Re: [asterisk-users] 302 moved temporally callerid behavior

[Kseniya Blashchuk]

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
> 2019-04-05 11:41:43 UTC
>
> Built from source,
>
> Douh
>
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Re: [asterisk-users] 302 moved temporally callerid behavior

[Kseniya Blashchuk]

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@1.2.3.4:5060
 Contact: 
 User-Agent: PolycomSoundPointIP-SPIP_650-UA/4.0.11.0583
 Allow-Events: conference,talk,hold
 Accept-Language: en
 Content-Length: 0

Polycom redirects it to number 

 SIP/2.0 302 Moved Temporarily
 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@1.2.3.4:5060
 Contact: 
 User-Agent: PolycomSoundPointIP-SPIP_650-UA/4.0.11.0583
 Accept-Language: en
 Diversion: "Test Peer" ;reason=deflection
 Content-Length: 0

I would like that the peer at number  is receiving the real number
123456789, but it is receiving test-peer internal number.

вт, 25 июн. 2019 г. в 18:05, Doug Lytle :

> >>> Surely that is "call forwarding", which is quite different from either
> a blind or attended transfer?
>
> That would be correct.
>
> The forward button on the polycom phones just do a redirect to the
> destination extension or external phone number.
>
> Doug
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Re: [asterisk-users] 302 moved temporally callerid behavior

[Kseniya Blashchuk]

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 my cell phone number, so I cannot reproduce.
>
> Doug
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Re: [asterisk-users] 302 moved temporally callerid behavior

[Kseniya Blashchuk]

The call is not actually picked up, there is a "Forward" button on the
phone. After pressing it the phone sends a 302 Moved Temporally to asterisk
and the call goes to another extension. I guess attended transfer is
something else. Anyway, how is it connected with transferring the real
callerid?

вт, 25 июн. 2019 г. в 17:35, Antony Stone <
antony.st...@asterisk.open.source.it>:

> On Tuesday 25 June 2019 at 15:06:55, Dovid Bender wrote:
>
> > Your doing an attended transfer what you want to do is a blind transfer.
>
> Surely "transfer calls without 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. 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
> > > transferring the call. I would like that the original number is shown.
> I
> > > am stuck at this point.
> > > I see messages like "Not accepting call completion offers from
> > > call-forward recipient" in the logs but I'm not sure if it's somehow
> > > related to the problem.
> > > Can anybody help?
> > > Asterisk 13.1.0 Ubuntu 16
>
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[asterisk-users] 302 moved temporally callerid behavior

[Kseniya Blashchuk]

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
transferring the call. I would like that the original number is shown. I am
stuck at this point.
I see messages like "Not accepting call completion offers from call-forward
recipient" in the logs but I'm not sure if it's somehow related to the
problem.
Can anybody help?
Asterisk 13.1.0 Ubuntu 16
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Re: [asterisk-users] multi step auth?

[Kseniya Blashchuk]

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 entry in sip.conf.  Perhaps I am answering my own question,
> but is that the only way this is going to work?
>
> Thanks,
>
> j
>
>
> On 05/08/2018 02:54 PM, Khalil Khamlichi wrote:
>
> try adding a + sign for the number
>
> same => n,Set(CALLERID(all)=17864089672 <+17864089672>)
>
>
>
>
> On Tue, May 8, 2018, 8:51 PM Jeff LaCoursiere 
> wrote:
>
>>
>> I *am* doing that, as I assumed it would be required just for the 911
>> mapping we have provided, but that doesn't change the SIP header.
>>
>> Cheers,
>>
>> j
>>
>> On 05/08/2018 02:41 PM, Khalil Khamlichi wrote:
>>
>> try setting the callerid with
>>
>> same => n,Set(CALLERID(all)=17864089672 <17864089672>)
>>
>> ofcourse for each customer you will need to provide his own did.
>>
>>
>> On Tue, May 8, 2018, 8:37 PM Jeff LaCoursiere 
>> wrote:
>>
>>> Hi,
>>>
>>> We have been using Voxbone for some time for origination, and they now
>>> offer E911 services.  We are trying to set this up and having trouble
>>> meeting their authentication requirements.
>>>
>>> I setup a peer as I normally would, with user/pass as they supplied
>>> ("lacoursj", "pass"), but my calls are rejected.  Their support is asking
>>> that I follow this auth mechanism:
>>>
>>> 1st step - You send an INVITE message.
>>> 2nd step - We respond with a 407.
>>> 3rd step - You send a RE INVITE message including your credentials.
>>>
>>>  The tricky bit seems to be that they want the original INVITE to look
>>> like:
>>>
>>> From: ;tag=as00771983.
>>> To:  .
>>> Contact: .
>>>
>>> The "1786..." above is meant to be the DID number that is placing the
>>> 911 call. Our DID numbers don't have peer or user entries in sip.conf. My
>>> peer isn't sending that, though, it is sending:
>>>
>>> From: ;tag=as00771983.
>>> To:  .
>>> Contact: .
>>>
>>> They claim that 'lacoursj' shouldn't be sent until step 3.
>>>
>>> I have never been asked to authenticate this way... can asterisk
>>> chan_sip do it?
>>>
>>> Cheers,
>>> j
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>>
>>
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>
>
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Re: [asterisk-users] Disable blind and attended transfer during call

[Kseniya Blashchuk]

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 configurable. They are only grabbed
> when the channel is first bridged, not as they are potentially used.
>
> Cheers,
>
> --
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> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Asterisk server as TLS/SRTP

[Kseniya Blashchuk]

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 12:06:51, Atux Atux wrote:
>
> > Hi. I have an Asterisk Server (A) where it acts as the main gateway to
> > offer services.
> > There are different asterisk servers (B -D) that connect as extensions to
> > the Server A.
>
> Why not use IAX?
>
> > I would like to implement TLS and SRTP for these extensions, but have the
> > non secure as well for other extensions.
> > for example the extensions 4500-4504 be with TLS/SRTP and the rest be non
> > secure(ordinary).
> > Is there a guide on how to implement that please?
>
> How about
> https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
> or
> https://wiki.vpsget.com/index.php/Asterisk_11_with_TLS_and_SRTP_on_Centos_6
>
> > I am running asterisk 11.
>
> TLS has been available since 1.6 and SRTP since 1.8, so 11 should have no
> problems.
>
>
> Regards,
>
>
> Antony.
>
> --
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>
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Re: [asterisk-users] disable call features

[Kseniya Blashchuk]

Finally I have accomplished what I needed it with options for Dial
application.

