Hi all!
I am running asterisk 13.1.0 on Ubuntu server 16.04. There are two IP
addresses from the same subnet set on one interface, and bindaddr is set to
the second on them in sip.conf and in iax.conf.
Incoming connections work as expected. However, for outgoing connections it
seems that asterisk t
Hey guys, any thoughts on that? Probably a bug or is it a default behavior?
On Thu, Mar 9, 2017, 2:05 PM Kseniya Blashchuk wrote:
> Hi all!
> I am running asterisk 13.1.0 on Ubuntu server 16.04. There are two IP
> addresses from the same subnet set on one interface, and bindaddr is set
Hi!
Attached sip.conf and interface config as well. In this case we use only
TLS, but I have checked with TCP - same situation, 192.168.0.172 is used as
a source. For UDP 192.168.0.177 is used as expected.
пн, 13 мар. 2017 г. в 2:37, Joshua Colp :
> On Sat, Mar 11, 2017, at 11:50 AM, Ksen
:55, Joshua Colp :
> On Mon, Mar 13, 2017, at 03:52 AM, Kseniya Blashchuk wrote:
> > Hi!
> > Attached sip.conf and interface config as well. In this case we use only
> > TLS, but I have checked with TCP - same situation, 192.168.0.172 is used
> > as
> > a source. For UD
r TCP
and UDP? I thought that maybe the application does not request the bound
address as a source in case of TCP...
пн, 13 мар. 2017 г. в 14:37, Joshua Colp :
> On Mon, Mar 13, 2017, at 08:31 AM, Kseniya Blashchuk wrote:
> > Yes, look:
> > netstat -nlp | egrep '506[01]'
Ah ok, thank you for checking.
I'll maybe also try with the latest asterisk and/or other distro and see if
this behavior is reproduced.
пн, 13 мар. 2017 г. в 14:46, Joshua Colp :
> On Mon, Mar 13, 2017, at 08:43 AM, Kseniya Blashchuk wrote:
> > Mmh sorry I'm afraid I did not u
Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel 4.4.0-66-generic and
Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same behavior.
Joshua, maybe you can advice what can be done further?
пн, 13 мар. 2017 г. в 14:52, Kseniya Blashchuk :
> Ah ok, thank you for checking.
>
Ok, thank you for the assistance!
пн, 13 мар. 2017 г. в 16:38, Joshua Colp :
> On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote:
> > Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel 4.4.0-66-generic
> > and
> > Centos 7 kernel 3.10.0-514.10.2.el7.x86_64
ame issue with pjsip, but more
> chances of support on the issues tracker of Asterisk to have help.
>
> Regards.
>
>
> --
> Ludovic Gasc (GMLudo)
> Lead Developer Architect at ALLOcloud
> https://be.linkedin.com/in/ludovicgasc
>
> 2017-03-13 14:41 GMT+01:00 Kseni
Can it happen that the routes lead the traffic through another interface?
Did you try a packet capture with tcpdump? Do the packets really leave the
usb adapter? Can asymmetric routing be in effect?
Maybe there were some static routes that disappeared when the adapter was
unplugged...
On Thu, Apr
e.linkedin.com/in/ludovicgasc
>
> 2017-04-16 21:36 GMT+02:00 Kseniya Blashchuk :
>
>> Hi!
>>
>> Unfortunately pjsip is broken in Ubuntu Asterisk installed from repo. Yes
>> I also thought to try with pjsip, just to know if it's also affected. I'll
&
If SIP goes to the same provider then yes. Still I would check a packet
capture for better understanding. BTW, did you try iax debug?
чт, 20 апр. 2017 г. в 19:46, Carlos Chavez :
> On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:
>
> Can it happen that the routes lead the traffic throug
12:36 AM Carlos Chavez wrote:
> On 4/20/17 2:37 PM, Kseniya Blashchuk wrote:
>
> If SIP goes to the same provider then yes. Still I would check a packet
> capture for better understanding. BTW, did you try iax debug?
>
> чт, 20 апр. 2017 г. в 19:46, Carlos Chavez :
>
>> On
JFYI - https://issues.asterisk.org/jira/browse/ASTERISK-26922
чт, 20 апр. 2017 г. в 11:38, Kseniya Blashchuk :
> Hi!
