Turn on sip debugging for this peer and watch for the options sending
and response.
If you are getting a response to your options asterisk shouldn't be
marking the peer as unavailable.
is your asterisk behind a firewall?
On 20 March 2015 at 13:42, thufir hawat.thu...@gmail.com wrote:
I wasn't
As the other posters said - try it!
Another option would be to use sox to combine files with some common
part of their filename.
On 28 September 2014 19:39, Steve Edwards asterisk@sedwards.com wrote:
On Sun, 28 Sep 2014, Anurag Rana wrote:
I am trying to record the call using MixMonitor.
check your ulimits :)
On 26 September 2014 17:15, CDR vene...@gmail.com wrote:
I am using Asterisk 12 svn, from today, and after a few thousand
calls, I run out of ports.
This happens eith PJSIOP and regular old SIP. I think it is RTP related.
Any idea how can I troblshoot this. It happened
Hi,
upto asterisk 1.8 you used to get this error if there were more than 1
m= line in an invite... Asterisk was just telling you it was declining
the second. I belive from 10.0 onwards asterisk now just replies back
with port 0 to the stream it isn't interested in...
You can ignore it - if its
libedit
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1.4 1.6 1.8 11.6.0
All compiled and all running on debian 6 or 7
On 16 December 2013 12:27, Dotan Cohen dotanco...@gmail.com wrote:
On Mon, Dec 16, 2013 at 12:41 AM, dotnetdub dotnet...@gmail.com wrote:
Always has cleared the entire line..
Interesting, thanks. From where is your Asterisk
Yup - its definitely doable in FS.
On 15 December 2013 21:18, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 12/15/2013 09:55 PM, CDR wrote:
I have had the issue for years. The problem is that Asterisk
developers are removed from the business. We desperately need simple
way to
Always has cleared the entire line..
On 15 December 2013 16:25, Dotan Cohen dotanco...@gmail.com wrote:
On Sun, Dec 15, 2013 at 3:58 PM, Tiago Geada tiago.ge...@gmail.com wrote:
I would guess you need to recompile ?
I was under the impression that the library was dynamically linked.
I am
Why?
On Wednesday, 20 November 2013, Damian Gonzalez wrote:
Hello,
I have a problem with movistar in Mexico with a sip calls. Movistar send
to me T38 and G729 in the INVITE and they say that I have to ignore T38 and
use G729 in the voice call.
When a fax call is made Movistar send only
Looks like a DNS issue.
On 21 April 2013 11:05, Dereck D derec...@gmail.com wrote:
Hello List.
Last month i started to face a strange issue on an asterisk server
1.8.9.3 built on Centos 5.3 x86_64 dedicated server.
out of the blue UDP stops responding .. and keep getting the following
would it be DNS issue while other users drop
unreachable too?
all operators and SIP Peers go unreachable .. not only unable to register.
oe peer using FQDN the rest are IP addresses.
On Sun, Apr 21, 2013 at 2:18 PM, dotnetdub dotnet...@gmail.com wrote:
Looks like a DNS issue.
On 21
Hi Guys,
Seeing an issue with 1.6.2.17.2 and also 1.6.2.14
When we do call forwarding if the call coming in to be forwarded
asterisk sends the invite out to our ITSP as
username@anonymous.invalid instead of username@domain.
When call comes in with CLI and is forwarded it sends it as
On 13 May 2012 17:05, Kaya Saman kayasa...@gmail.com wrote:
[May 13 13:15:49] DEBUG[3056] pbx.c: FONALITY: This thread has already
held the conlock, skip locking
You should really be posting on the trixbox forums.
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On 27 January 2012 04:49, asterisk jobs asteriskcod...@gmail.com wrote:
Hello everyone,
I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this?
Chain fail2ban-ASTERISK (1 references)
num target prot opt source destination
1
rid of the channel without restarting?
Regards,
Hi Orn
I didn't find a way except a restart once active calls drop to zero.
Regards,
Brian
On Wed, Jul 28, 2010 at 9:45 PM, dotnetdub dotnet...@gmail.com wrote:
On 28 July 2010 21:42, Stefan Schmidt s...@sil.at wrote:
dotnetdub
On 2 October 2011 16:20, Sebastian Arcus s...@open-t.co.uk wrote:
Hello list,
My setup is as follows:
Trunks: 2 sip trunks, one with voipcheap.co.uk, one with sipgate.co.uk
Extensions: 1 hardware sip phone
Asterisk: 1.8.7.0
Everything is working fine, except bridging between the sipgate
On 2 October 2011 21:36, Sebastian Arcus s...@open-t.co.uk wrote:
Just a follow up. I've opened up udp ports 1-2 on the Linux box
(where Asterisk is) and now I have sound. However, bear in mind that the
Netgear router/modem which is connected to the Internet (the Linux/Asterisk
box is
On 3 June 2011 22:41, Hans Witvliet h...@a-domani.nl wrote:
On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote:
Are you suggesting that there are no bugs in 1.4 or 1.6?