пт, 12 янв. 2018 г. в 12:48, Kseniya Blashchuk <ksybl...@gmail.com>:

> 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 <ksybl...@gmail.com>:
>
>> Ah seems I can just unload res_features.so ))
>>
>> On Thu, Jan 11, 2018, 4:56 PM Kseniya Blashchuk <ksybl...@gmail.com>
>> wrote:
>>
>>> 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
>>>
>>
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Re: [asterisk-users] disable call features

[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 )

чт, 11 янв. 2018 г. в 18:14, Kseniya Blashchuk <ksybl...@gmail.com>:

> Ah seems I can just unload res_features.so ))
>
> On Thu, Jan 11, 2018, 4:56 PM Kseniya Blashchuk <ksybl...@gmail.com>
> wrote:
>
>> 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
>>
>
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Re: [asterisk-users] disable call features

[Kseniya Blashchuk]

Ah seems I can just unload res_features.so ))

On Thu, Jan 11, 2018, 4:56 PM Kseniya Blashchuk <ksybl...@gmail.com> wrote:

> 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
>
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[asterisk-users] disable call features

[Kseniya Blashchuk]

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
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Re: [asterisk-users] asterisk and Hyper-V

[Kseniya Blashchuk]

I am not sure if the problem is within the system or within a hypervisor or
a vswitch. My test is quite simple - when the traffic is captured from a VM
interface, everything is fine, and there is no sound distortion; but when
the traffic is captured from the hypervisor host interface, wireshark shows
crazy delta and skew values for RTP stream, and the sound is already
distorted. The host and the vswitch are not loaded, so this makes me think
that the clock differs somehow on the VM or something like that.
Good idea about recording - I will try. I'll try with lowlatency kernel as
well.

ср, 20 дек. 2017 г. в 14:42, Tzafrir Cohen <tzafrir.co...@xorcom.com>:

> 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.
> > I will try with Centos 6 and kernel 3.10 to check if something changes.
> > VSwitch shows 1-2% load on the interfaces, and this host is not
> overloaded
> > at all, so I don't think the VM has some lack of resources.
>
> What indication do you have that the problem is with the kernel or
> within the system?
>
> If you call from Asterisk to itself (with no networking involved), is
> there still distortion?
>
> Consider making a conference of several local channels (Echo, Playback,
> and whatever), and record whatever channel.
>
> Ubuntu has a "lowlatency" kernel. Does it matter if you use that
> variant?
>
> --
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> +972-50-7952406 <+972%2050-795-2406>   mailto:
> tzafrir.co...@xorcom.com
> http://www.xorcom.com
>
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Re: [asterisk-users] asterisk and Hyper-V

[Kseniya Blashchuk]

No problem, I will try to test with CentOS if I have time

пн, 18 дек. 2017 г. в 13:33, Dmitriy Ermakov <demoni...@gmail.com>:

> 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 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.
> I will try with Centos 6 and kernel 3.10 to check if something changes.
> VSwitch shows 1-2% load on the interfaces, and this host is not overloaded
> at all, so I don't think the VM has some lack of resources.
>
> пн, 18 дек. 2017 г. в 13:07, Dmitriy Ermakov <demoni...@gmail.com>:
>
>> I am using CentOS 6, kernel 3.10 from elrepo.org kernels
>> (3.10.102-1.el6.elrepo.x86_64).
>> Asterisk version is 11.21.2 and Asterisk 13.X.X (I can't get it's version
>> now).
>>
>> Is it possible that your network switches' interfaces which are connected
>> to Hyper-V Server are 100% busy?
>> It is possible that my installation works well because my Hyper-V server
>> is not high-load server so it has plenty free CPU, Disk, Memory, Network
>> 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 configuration(
>>
>> On 18 Dec 2017 12:09 p.m., "Kseniya Blashchuk" <ksybl...@gmail.com>
>> wrote:
>>
>>> 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 <ksybl...@gmail.com>:
>>>
>>>> Thank you for a quick answer, Dmitry!
>>>>
>>>> We have tried the settings you suggested but nothing helped. The
>>>> machine is running 4.4.0-104 kernel, 4 cores, Intel(R) Xeon(R) CPU E5-2620
>>>> v3 @ 2.40GHz, clocksource is hyperv_clocksource_tsc_page, timing module is
>>>> res_timing_timerfd.so. We have also tried to set 50% Reserve - no luck :(.
>>>>
>>>> пн, 18 дек. 2017 г. в 10:49, Dmitriy Ermakov <demoni...@gmail.com>:
>>>>
>>>>> Hi, Kseniya!
>>>>>
>>>>> I have three installations of Asterisk (as FreePBX but I think it is
>>>>> not important). They work fine.
>>>>>
>>>>> I have made some settings in Asterisk and Hyper-V:
>>>>>
>>>>> Asterisk: timing interface - timerfd.
>>>>>
>>>>> Hyper-V:
>>>>>
>>>>>1. Virtual Machine => Network Adapter => Hardware Acceleration =>
>>>>>Virtual Machine Queue - Disable it;
>>>>>    2. Virtual Machine => Processor => Virtual Machine Reserve
>>>>>(percentage) - set at least 25% (if you have 4 virtual cores for your
>>>>>Asterisk). You can try to set reserve more or less then 25% - test it 
>>>>> and,
>>>>>please, send email here;
>>>>>3. Virtual Machine => Integration Services => Time synchronisation
>>>>>- enable it.
>>>>>
>>>>> This settings helped me.
>>>>>
>>>>> Also 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 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
>>>>> when a capture is taken from a VM itself. I have read that there can be
>>>>> timing problems with Hyper-V. I have tried to disable time sync with the
>>>>> machine and tried different clocksources. I have also tried to change
>>>>> asterisk timing interface to dahdi (dummy) - nothing helped so far. 
>>>>> Hyper-V
>>>>> version is 12p2.
>>>>> Does anybody have a working setup with Hyper-V?
>>>>>
>>>>>
&

Re: [asterisk-users] asterisk and Hyper-V

[Kseniya Blashchuk]

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.
I will try with Centos 6 and kernel 3.10 to check if something changes.
VSwitch shows 1-2% load on the interfaces, and this host is not overloaded
at all, so I don't think the VM has some lack of resources.

