doesnt work will inform the thread otherwise will not
disturb you.
http://www.quickconnectusa.com/resources/sipura.asp
Cheers,
Amol
On Tue, Jun 21, 2011 at 10:26
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