similar to: 302 moved temporally callerid behavior

On Tuesday 25 June 2019 at 16:49:23, Doug Lytle wrote: > 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. Surely that is "call forwarding", which is quite different from either a blind or attended transfer? A transfer involves a call coming

Thanks for trying, what asterisk version do you use? вт, 25 июн. 2019 г. в 17:50, Doug Lytle <support at drdos.info>: > 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.

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

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

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

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 <gmludo at gmail.com> wrote: > Hi Kseniya, > > You might test with chan_pjsip: We have less production experience with > chan_pjsip than

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 :(. ??, 18 ???. 2017 ?. ? 10:49, Dmitriy Ermakov <demonihin at

Ok, thank you for the assistance! ??, 13 ???. 2017 ?. ? 16:38, Joshua Colp <jcolp at 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

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180111/8964a156/attachment.html>

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 > <mydestip>.5061 > If it was a system/kernel issue, then why is the behavior different for

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? ??, 20 ???. 2017 ?. ? 19:46, Carlos Chavez <cursor at 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

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

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? ??, 13 ???. 2017 ?. ? 14:52, Kseniya Blashchuk <ksyblast at gmail.com>: > Ah ok, thank you for checking. > I'll maybe also try with the latest asterisk and/or other distro and

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

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

On Mon, Mar 13, 2017, at 08:31 AM, Kseniya Blashchuk wrote: > Yes, look: > netstat -nlp | egrep '506[01]' > tcp 0 0 192.168.0.177:5061 0.0.0.0:* > LISTEN > 13255/asterisk > udp 0 0 192.168.0.177:5060 0.0.0.0:* > 13255/asterisk > Still, the problem is with *outgoing* *TCP* packets originated from > asterisk.

Again - when Originate is run from dialplan, i get: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin at 8000)->(slin at 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

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.Stone at asterisk.open.source.it> wrote: > On

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 <support at telium.ca> wrote: > Keep in mind that the attacks you are seeing in the log are ONLY the

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