пн, 18 дек. 2017 г. в 13:07, Dmitriy Ermakov <demoni...@gmail.com>:

> I am using CentOS 6, kernel 3.10 from elrepo.org kernels
> (3.10.102-1.el6.elrepo.x86_64).
> Asterisk version is 11.21.2 and Asterisk 13.X.X (I can't get it's version
> now).
>
> Is it possible that your network switches' interfaces which are connected
> to Hyper-V Server are 100% busy?
> It is possible that my installation works well because my Hyper-V server
> is not high-load server so it has plenty free CPU, Disk, Memory, Network
> 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 configuration(
>
> On 18 Dec 2017 12:09 p.m., "Kseniya Blashchuk" <ksybl...@gmail.com> wrote:
>
>> 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 <ksybl...@gmail.com>:
>>
>>> Thank you for a quick answer, Dmitry!
>>>
>>> We have tried the settings you suggested but nothing helped. The machine
>>> is running 4.4.0-104 kernel, 4 cores, Intel(R) Xeon(R) CPU E5-2620 v3 @
>>> 2.40GHz, clocksource is hyperv_clocksource_tsc_page, timing module is
>>> res_timing_timerfd.so. We have also tried to set 50% Reserve - no luck :(.
>>>
>>> пн, 18 дек. 2017 г. в 10:49, Dmitriy Ermakov <demoni...@gmail.com>:
>>>
>>>> Hi, Kseniya!
>>>>
>>>> I have three installations of Asterisk (as FreePBX but I think it is
>>>> not important). They work fine.
>>>>
>>>> I have made some settings in Asterisk and Hyper-V:
>>>>
>>>> Asterisk: timing interface - timerfd.
>>>>
>>>> Hyper-V:
>>>>
>>>>1. Virtual Machine => Network Adapter => Hardware Acceleration =>
>>>>Virtual Machine Queue - Disable it;
>>>>2. Virtual Machine => Processor => Virtual Machine Reserve
>>>>(percentage) - set at least 25% (if you have 4 virtual cores for your
>>>>Asterisk). You can try to set reserve more or less then 25% - test it 
>>>> and,
>>>>please, send email here;
>>>>3. Virtual Machine => Integration Services => Time synchronisation
>>>>- enable it.
>>>>
>>>> This settings helped me.
>>>>
>>>> Also 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 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
>>>> when a capture is taken from a VM itself. I have read that there can be
>>>> timing problems with Hyper-V. I have tried to disable time sync with the
>>>> machine and tried different clocksources. I have also tried to change
>>>> asterisk timing interface to dahdi (dummy) - nothing helped so far. Hyper-V
>>>> version is 12p2.
>>>> Does anybody have a working setup with Hyper-V?
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> С уважением, Дмитрий Ермаков
>>>>
>>>> --
>>>> _
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>
>>>> Check out the new Asterisk community forum at:
>>>> https://community.asterisk.org/
>>>>
>>>> New to Asterisk? Start here:
>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or

Re: [asterisk-users] asterisk and Hyper-V

[Kseniya Blashchuk]

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 <ksybl...@gmail.com>:

> Thank you for a quick answer, Dmitry!
>
> We have tried the settings you suggested but nothing helped. The machine
> is running 4.4.0-104 kernel, 4 cores, Intel(R) Xeon(R) CPU E5-2620 v3 @
> 2.40GHz, clocksource is hyperv_clocksource_tsc_page, timing module is
> res_timing_timerfd.so. We have also tried to set 50% Reserve - no luck :(.
>
> пн, 18 дек. 2017 г. в 10:49, Dmitriy Ermakov <demoni...@gmail.com>:
>
>> Hi, Kseniya!
>>
>> I have three installations of Asterisk (as FreePBX but I think it is not
>> important). They work fine.
>>
>> I have made some settings in Asterisk and Hyper-V:
>>
>> Asterisk: timing interface - timerfd.
>>
>> Hyper-V:
>>
>>1. Virtual Machine => Network Adapter => Hardware Acceleration =>
>>Virtual Machine Queue - Disable it;
>>2. Virtual Machine => Processor => Virtual Machine Reserve
>>(percentage) - set at least 25% (if you have 4 virtual cores for your
>>Asterisk). You can try to set reserve more or less then 25% - test it and,
>>please, send email here;
>>3. Virtual Machine => Integration Services => Time synchronisation -
>>enable it.
>>
>> This settings helped me.
>>
>> Also 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 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 when a
>> capture is taken from a VM itself. I have read that there can be timing
>> problems with Hyper-V. I have tried to disable time sync with the machine
>> and tried different clocksources. I have also tried to change asterisk
>> timing interface to dahdi (dummy) - nothing helped so far. Hyper-V version
>> is 12p2.
>> Does anybody have a working setup with Hyper-V?
>>
>>
>>
>>
>> --
>> С уважением, Дмитрий Ермаков
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] asterisk and Hyper-V

[Kseniya Blashchuk]

Thank you for a quick answer, Dmitry!

We have tried the settings you suggested but nothing helped. The machine is
running 4.4.0-104 kernel, 4 cores, Intel(R) Xeon(R) CPU E5-2620 v3 @
2.40GHz, clocksource is hyperv_clocksource_tsc_page, timing module is
res_timing_timerfd.so. We have also tried to set 50% Reserve - no luck :(.

пн, 18 дек. 2017 г. в 10:49, Dmitriy Ermakov <demoni...@gmail.com>:

> Hi, Kseniya!
>
> I have three installations of Asterisk (as FreePBX but I think it is not
> important). They work fine.
>
> I have made some settings in Asterisk and Hyper-V:
>
> Asterisk: timing interface - timerfd.
>
> Hyper-V:
>
>1. Virtual Machine => Network Adapter => Hardware Acceleration =>
>Virtual Machine Queue - Disable it;
>2. Virtual Machine => Processor => Virtual Machine Reserve
>(percentage) - set at least 25% (if you have 4 virtual cores for your
>Asterisk). You can try to set reserve more or less then 25% - test it and,
>please, send email here;
>3. Virtual Machine => Integration Services => Time synchronisation -
>enable it.
>
> This settings helped me.
>
> Also 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 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 when a
> capture is taken from a VM itself. I have read that there can be timing
> problems with Hyper-V. I have tried to disable time sync with the machine
> and tried different clocksources. I have also tried to change asterisk
> timing interface to dahdi (dummy) - nothing helped so far. Hyper-V version
> is 12p2.
> Does anybody have a working setup with Hyper-V?
>
>
>
>
> --
> С уважением, Дмитрий Ермаков
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] asterisk and Hyper-V