> The issue did not reproduce with pjsip. As for ppa - somebody recommended
> me ppa:sapian/asterisk. Does anybody use it maybe?
>
>
> вт, 18 апр. 2017 г. в
Hi all!
We are going to move from a Shoretel PBX system to asterisk shortly, and we
are looking for some similar functionality as described in this video:
https://www.youtube.com/watch?v=tqESWqrUxGA. I know about Flash Operator
Panel for asterisk, but maybe somebody can make other recommendations?
Not sure maybe there's a better solution but I thought about using another
peer with type=user for incoming connections.
On Mon, May 22, 2017, 6:13 PM Benoit Panizzon
wrote:
> Hello List
>
> I work at an SIP Provider and we have added and SBC in front of our
> Voice Switch to protect it.
>
> Thi
Same about me - need to re-enable membership all the time. Annoying ((
пн, 12 июн. 2017 г. в 15:59, John Novack :
> Not just gmail
> Happening as well with Comcast.net
>
> My Comcast address is set to forward to another domain, as Comcast seems
> to now block sending mail with a non Comcast "from
Hi!
You can also consider using fail2ban but it's more suitable to block
bruteforce attempts.
On Tue, Aug 15, 2017, 11:56 PM Patrick Laimbock
wrote:
> Hi Mike,
>
> On 15-08-17 21:37, mdiehl wrote:
> > Hi all,
> >
> > Lately, I've seen an increase in the number of attacks against my system
> from
Well, correct me if I'm wrong, but I would say this conversation you have
posted is a bit outdated, now fail2ban can be used with asterisk security
log
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger.
On Thu, Aug 17, 2017, 4:53 AM Telium Technical Support
wrote:
> Keep
Hi all!
Asterisk 13.1.0 Ubuntu 16.04, all latest.
Can anybody explain this to me - I run Originate command from dialplan:
same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum})
and I get crazy sound distortion in the conference, and I see that
transcoding takes place here:
I action Originate has param "Codecs". I think it helps.
>
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+ManagerAction_Originate
>
> 22.11.2017 13:24, Kseniya Blashchuk пишет:
>
> Hi all!
>
>
> Asterisk 13.1.0 Ubuntu 16.04, all latest.
> Can anybod
This thread is not the point how to create call files and not about manager
commands.
I am asking why naive codec is different for channel Local with cmd
Originate when comparing to the same action in a call file. Call files work
for me in this way. If it doesn't work for you, maybe you have a diff
s: (slin)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: No
ReadTranscode: No
Please note - I do not use any manager API
Can anybody explain how the native format is chosen in these cases?
On Wed, Nov 22, 2017, 9:33 PM Kseniya Blashchuk wrote:
> This thread is not the point how to
Hmm thanks, I guess I should try the latest version just to check.
Unfortunately Ubuntu asterisk is not so frequently updated, just backports
on security updates
On Wed, Nov 22, 2017, 10:29 PM Richard Mudgett wrote:
> On Wed, Nov 22, 2017 at 12:38 PM, Kseniya Blashchuk
> wrote:
>
Hi all!
Does anybody have experience with asterisk on Hyper-V? My test setup with
Ubuntu 16 and asterisk 13.1 (ubuntu repo) shows sound distortion. I have
analyzed the RTP flow with wireshark and I see high skew and delta values
when the traffic leaves the hypervisor, however everything is okay whe
check your Linux Kernel version - it must be 3.10 or newer. I saw
> very bad "timing test" results on kernel 2.6.32.
>
> On 12/18/2017 10:26 AM, Kseniya Blashchuk wrote:
>
> Hi all!
> Does anybody have experience with asterisk on Hyper-V? My test setup with
> Ubuntu 16 and
Dmitry, are you using CentOS? What kernel version are you using? I will try
with the same to see if it can be also a kernel-related issue.
пн, 18 дек. 2017 г. в 11:35, Kseniya Blashchuk :
> Thank you for a quick answer, Dmitry!
>
> We have tried the settings you suggested but nothing he
resources to give them to Asterisk VM.
>
> Kseniya, could you try to reserve 100% of virtual CPUs for Asterisk VM
> (just to test this configuration)?