I presume that you are aware of the fact that it is impossible to prove
the absence of bugs in any piece of
On 20 January 2011 18:01, JR Richardson jmr.richard...@gmail.com wrote:
Or is there another work around to get ssh console colors using the
Debian * 1.6.0.28 LSB init script?
I also tried 'nocolor = no' in the [options] section of asterisk.conf
with no effect.
Try running asterisk using
On 29 November 2010 18:52, C F shma...@gmail.com wrote:
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote:
Sorry,
what I meant was:
server*CLI remove extension (hit tab)
segfault..
1.4.22
It could be an extension name Where is the error trapping
Beautiful..
Asterisk 1.4.22
remove extension and hit tab from the CLI..
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Sorry,
what I meant was:
server*CLI remove extension (hit tab)
segfault..
1.4.22
It could be an extension name Where is the error trapping if this is the
case.. Who writes this shit?
On 28 November 2010 22:21, dotnetdub dotnet...@gmail.com wrote:
Beautiful..
Asterisk 1.4.22
Hi List,
Anybody any ideas on these?
[Nov 26 15:14:10] WARNING[3265] chan_sip.c: Remote host can't match request
NOTIFY to call '1c4890a52552c39b0b81702353087...@192.168.33.12'. Giving up.
[Nov 26 15:16:44] WARNING[3265] chan_sip.c: Remote host can't match request
NOTIFY to call
On 16 November 2010 22:43, Juan David Diaz juanch...@gmail.com wrote:
Juan.
Linux User #441131
Maybe best on the linux-ha lists...
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On 1 November 2010 21:11, Silver Thorne zora...@gmail.com wrote:
Hey;
Anyone see this before:
[Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
6839, digest has 3169
G
`
Is it causing a problem for you?
--
On 1 November 2010 21:20, Steve Edwards asterisk@sedwards.com wrote:
On Mon, 1 Nov 2010, Cary Fitch wrote:
Any small system should:
Use IPTABLES and block any parts of the world you don't need access
to/from. Start with any Class A address that is probing your system.
Make your
On 30 October 2010 19:28, Zeeshan Zakaria zisha...@gmail.com wrote:
My main asterisk server is under unusual heavy attack, and so far Fail2Ban
has blocked about 30 IPs, from various different countries. At this time it
is blocking about 1 IP address every few minutes.
Just wondering if
Hi List,
I need to modify the callerID name of the call coming back when a parked
call returns to the extension that parked it when it times out.
Looking at app_parkandannounce.c
/* Now place the call to the extention */
snprintf(buf, sizeof(buf), %d, lot);
memset(oh, 0,
On 5 October 2010 21:16, bakko asannu...@gmail.com wrote:
Hello,
I'm trying to configure Asterisk with Radius cdr support.
Asterisk version 1.6.2.13
Server Radius: Freeradius version 1.X
Radius client: radiusclient-ng version 0.5.5
With the Asterisk core debug on 1 when a call terminate,
On 3 October 2010 15:34, jagan thoutam jaganthou...@gmail.com wrote:
how can i disable other end hangup when i recive incomming call tfrom
asterisk
Get some hot girls to talk to the other end?
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On 26 September 2010 18:48, bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
I am stack between two identical systems (2U Twin2, 4 nodes, SuperMicro)
servers that have the same exact specs except for HDDs. These nodes will all
either have Asterisk installed with CentOS or will have
On 20 September 2010 05:33, dashy dude dashy_v2...@yahoo.com wrote:
Hi,
I tried disabling cdr_addon_mysql.so.
Still error comes let's say once a day or so.
Is there anything else I can do about?
rgds
--- On Thu, 9/9/10, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de
On 16 September 2010 15:03, Jerry Geis ge...@pagestation.com wrote:
Jerry Geis wrote:
below is the results of the command.
grep -r ztconfig /etc/.
grep: /etc/./httpd/run/asterisk.ctl: No such device or address
grep: /etc/./httpd/run/dbus/system_bus_socket: No such device or address
Hi All,
Is it possible to specify more than 1 localnet? I know this is an odd
question. I have a customer that has multiple sites linked by VPN.
Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24
We want to allow some access to the public IP address at the main site. For
this to
On 12 September 2010 23:56, Thorolf Godawa nos...@godawa.de wrote:
Hi Luki an all others who answered,
Try kill -6 (i.e. SIGABRT). That usually triggers a core dump for me.
yes, that works for testing and creates a coredump.
Thank you very much for your answer!