[Kseniya Blashchuk]

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 when a
capture is taken from a VM itself. I have read that there can be timing
problems with Hyper-V. I have tried to disable time sync with the machine
and tried different clocksources. I have also tried to change asterisk
timing interface to dahdi (dummy) - nothing helped so far. Hyper-V version
is 12p2.
Does anybody have a working setup with Hyper-V?
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Re: [asterisk-users] Chan Local, Originate and slin

[Kseniya Blashchuk]

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 <rmudg...@digium.com> wrote:

> On Wed, Nov 22, 2017 at 12:38 PM, Kseniya Blashchuk <ksybl...@gmail.com>
> wrote:
>
>> Again - when Originate is run from dialplan, i get:
>>
>> NativeFormats: (slin192)
>> WriteFormat: slin
>>  ReadFormat: slin192
>>  WriteTranscode: Yes (slin@8000)->(slin@192000)
>>   ReadTranscode: No
>>
>> When it's made with a call file (no matter how a call file is created), I
>> see
>>
>> NativeFormats: (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?
>>
> Version 13.1 is a very old version of Asterisk 13.  The current version of
> Asterisk 13 is 13.18.2.
> I also recall an issue where local channels tended to use slin192 when
> there was no need.  However,
> I'm unable to find the issue to know when that was done.
>
> Richard
>
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Re: [asterisk-users] Chan Local, Originate and slin

[Kseniya Blashchuk]

Again - when Originate is run from dialplan, i get:

NativeFormats: (slin192)
WriteFormat: slin
 ReadFormat: slin192
 WriteTranscode: Yes (slin@8000)->(slin@192000)
  ReadTranscode: No

When it's made with a call file (no matter how a call file is created), I
see

NativeFormats: (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 <ksybl...@gmail.com> wrote:

> 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 different
> OS or Asterisk version.
>
> On Wed, Nov 22, 2017, 8:02 PM Steve Edwards <asterisk@sedwards.com>
> wrote:
>
>> On Wed, 22 Nov 2017, Dmitriy Serov wrote:
>>
>> >  same => n,System(printf "Action: Originate\nActionID: 1\nChannel:
>> Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" >
>> > /var/spool/asterisk/outgoing/${number}-${confnum})
>>
>> I get:
>>
>> Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/...
>> Unknown keyword 'ActionID' at line 2 of /var/spool/asterisk/outgoing/...
>>
>> These are 'AMI' commands, not call file commands.
>>
>> Also, just in case you're not aware, 'best practice' is to create the call
>> file in a 'temp' directory on the same partition as
>> ${astspooldir}/outgoing/ and then 'mv' the file to that directory.
>>
>> --
>> Thanks in advance,
>> -
>> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>>  https://www.linkedin.com/in/steve-edwards-4244281--
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] Chan Local, Originate and slin

[Kseniya Blashchuk]

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 different
OS or Asterisk version.

On Wed, Nov 22, 2017, 8:02 PM Steve Edwards 
wrote:

> On Wed, 22 Nov 2017, Dmitriy Serov wrote:
>
> >  same => n,System(printf "Action: Originate\nActionID: 1\nChannel:
> Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" >
> > /var/spool/asterisk/outgoing/${number}-${confnum})
>
> I get:
>
> Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/...
> Unknown keyword 'ActionID' at line 2 of /var/spool/asterisk/outgoing/...
>
> These are 'AMI' commands, not call file commands.
>
> Also, just in case you're not aware, 'best practice' is to create the call
> file in a 'temp' directory on the same partition as
> ${astspooldir}/outgoing/ and then 'mv' the file to that directory.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>  https://www.linkedin.com/in/steve-edwards-4244281--
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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> https://community.asterisk.org/
>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Chan Local, Originate and slin

[Kseniya Blashchuk]

First, I cannot set codecs with cmd Originate from dialplan (or can I?).
Second - the difference in my example is only between cmd Originate from
dialplan and Originate from a call file. To my mind, it shouldn't make any
difference.

ср, 22 нояб. 2017 г. в 16:44, Dmitriy Serov <serov@gmail.com>:

> AMI 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 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:
>
>   NativeFormats: (slin192)
> WriteFormat: slin
>  ReadFormat: slin192
>  WriteTranscode: Yes (slin@8000)->(slin@192000)
>   ReadTranscode: No
>
> When I do the same from a call file like:
>
>  same => n,System(printf "Action: Originate\nActionID: 1\nChannel:
> Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" >
> /var/spool/asterisk/outgoing/${number}-${confnum})
>
> the sound is perfect and this is what I see on the channel params:
>
>   NativeFormats: (slin)
> WriteFormat: slin
>  ReadFormat: slin
>  WriteTranscode: No
>   ReadTranscode: No
>
> Can anybody explain what is going on?
>
>
>
>
> --
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>
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[asterisk-users] Chan Local, Originate and slin

[Kseniya Blashchuk]

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:

  NativeFormats: (slin192)
WriteFormat: slin
 ReadFormat: slin192
 WriteTranscode: Yes (slin@8000)->(slin@192000)
  ReadTranscode: No

When I do the same from a call file like:

 same => n,System(printf "Action: Originate\nActionID: 1\nChannel:
Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" >
/var/spool/asterisk/outgoing/${number}-${confnum})

the sound is perfect and this is what I see on the channel params:

  NativeFormats: (slin)
WriteFormat: slin
 ReadFormat: slin
 WriteTranscode: No
  ReadTranscode: No

Can anybody explain what is going on?
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Re: [asterisk-users] Detecting DoS attacks via SIP

[Kseniya Blashchuk]