>
>
> I am sorry I don't have free hardware and time to test your Ubuntu 16,
> asterisk 13.1, kernel 4.4.0-104 configura
No problem, I will try to test with CentOS if I have time
пн, 18 дек. 2017 г. в 13:33, Dmitriy Ermakov :
> Ok. I am sorry, I don't have any more ideas(
>
> Please, send here email with your testing results.
>
> On 12/18/2017 01:22 PM, Kseniya Blashchuk wrote:
>
> To be
.
ср, 20 дек. 2017 г. в 14:42, Tzafrir Cohen :
> On Mon, Dec 18, 2017 at 10:22:33AM +, Kseniya Blashchuk wrote:
> > To be honest we are a bit afraid to set 100% )), but we have tried to set
> > 90% - no luck. I have also tested with 4.8 and 4.11 kernels - same
> results.
>
Hi all!
Does anybody know if it's possible to completely disable all asterisk call
features (even the default ones like xfer)?
Thanks in advance
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out
Ah seems I can just unload res_features.so ))
On Thu, Jan 11, 2018, 4:56 PM Kseniya Blashchuk wrote:
> Hi all!
> Does anybody know if it's possible to completely disable all asterisk call
> features (even the default ones like xfer)?
>
Well it seems that now it's not a module but a part of the kernel, however
if anybody knows how to completely turn the features off, please tell )
чт, 11 янв. 2018 г. в 18:14, Kseniya Blashchuk :
> Ah seems I can just unload res_features.so ))
>
> On Thu, Jan 11, 2018, 4:56 PM Kse
Finally I have accomplished what I needed it with options for Dial
application.
пт, 12 янв. 2018 г. в 12:48, Kseniya Blashchuk :
> Well it seems that now it's not a module but a part of the kernel, however
> if anybody knows how to completely turn the features off, please tell )
>
Hi!
I have used this document
https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport
You can specify transport=tls and encryption=yes for those peers which need
to use encryption.
пн, 5 мар. 2018 г. в 14:20, Antony Stone <
antony.st...@asterisk.open.source.it>:
> On Monday 05 March 2018 at
You can probably manage this with dial options (T or t for ex)
On Tue, Apr 17, 2018, 4:22 PM Joshua Colp wrote:
> On Fri, Apr 13, 2018, at 6:09 PM, Andrzej Nowrot wrote:
> > Hi
> >
> > Is there a way to disable blind and attended transfer during a call.
>
> No, DTMF features are not call time co
Try to set fromuser=number in your sip provider peer configuration
On Tue, May 8, 2018, 11:05 PM Jeff LaCoursiere wrote:
>
> Thats till doesn't change the SIP header. Basically they want to send a
> RE INVITE and authenticate my DID number. But my DID number does not have
> a peer or user entr
Hello!
I have a Polycom phone and sometimes I need to transfer calls without
picking them up to local extensions. Polycom has a transfer button which
sends SIP 302 packet to asterisk. Another local extension, receiving the
call, sees not the original number but the local number that was
transferrin
ut picking them up" is a blind transfer?
>
>
> Antony.
>
> > On Tue, Jun 25, 2019 at 8:41 AM Kseniya Blashchuk wrote:
> > > Hello!
> > > I have a Polycom phone and sometimes I need to transfer calls without
> > > picking them up to local extensions. Po
Thanks for trying, what asterisk version do you use?
вт, 25 июн. 2019 г. в 17:50, Doug Lytle :
> We have Polycom phones (I'm using a VVX601, the destination is a VVX301).
> We're also on Asterisk 13.
>
> I forwarded my call to the VVX301 and then dialed my phones DID. The
> forwarded call showed
This is what is actually going on:
Call is made to test-peer from number 123456789
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4baae0d7;rport
From: "Empty" ;tag=as24ef1afd
To: "Test Peer" ;tag=93AFFFD9-7DF89662
CSeq: 102 INVITE
Call-ID: 6143ff1e2dc860f04ebf7dc518fcb00d@1
Lol, everything was too simple. It was just a macro with app Dial with 'f'
option configured. Normally I don't use 'f', so I haven't checked that :)
вт, 25 июн. 2019 г. в 19:05, Doug Lytle :
> core show version
>
> Asterisk 13.26.0 built by doug @ asterisk on a x86_64 running Linux on
> 2019-04-0
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