PS: Running Asterisk under
On 3 August 2010 19:54, Paul Belanger paul.belan...@polybeacon.com wrote:
On Tue, Aug 3, 2010 at 2:26 PM, Duncan Turnbull dun...@e-simple.co.nz
wrote:
FreePBX is still the same, V3 is still the same, this is a fork from some
guys who had got involved (or maybe paid some money)
That is how
Hi List,
Asterisk 1.4.22 built by root @ carl on a i686
Purely SIP
Linksys SPA962 with 932 sidecar and also Cisco SPA508 / 525G with Sidecars
Have an issue with this happening with a number of my customers.
Customer hits the ringing BLF on the sidecar to pickup the call incoming on
another
On 28 July 2010 21:42, Stefan Schmidt s...@sil.at wrote:
dotnetdub schrieb:
Hi List,
snip
core show channels
Channel Location State Application(Data)
SIP/102--08e1 *...@from-inside Down(None)
SIP/102--08d6 *...@from-inside Ring(None)
SIP/102
On 13 July 2010 09:52, Randy R randulo2...@gmail.com wrote:
Many of you are interested in and have used or recommended fail2ban
for your linux boxes. I finally installed it on our FreeBSD server (no
asterisk, hence the OT) with the help of a friend from the VoIP Users
Conference and Asterisk
On 30 June 2010 10:28, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
I also thought about echo because the Zoiper softphone is used with a
headset. But that didn't explain why the echo also appeared on the analogue
phone + gateway.
It will present it self on the analogue phone when
On 24 June 2010 19:54, Mark G. Thomas m...@misty.com wrote:
Hi,
I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to
both a local Asterisk server and also with a trunk directly to
a VOIP provider. Everything works great, except I'm having a problem
setting the outbound caller ID
On 25 June 2010 16:23, Nicholas Hart nh...@partsauthority.com wrote:
Hi,
I am relatively new to Asterisk and am looking for help in configuring an
IP based phone. This phone is not on the same subnet as the PBX. I read
that there could be an issue with NAT so I am bypassing this by
On Monday, June 21, 2010, Douglas Mortensen d...@impalanetworks.com wrote:
I just had a user report that they called out to someone on a cell phone this
morning, and after a short conversation, the call was dropped/lost. The
person on the cell phone says that this is very rare.
Snip
I
On 18 June 2010 10:38, Deepika Nijhawan deepika.nijha...@oxygen8.comwrote:
Cdr status shows:
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: no
It is not showing ‘CDR registered backend’
Thanks,
Deepika
Have you compiled asterisk-addons and selected to
On 17 June 2010 16:09, Steve Howes steve-li...@geekinter.net wrote:
On 17 Jun 2010, at 15:58, Mike wrote:
I have a Cisco SPA525G latest firmware, and very often when I attempt a
transfer I get a network error message when I press Dial on the transfer.
I never get that erroron a simple
The trouble with whitelisting, or using iptables to block 5060 (in fact
* is behind a router - 5060 is port forwarded) is that traveling
employees wouldn't be able to register with inbound extensions. We set
up our travelers so they can connect from wherever, and be treated as if
they were
On 6 June 2010 19:48, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Just did an Asterisk 1.6.x (repo install) and FreePBX (source install).
When trying to dial a number, I get this:
tel*CLI Use of uninitialized value in hash element at /var/www/html/panel/
op_server.pl line 3367.
Use of
Hi Luki,
Thank you so much.. The soft xx worked perfectly. The rtptimeout is
excellent also.
Regards,
S.
On 5 May 2010 23:59, Luki lugos...@gmail.com wrote:
Are there any CLI commands to free this up or any other ways without
having
to restart asterisk.
Did you try soft hangup channel?
Hi List,
If we have a scenario where a customer is using a telephone and their WAN
link goes down for example the channel in asterisk stays marked as in use
and this affects the subscribe also.
*CLI core show channels
Channel Location State Application(Data)
Do you seperate your voice and data networks?
On 9 April 2010 14:56, Tim Nelson tnel...@rockbochs.com wrote:
- dotnetdub dotnet...@gmail.com wrote:
I would not think you'd need to worry about jitter on a normal 100mbit
LAN unless there is heavy traffic or people are running
On 9 April 2010 16:46, Tim Nelson tnel...@rockbochs.com wrote:
- dotnetdub dotnet...@gmail.com wrote:
Do you seperate your voice and data networks?
Un-top-posting...
Yes, I separate voice and data. Typically this is done using separate
switches where possible, other times, using
I would not think you'd need to worry about jitter on a normal 100mbit
LAN unless there is heavy traffic or people are running their PC's through
the phone (don't remember if the 501 has two ethernet ports...). Typically
the quality issues are introduced on your WAN connectivity between the
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