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 in mind that the attacks you are seeing in the log are ONLY the ones
> that Asterisk is detecting and rejecting.  All other attacks aren't even
> showing up!
>
> There's a good discussion of how to secure your PBX here:
> https://www.voip-info.org/wiki/view/asterisk+security
>
> In general, don't let the malevolent traffic get as far as the PBX (block
> at
> the firewall).  Also, Digium regularly warns users that fail2ban is NOT a
> security system: http://forums.asterisk.org/viewtopic.php?p=159984
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mdiehl
> Sent: Tuesday, August 15, 2017 3:38 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Detecting DoS attacks via SIP
>
> Hi all,
>
> Lately, I've seen an increase in the number of attacks against my system
> from the so-called "Friendly Scanner."  When one of these script kiddies
> targets my server, all I see for symptoms is a few of my trunks become
> lagged due to server load and a stream of messages on the console that
> resemble this:
>
> [Aug  2 20:27:50]   == Using SIP VIDEO CoS mark 6
> [Aug  2 20:27:50]   == Using SIP RTP TOS bits 24
> [Aug  2 20:27:50]   == Using SIP RTP CoS mark 5
> [Aug  2 20:32:47]   == Using SIP VIDEO TOS bits 24
> [Aug  2 20:32:47]   == Using SIP VIDEO CoS mark 6
> [Aug  2 20:32:47]   == Using SIP RTP TOS bits 24
> [Aug  2 20:32:47]   == Using SIP RTP CoS mark 5
> [Aug  2 20:34:26]   == Using SIP VIDEO TOS bits 24
> [Aug  2 20:34:26]   == Using SIP VIDEO CoS mark 6
>
>
> I have to turn on sip debugging to find out who's hitting me.  However, I
> can't just leave it on because it would kill my logging system.
>
> So, how are other people handling this?  Is there an AMI event I want watch
> for?  I watch for PeerStatus, but since there's no actual peer in the
> attack, I don't seem to get an event from AMI.
>
> Any ideas?
>
> Mike Diehl.
>
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Re: [asterisk-users] Detecting DoS attacks via SIP

[Kseniya Blashchuk]

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 the so-called "Friendly Scanner."  When one of these script kiddies
> targets my server, all I see for symptoms is a few of my trunks become
> lagged due to server load and a stream of messages on the console that
> resemble this:
> [snip]
> > I have to turn on sip debugging to find out who's hitting me.  However,
> I can't just leave it on because it would kill my logging system.
> >
> > So, how are other people handling this?  Is there an AMI event I want
> watch for?  I watch for PeerStatus, but since there's no actual peer in the
> attack, I don't seem to get an event from AMI.
> >
> > Any ideas?
>
> You can block sipvicious/friendly scanner in iptables with something like:
>
> -A INPUT -p udp --dport 5060 -m string --string "friendly-scanner"
> --algo bm -j DROP
>
> You can also look at xtables with geoip to drop countries (per
> destination port) that should not connect to your Asterisk box. It's a
> big hammer but it works really well.
>
> Or put a proxy like Kamailio or OpenSIPS in front of the Asterisk box.
> That's what the telco's/service providers do.
>
> HTH,
> Patrick
>
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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[Kseniya Blashchuk]

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" address. they turned
> that on a couple years ago with no  notice.
>
> John Novack
>
>
> Jonathan H wrote:
>
> Me too, also gmail. I emailed the list owner a couple of days ago, but no 
> reply.
>
> Is everyone else affected also forwarding to another email address
> (gmail or not)?
>
> Could be wrong, but I'm guessing there may be an incorrect DMARC
> policy somewhere - although this is the only fail I could find in the
> headers.
> boun...@lists.digium.com;
>dmarc=fail (p=NONE sp=NONE dis=NONE) header.from=gmail.com
>
>
>
> On 12 June 2017 at 09:12, Steve Davies  
>  wrote:
>
> I am also getting this, three or four times in the last month after years of
> no problems.
>
> I agree that Gmail is the likely common factor, but I would love to have
> access to these bounce messages to know whether it is actually an
> overly-paranoid list server!
>
> Steve
>
> On Mon, 12 Jun 2017 at 09:09 Andrew Furey  
>  wrote:
>
>
> Ditto; a Gmail issue?
>
> Andrew
>
> On 12 June 2017 at 16:00, Marcelo Terres  
>  wrote:
>
>
> It is happening the same with me.
>
> Regards,
> Marcelo H. Terres  
> IM: 
> mhter...@jabber.mundoopensource.com.brhttps://www.mundoopensource.com.brhttps://twitter.com/mhterreshttps://linkedin.com/in/marceloterres
>
>
> On 12 June 2017 at 08:07, Olivier   
> wrote:
>
> Hello,
>
> I'm a faithful reader of this mailing list, for several years now.
>
> Lately, I'm receiving emails asking me to re-enable my list
> subscription due
> to "excessive bouncing".
>
> What does this exactly mean and why am I receiving this ?
> Beside re-enabling my subscription, what can I do to improve things ?
>
> Regards
>
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>
>
>
>
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> Language, be it English or something else, is the only tool flexible
> enough to accomplish a sufficiently broad range of tasks.
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Re: [asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)

[Kseniya Blashchuk]

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.
>
> This requires all our SIP Trunk customers to register via a 'proxy'.
>
> I struggle with Asterisk to work over a proxy.
>
> This is what I have done so far.
>
> register => usern...@sip.example.com:passw...@sbc.example.com
>
> This works fine, asterisk is sending registrations via the SBC to the
> voice switch defined by URI.
>
> [username]
> type=peer
> secret=password
> host=sip.example.com
> outboundproxy=sbc.example.com
> context=from-ISP-X
>
> From the Dialplan that string is dialed:
>
> Dial(SIP/username/${EXTEN})
>
> This works fine, asterisk sends the call to the outboundproxy defined
> in the sip.conf section of [username].
>
> Before adding outboundproxy setting, incomming calls were
> matched because they originated from the host and passed to the correct
> context.
>
> I have set allowguest=no to challenge all those sip attackers in
> [default] who occasionaly managed to call internal extensions defined
> there.
>
> Now incomming calls do not originate from the ip of sip.example.com
> anymore, but from the ip of sbc.example.com and are not set to the
> context [from-ISP-X] but probably to [default] and challenged.
>
> Of course, I could allow guests, but that would bring back the problem
> of having unwanted calls from sip scanners.
>
> So how do I tell the asterisk to also match calls from the ip of the
> outbound proxy?
>
> -Benoît Panizzon-
> --
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>
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[asterisk-users] operator panel

[Kseniya Blashchuk]

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?
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Re: [asterisk-users] tcpbind and source IP address

[Kseniya Blashchuk]

JFYI - https://issues.asterisk.org/jira/browse/ASTERISK-26922

чт, 20 апр. 2017 г. в 11:38, Kseniya Blashchuk <ksybl...@gmail.com>:

