similar to: Empty SIP Packet

Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can anyone imagine a scenario when enabling this parameter (even for peers that don't require it) can cause problems? Regards and thanks in advance, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there ..... ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List, I think I've discovered a little bug in t.38 bug in 1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes. Asterisk always responds with a=T38MaxBitRate:2400. I've tried with Patton and Grandstream devices and the result is always the same. Patton ignores the parameter and sends the fax at 9600.

Hello list, I need a hand to find the best dialplan failover solution when using two SIP Trunks. My reasons to do failover are: a) one of the two providers could be unreachable b) both providers may be UP but one of them could return a SIP error message (maybe caused by DOWN E1s) Googling I found a few possible solutions: 1.

Works for me.... Thanks, Hristo Benev -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc Sent: Monday, May 17, 2010 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration

Hello. I'm having a strange problem with the IAX2 channel and IAXmodem and I was hoping to get some light from someone in these lists. On my logs and on the console I'm getting a bunch of lines with: [Oct 23 14:26:18] NOTICE[4417] chan_iax2.c: Peer 'XXX' is now UNREACHABLE! Time: 3 [Oct 23 14:26:28] NOTICE[4413] chan_iax2.c: Peer 'XXX' is now REACHABLE!

Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I've tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex -------------- next part -------------- An HTML

Hello list! I'm having a strange problem with the VoIP Gateway that I'm using to go on the PSTN: if the number on the other end is busy or unavailable I hear an initial RING, generated by Asterisk from what I'm seeing and after that the line goes down with busy signal: Here is the scenario: Softphone *ASTERISK

Hello list! I've run into a strange problem today and I was hoping that someone here has seen this before and maybe can give me a hand: I'm using asterisk 1.6.0.22 in this config: (A)PATTON ISDN ->(B) ASTERISK -> (C)PATTON PRI -> PSTN -> (D)OTHER PBX Strange Problem: USER A calls makes a call to a PBX over the PSTN and ends into an IVR. When the user makes a selection and

Hi, Phone vendors (Snom, Thomson-Technicolor, ...) are on the way to support TR-069 (see http://en.wikipedia.org/wiki/TR-069). Has someone experienced with TR-069 ? What do you think of this protocol set ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100216/2cbda4a8/attachment.htm

What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. Yuan Liu

I have both the PJSIP add and the chan_sip way of adding SIP headers in there. The Verbose is showing the variable value is there. The INVITE to PJSIP/Agent1 does not include either X-My-DNID or X-My-DNID2 headers. exten => 1234,1,Verbose(X-My-DNID:${MY_DNID}) same => n,Set(X-My-DNID=${MY_DNID}) same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID}) same => n,Dial(PJSIP/Agent1)

Hello list, I have nat=no and qualify=no in my sip peer definition and still my CLI is flooded with : [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms / 2000ms) [Mar 12 10:17:26]

Hi. I configured icecast to serve a stream over ssl and it plays nice in Chrome and VLC. But on plain http I get *connection reset*. This stream's url is the same for many years now, and it is included in many radio directories, TuneIn and other apps. I can't change it or break it. Is there a way to have a stream that servs the content over ssl and plain http in the same time, depending

I had similar problems when my auth.php was on password protected http server...but after applying the following configuration i've got it working: <mount> <mount-name>/Test</mount-name> <authentication type="url"> <option name="listener_add" value="http://user:pass@127.0.0.1/auth/action.php"/> <option

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi All, Does anyone know, if there is a tool, which is doing the follwing: - Testprogram on host A establishes a sip connection to testprogramm on host B - Testprogram on host A plays a tone and Testprogram B verifies, if tone is playing correctly (without any interruptions) Thank You.

Marius Hofert-4------------------------------ > Den 2015-10-09 kl. 12:14, skrev Martin Maechler: > I think so: the code above doesn't seem to do the right thing. Consider > the following example: > > > x <- c(1, 1, 2, 3) > > rank2(x, ties.method = "last") > [1] 1 2 4 3 > > That doesn't look right to me -- I had expected > > >

On Fri, Jul 6, 2018 at 8:18 AM G?ran Brostr?m <goran.brostrom at umu.se> wrote: > > > > Den 2018-07-06 kl. 16:28, skrev Dirk Eddelbuettel: > > > > On 6 July 2018 at 12:31, Enrico Schumann wrote: | Just as one more > > datapoint: I cannot reproduce the segfault, with | R 3.5.1 on > > (L)Ubuntu 18.04. (I use the Ubuntu package, i.e. I did not | build >

m?ndag 4 maj 2020 kl. 20:45:37 CEST skrev Rowland penny via samba: > On 04/05/2020 19:24, Magnus Holmgren via samba wrote: > > Sunday 3 maj 2020 kl. 13:14:24 CEST, Rowland penny via samba wrote: > >> As for 'systemd', not sure what this actually does, but when I am forced > >> to use systemd (e.g. on my rpi), everything works even though I remove > >>