Re: [asterisk-users] IAX2 getting stuck

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

Re: [asterisk-users] IAX2 getting stuck

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.

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

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? > > >

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

ect 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 kn

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

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 Kseniya Blashc

Re: [asterisk-users] IAX2 getting stuck

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

[asterisk-users] tcpbind and source IP address

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

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

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 o

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

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,

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

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 '5

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

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

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

lp <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 >

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

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

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

17, 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

Re: [asterisk-users] Detecting DoS attacks via SIP

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

Re: [asterisk-users] Detecting DoS attacks via SIP

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

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

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

Re: [asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)

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

[asterisk-users] operator panel

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

[asterisk-users] Chan Local, Originate and slin

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:

Re: [asterisk-users] Chan Local, Originate and slin

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

Re: [asterisk-users] Chan Local, Originate and slin

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

Re: [asterisk-users] Chan Local, Originate and slin

lin) 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: >

Re: [asterisk-users] Chan Local, Originate and slin

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

[asterisk-users] asterisk and Hyper-V

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

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

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

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

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 setting

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

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 configu

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

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, Ks

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

, 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 ker

Re: [asterisk-users] multi step auth?

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

[asterisk-users] disable call features

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

Re: [asterisk-users] disable call features

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)? >

Re: [asterisk-users] disable call features

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

Re: [asterisk-users] disable call features

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

Re: [asterisk-users] Disable blind and attended transfer during call

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

Re: [asterisk-users] Asterisk server as TLS/SRTP

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

[asterisk-users] 302 moved temporally callerid behavior

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

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

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

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

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

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

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:

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

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 >