> 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 2:24, Ludovic Gasc <gml...@gmail.com>:
>
>> Hi,
>>
>> I recommend you to install from sources, especially because the latest
>> Asterisk 13 has several bugfixes for pjsip.
>> To my knowledge, nobody proposes ppa or Debian backports for Asterisk.
>>
>> Wazo has Debian packages, but it's only for Debian Jessie and with extra
>> patches for Wazo.
>>
>> --
>> Ludovic Gasc (GMLudo)
>> Lead Developer Architect at ALLOcloud
>> https://be.linkedin.com/in/ludovicgasc
>>
>> 2017-04-16 21:36 GMT+02:00 Kseniya Blashchuk <ksybl...@gmail.com>:
>>
>>> 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 try to make a test next days.
>>>
>>> On Sun, Apr 16, 2017, 8:18 PM Ludovic Gasc <gml...@gmail.com> wrote:
>>>
>>>> Hi Kseniya,
>>>>
>>>> You might test with chan_pjsip: We have less production experience with
>>>> chan_pjsip than chan_sip, however, for now, we are more and more confident
>>>> in this new stack while we're digging in documentation and we're testing on
>>>> production.
>>>>
>>>> However, I've no idea if you'll have the same 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 Kseniya Blashchuk <ksybl...@gmail.com>:
>>>>
>>>>> Ok, thank you for the assistance!
>>>>>
>>>>> пн, 13 мар. 2017 г. в 16:38, Joshua Colp <jc...@digium.com>:
>>>>>
>>>>>> 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. Absolutely the same
>>>>>> behavior.
>>>>>> > Joshua, maybe you can advice what can be done further?
>>>>>>
>>>>>> You can file an issue but chan_sip is a community supported module, so
>>>>>> there is no guarantee of when it would be looked at and resolved.
>>>>>> Ultimately though someone has to spend the time to replicate what is
>>>>>> going on, look into the code, and understand what is going on.
>>>>>>
>>>>>> --
>>>>>> Joshua Colp
>>>>>> Digium, Inc. | Senior Software Developer
>>>>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>>>>> Check us out at: www.digium.com & www.asterisk.org
>>>>>>
>>>>>> --
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>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>>
>>>>>> Check out the new Asterisk community forum at:
>>>>>> https://community.asterisk.org/
>>>>>>
>>>>>> New to Asterisk? Start here:
>>>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>>>
>>>>>> asterisk-users mailing list
>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>
>>>>>
>>>>> --
>>>>> _
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>>>>>
>>>>> Check out the new Asterisk community forum at:
>>>>> https://community.asterisk.org/
>>>>>
>>>>> New to Asterisk? Start here:
>>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>>
>>>>> asterisk-users m

Re: [asterisk-users] IAX2 getting stuck

[Kseniya Blashchuk]

Hmmm.. So if you are sure that the poke packets leave the network interface
(I would still check with tcpdump as well, maybe a firewall issue?) then it
makes sense to check the other side to make sure the poke packets reach
other servers.
I mean with tcpdump you may see if there are incoming packets from your
peers on the interface. If there are, then they are dropped or ignored by
your servers. If no, then it's better to check the other side.
you may try smth like 'tcpdump -npi  host  and port 4569'
Do you have a firewall configured on this server?

On Fri, Apr 21, 2017, 12:36 AM Carlos Chavez <cur...@telecomab.mx> 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 <cur...@telecomab.mx>:
>
>> On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:
>>
>> 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 20, 2017, 12:41 AM Antony Stone <
>> antony.st...@asterisk.open.source.it> wrote:
>>
>>> On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:
>>>
>>> > On 4/19/17 4:23 PM, Antony Stone wrote:
>>> > >
>>> > > You say the USB ethernet adapter got unplugged and then
>>> reconnected...
>>> > >
>>> > > 1. What's the name of the network device for this adapter?  Is it the
>>> > > same name as it previously had?
>>> > >
>>> > > 2. What does 'ifconfig' say the IP address is for this adapter?
>>> > >
>>> > > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
>>> > > 'bindport'?
>>> > >
>>> > > 4. Do you have SIP connections on the same network interface, and are
>>> > > those working as normal?
>>> > >
>>> > >
>>> > > Antony.
>>> >
>>> > 1- No changes to device names.  eth0 is the main link to the network,
>>> > eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
>>> > goes to another SIP provider.  All IAX trunks use eth0
>>> >
>>> > 2- ifconfig gives the proper IP and netmask for all interfaces
>>> >
>>> > 3- We do not specify bindaddr or bindport in the config file as the
>>> > default is to bind to 0.0.0.0
>>> >
>>> > 4- We had to make new SIP trunks to replace the IAX2 trunks to all
>>> > servers.  The SIP trunk is working with no problems.  Except for two
>>> SIP
>>> > links to PSTN all internal extensions use the same network interface.
>>>
>>> Ugh :(
>>>
>>> Sorry, I have no more ideas, then.
>>>
>>> I hope someone else comes into this thread with a helpful suggestion.
>>>
>>>
>>> If routing was the problem then the SIP trunk would not work.
>> Usually IAX2 is a little more forgiving about routing than SIP.
>>
>> The new SIP trunks are replacing the IAX2 trunks to our other
> Asterisk servers and use exactly the same network paths, that is why I know
> it is not a network infrastructure issue.  We did turn on IAX debug and we
> only se the server trying to poke the other servers but there is not
> response or any incoming traffic.
>
> --
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Re: [asterisk-users] IAX2 getting stuck

[Kseniya Blashchuk]

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 <cur...@telecomab.mx>:

> On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:
>
> 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 20, 2017, 12:41 AM Antony Stone <
> antony.st...@asterisk.open.source.it> wrote:
>
>> On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:
>>
>> > On 4/19/17 4:23 PM, Antony Stone wrote:
>> > >
>> > > You say the USB ethernet adapter got unplugged and then reconnected...
>> > >
>> > > 1. What's the name of the network device for this adapter?  Is it the
>> > > same name as it previously had?
>> > >
>> > > 2. What does 'ifconfig' say the IP address is for this adapter?
>> > >
>> > > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
>> > > 'bindport'?
>> > >
>> > > 4. Do you have SIP connections on the same network interface, and are
>> > > those working as normal?
>> > >
>> > >
>> > > Antony.
>> >
>> > 1- No changes to device names.  eth0 is the main link to the network,
>> > eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
>> > goes to another SIP provider.  All IAX trunks use eth0
>> >
>> > 2- ifconfig gives the proper IP and netmask for all interfaces
>> >
>> > 3- We do not specify bindaddr or bindport in the config file as the
>> > default is to bind to 0.0.0.0
>> >
>> > 4- We had to make new SIP trunks to replace the IAX2 trunks to all
>> > servers.  The SIP trunk is working with no problems.  Except for two SIP
>> > links to PSTN all internal extensions use the same network interface.
>>
>> Ugh :(
>>
>> Sorry, I have no more ideas, then.
>>
>> I hope someone else comes into this thread with a helpful suggestion.
>>
>>
>> If routing was the problem then the SIP trunk would not work.
> Usually IAX2 is a little more forgiving about routing than SIP.
>
> --
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> Carlos Chávez+52 (55)8116-9161 <+52%2055%208116%209161>
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Re: [asterisk-users] tcpbind and source IP address

