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 wrote:
> I wasn't able to get much out
check your ulimits :)
On 26 September 2014 17:15, CDR 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 teh same with
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 wrote:
> On Sun, 28 Sep 2014, Anurag Rana wrote:
>
>> I am trying to record the call using MixMonitor.
>
>
> ...
>
>> Now I k
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 bo
> 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 wrote:
> On Mon, Dec 16, 2013 at 12:41 AM, dotnetdub wrote:
>> Always has cleared the entire line..
>>
>
> Interesting, thanks. From where is your Asterisk? Se
Always has cleared the entire line..
On 15 December 2013 16:25, Dotan Cohen wrote:
> On Sun, Dec 15, 2013 at 3:58 PM, Tiago Geada wrote:
>> I would guess you need to recompile ?
>>
>
> I was under the impression that the library was dynamically linked.
>
> I am using the Ubuntu binaries for Aste
Yup - its definitely doable in FS.
On 15 December 2013 21:18, Patrick Lists
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 eliminate transcoding when un
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 o
ow 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 wrote:
> > Looks like a DNS issue.
>
Looks like a DNS issue.
On 21 April 2013 11:05, Dereck D 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
> output:
>
> --
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
username@dom
On 13 May 2012 17:05, Kaya Saman 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 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
> 1DROP all
> Did you figure out a way to get 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 wrote:
>>
>>
>> On 2
On 2 October 2011 21:36, Sebastian Arcus 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 behind it, in
On 2 October 2011 16:20, Sebastian Arcus 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 and
> voi
On 3 June 2011 22:41, Hans Witvliet 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 software
> You m
On 20 January 2011 18:01, JR Richardson 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 safe_asterisk..
Wo
On 29 November 2010 18:52, C F wrote:
> On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub 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 erro
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 wrote:
> Beautiful..
>
> Asterisk 1.4.22
>
>
Beautiful..
Asterisk 1.4.22
remove extension and hit tab from the CLI..
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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 '7920154238a73e56
On 16 November 2010 22:43, Juan David Diaz wrote:
>
> Juan.
> Linux User #441131
>
>
Maybe best on the linux-ha lists...
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On 1 November 2010 21:20, Steve Edwards 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 SIP IDs 8-12
On 1 November 2010 21:11, Silver Thorne 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?
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On 30 October 2010 19:28, Zeeshan Zakaria 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 anybody else is als
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, size
On 5 October 2010 21:16, bakko 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 the con
On 3 October 2010 15:34, jagan thoutam 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 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 Asterisk install in
>
On 16 September 2010 15:03, Jerry Geis 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
> > grep
On 20 September 2010 05:33, dashy dude 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> wrote:
>
>
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 work
On 12 September 2010 23:56, Thorolf Godawa 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 GDB unf
On 3 August 2010 19:54, Paul Belanger wrote:
> On Tue, Aug 3, 2010 at 2:26 PM, Duncan Turnbull
> 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 I read the announcement.
from freepbx.
On 28 July 2010 21:42, Stefan Schmidt wrote:
> dotnetdub schrieb:
> > Hi List,
>
> > core show channels
> > Channel Location State Application(Data)
> >
> > SIP/102--08e1 *...@from-inside Down(None)
> > SIP/102--08d6 *
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 hands
On 13 July 2010 09:52, Randy R 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 community.
>
> Aft
On 30 June 2010 10:28, Jonas Kellens 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 it is introduced
On 25 June 2010 16:23, Nicholas Hart 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 connecting
> temporarily
On 24 June 2010 19:54, Mark G. Thomas 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 to a value
On Monday, June 21, 2010, Douglas Mortensen 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 really would suspect the
On 18 June 2010 10:38, Deepika Nijhawan wrote:
> 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 compile
On 17 June 2010 16:09, Steve Howes 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 call out Asterisk
>
>
> 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
On 6 June 2010 19:48, bruce bruce 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 uninitiali
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 wrote:
> > Are there any CLI commands to free this up or any other ways without
> having
> > to restart asterisk.
>
> Did you try soft hangup ? Or set an RTP timeo
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)
SIP/107-C
On 9 April 2010 16:46, Tim Nelson wrote:
> - "dotnetdub" 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
Do you seperate your voice and data networks?
On 9 April 2010 14:56, Tim Nelson wrote:
> - "dotnetdub" wrote:
> >
> >
>>
>> >
>> >
>> I would not think you'd need to worry about jitter on a "normal" 100mbit
>>
>
>
> 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 betwe
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