[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 2:24, Ludovic Gasc <gml...@gmail.com>:

> Hi,
>
> I recommend you to install from sources, especially because the latest
> Asterisk 13 has several bugfixes for pjsip.
> To my knowledge, nobody proposes ppa or Debian backports for Asterisk.
>
> Wazo has Debian packages, but it's only for Debian Jessie and with extra
> patches for Wazo.
>
> --
> Ludovic Gasc (GMLudo)
> Lead Developer Architect at ALLOcloud
> https://be.linkedin.com/in/ludovicgasc
>
> 2017-04-16 21:36 GMT+02:00 Kseniya Blashchuk <ksybl...@gmail.com>:
>
>> 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
>> try to make a test next days.
>>
>> On Sun, Apr 16, 2017, 8:18 PM Ludovic Gasc <gml...@gmail.com> wrote:
>>
>>> Hi Kseniya,
>>>
>>> You might test with chan_pjsip: We have less production experience with
>>> chan_pjsip than chan_sip, however, for now, we are more and more confident
>>> in this new stack while we're digging in documentation and we're testing on
>>> production.
>>>
>>> However, I've no idea if you'll have the same 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 Kseniya Blashchuk <ksybl...@gmail.com>:
>>>
>>>> Ok, thank you for the assistance!
>>>>
>>>> пн, 13 мар. 2017 г. в 16:38, Joshua Colp <jc...@digium.com>:
>>>>
>>>>> 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. Absolutely the same
>>>>> behavior.
>>>>> > Joshua, maybe you can advice what can be done further?
>>>>>
>>>>> You can file an issue but chan_sip is a community supported module, so
>>>>> there is no guarantee of when it would be looked at and resolved.
>>>>> Ultimately though someone has to spend the time to replicate what is
>>>>> going on, look into the code, and understand what is going on.
>>>>>
>>>>> --
>>>>> Joshua Colp
>>>>> Digium, Inc. | Senior Software Developer
>>>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>>>> Check us out at: www.digium.com & www.asterisk.org
>>>>>
>>>>> --
>>>>> _
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>
>>>>> Check out the new Asterisk community forum at:
>>>>> https://community.asterisk.org/
>>>>>
>>>>> New to Asterisk? Start here:
>>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>> --
>>>> _
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>>>>
>>>> Check out the new Asterisk community forum at:
>>>> https://community.asterisk.org/
>>>>
>>>> New to Asterisk? Start here:
>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>
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>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>>>
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>>>
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>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   

Re: [asterisk-users] IAX2 getting stuck

[Kseniya Blashchuk]

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 20, 2017, 12:41 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:
>
> > On 4/19/17 4:23 PM, Antony Stone wrote:
> > >
> > > You say the USB ethernet adapter got unplugged and then reconnected...
> > >
> > > 1. What's the name of the network device for this adapter?  Is it the
> > > same name as it previously had?
> > >
> > > 2. What does 'ifconfig' say the IP address is for this adapter?
> > >
> > > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
> > > 'bindport'?
> > >
> > > 4. Do you have SIP connections on the same network interface, and are
> > > those working as normal?
> > >
> > >
> > > Antony.
> >
> > 1- No changes to device names.  eth0 is the main link to the network,
> > eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
> > goes to another SIP provider.  All IAX trunks use eth0
> >
> > 2- ifconfig gives the proper IP and netmask for all interfaces
> >
> > 3- We do not specify bindaddr or bindport in the config file as the
> > default is to bind to 0.0.0.0
> >
> > 4- We had to make new SIP trunks to replace the IAX2 trunks to all
> > servers.  The SIP trunk is working with no problems.  Except for two SIP
> > links to PSTN all internal extensions use the same network interface.
>
> Ugh :(
>
> Sorry, I have no more ideas, then.
>
> I hope someone else comes into this thread with a helpful suggestion.
>
>
> Antony.
>
> --
> The first fifty percent of an engineering project takes ninety percent of
> the
> time, and the remaining fifty percent takes another ninety percent of the
> time.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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>
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Re: [asterisk-users] tcpbind and source IP address

[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
try to make a test next days.

On Sun, Apr 16, 2017, 8:18 PM Ludovic Gasc <gml...@gmail.com> wrote:

> Hi Kseniya,
>
> You might test with chan_pjsip: We have less production experience with
> chan_pjsip than chan_sip, however, for now, we are more and more confident
> in this new stack while we're digging in documentation and we're testing on
> production.
>
> However, I've no idea if you'll have the same 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 Kseniya Blashchuk <ksybl...@gmail.com>:
>
>> Ok, thank you for the assistance!
>>
>> пн, 13 мар. 2017 г. в 16:38, Joshua Colp <jc...@digium.com>:
>>
>>> 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. Absolutely the same
>>> behavior.
>>> > Joshua, maybe you can advice what can be done further?
>>>
>>> You can file an issue but chan_sip is a community supported module, so
>>> there is no guarantee of when it would be looked at and resolved.
>>> Ultimately though someone has to spend the time to replicate what is
>>> going on, look into the code, and understand what is going on.
>>>
>>> --
>>> Joshua Colp
>>> Digium, Inc. | Senior Software Developer
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>> Check us out at: www.digium.com & www.asterisk.org
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
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>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] tcpbind and source IP address

[Kseniya Blashchuk]

Ok, thank you for the assistance!

пн, 13 мар. 2017 г. в 16:38, Joshua Colp <jc...@digium.com>:

> 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. Absolutely the same behavior.
> > Joshua, maybe you can advice what can be done further?
>
> You can file an issue but chan_sip is a community supported module, so
> there is no guarantee of when it would be looked at and resolved.
> Ultimately though someone has to spend the time to replicate what is
> going on, look into the code, and understand what is going on.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] tcpbind and source IP address

[Kseniya Blashchuk]

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 <ksybl...@gmail.com>:

> 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 <jc...@digium.com>:
>
> On Mon, Mar 13, 2017, at 08:43 AM, Kseniya Blashchuk wrote:
> > Mmh sorry I'm afraid I did not understand your last message. Yes the code
> > does that but only with UDP, for TCP the source address is 192.168.0.172
> > though it's bound to 192.168.0.177:
> > IP 192.168.0.172.47596 > .5061
> > If it was a system/kernel issue, then why is the behavior different for
> > TCP
> > and UDP? I thought that maybe the application does not request the bound
> > address as a source in case of TCP...
>
> The chan_sip module, from looking at the code, does use the bound
> address when connecting. Someone would need to dig deeper to understand
> if the problem is somehow in Asterisk or if it is the system somehow
> doing it.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] tcpbind and source IP address

[Kseniya Blashchuk]

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 <jc...@digium.com>:

> On Mon, Mar 13, 2017, at 08:43 AM, Kseniya Blashchuk wrote:
> > Mmh sorry I'm afraid I did not understand your last message. Yes the code
> > does that but only with UDP, for TCP the source address is 192.168.0.172
> > though it's bound to 192.168.0.177:
> > IP 192.168.0.172.47596 > .5061
> > If it was a system/kernel issue, then why is the behavior different for
> > TCP
> > and UDP? I thought that maybe the application does not request the bound
> > address as a source in case of TCP...
>
> The chan_sip module, from looking at the code, does use the bound
> address when connecting. Someone would need to dig deeper to understand
> if the problem is somehow in Asterisk or if it is the system somehow
> doing it.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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> https://community.asterisk.org/
>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] tcpbind and source IP address

[Kseniya Blashchuk]

Mmh sorry I'm afraid I did not understand your last message. Yes the code
does that but only with UDP, for TCP the source address is 192.168.0.172
though it's bound to 192.168.0.177:
IP 192.168.0.172.47596 > .5061
If it was a system/kernel issue, then why is the behavior different for 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 <jc...@digium.com>:

> On Mon, Mar 13, 2017, at 08:31 AM, Kseniya Blashchuk wrote:
> > Yes, look:
> > netstat -nlp | egrep '506[01]'
> > tcp0  0 192.168.0.177:5061  0.0.0.0:*
> > LISTEN
> >  13255/asterisk
> > udp0  0 192.168.0.177:5060  0.0.0.0:*
> > 13255/asterisk
> > Still, the problem is with *outgoing* *TCP* packets originated from
> > asterisk. Source IP is set to the first IP address of the interface only
> > when TCP is used. As I understand, the application (chan_sip in this
> > case)
> > should request kernel to use the specific source IP address (used in bind
> > directive) for outgoing packets, however it seems to be done only for
> > UDP.
> > For outgoing packets on TCP/5061 I see the following:
> > IP *192.168.0.172*.47596 > .5061: Flags [S], seq 2529313754
> <(252)%20931-3754>,
> > win
> > 29200, options [mss 1460,sackOK,TS val 82765588 ecr 0,nop,wscale 7],
> > length
> > 0
> > And with UDP as transport:
> > IP *192.168.0.177*.5060 > .5060: SIP: OPTIONS 
>
> The underlying code does this already. It connects using the bound
> socket (which would be bound to the IP address you've provided). This
> should have the system use the source IP address as you want, but it's
> evidently not.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> https://community.asterisk.org/
>
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Re: [asterisk-users] tcpbind and source IP address

[Kseniya Blashchuk]

Yes, look:
netstat -nlp | egrep '506[01]'
tcp0  0 192.168.0.177:5061  0.0.0.0:*   LISTEN
 13255/asterisk
udp0  0 192.168.0.177:5060  0.0.0.0:*
13255/asterisk
Still, the problem is with *outgoing* *TCP* packets originated from
asterisk. Source IP is set to the first IP address of the interface only
when TCP is used. As I understand, the application (chan_sip in this case)
should request kernel to use the specific source IP address (used in bind
directive) for outgoing packets, however it seems to be done only for UDP.
For outgoing packets on TCP/5061 I see the following:
IP *192.168.0.172*.47596 > .5061: Flags [S], seq 2529313754, win
29200, options [mss 1460,sackOK,TS val 82765588 ecr 0,nop,wscale 7], length
0
And with UDP as transport:
IP *192.168.0.177*.5060 > .5060: SIP: OPTIONS 




пн, 13 мар. 2017 г. в 13:55, Joshua Colp <jc...@digium.com>:

> 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 UDP 192.168.0.177 is used as expected.
>
> Does the output of netstat -a confirm that it is bound to only that IP
> address? If so, then it seems chan_sip has done its part.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
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>
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Re: [asterisk-users] tcpbind and source IP address

[Kseniya Blashchuk]

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 <jc...@digium.com>:

> On Sat, Mar 11, 2017, at 11:50 AM, Kseniya Blashchuk wrote:
> > Hey guys, any thoughts on that? Probably a bug or is it a default
> > behavior?
>
> I'd suggest providing the configuration to make sure it is correct.
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
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>
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config_sip
Description: Binary data


ip_a
Description: Binary data
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Re: [asterisk-users] tcpbind and source IP address

[Kseniya Blashchuk]

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 <ksybl...@gmail.com> 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 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 tells the kernel to use the specific "bind" address
> only in case of UDP usage (both SIP and IAX work like that). In case of
> outgoing TCP connections (SIP TCP and TLS) the first IP address from the
> interface is used.
> In my understanding, normally 'bind' should not only tell on which address
> to listen, but also which source address to request for outgoing
> connections, but it works only for UDP connections for some reason.
> Can anybody explain if it's a normal behavior?
>
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[asterisk-users] tcpbind and source IP address

[Kseniya Blashchuk]

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 tells the kernel to use the specific "bind" address
only in case of UDP usage (both SIP and IAX work like that). In case of
outgoing TCP connections (SIP TCP and TLS) the first IP address from the
interface is used.
In my understanding, normally 'bind' should not only tell on which address
to listen, but also which source address to request for outgoing
connections, but it works only for UDP connections for some reason.
Can anybody explain if it's a normal behavior?
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users