similar to: congested/busy on trunk?

ive enabled logging. aside from a realm error i see on my endpoint, im still not sure whats up Asterisk GIT-master-0cde95ec89, Copyright (C) 1999 - 2018, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General

On Sat, Mar 14, 2020 at 2:02 PM John Roman <john at dev1ce.com> wrote: > greetings asterisk users :) > ive just deployed version 17 and migrated as best I can to pjsip. I can > receive calls, and get to my mailbox prompt, however placing calls seems > impossible with the following error on dial: > > Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel

Hi, i upgraded from Debian sarge to etch, so from samba 3.0.14a-3 to 3.0.24 When someone creates/modify a file the user attribut is set to r (instead of rwx) so the next time a user opens the file is's on read only. What can i do so that the file keeps the attribut rwx for the owner ? Another question : the file has the rigthts : -r--rwx--- but although the user is member of the group

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c:

Hello! Looks like faq, but... Could you , please, point me on how to convert this [cisco] type=friend host=192.168.22.253 insecure=port,invite to pjsip? as you can see another side is very old cisco router, so I can't change anything there. I don't see any examples here

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

If understand correctly type=identify is more for sip trunk configuration ? ;[mytrunk] ;type=identify ;endpoint=mytrunk ;match=198.51.100.1 ;match=198.51.100.2 In chan_sip it was just reply 200 OK on keepalive packet without need define trunks. volga629 On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:42 AM, volga629 at

Hello guys, I'm at my wits end.. hope someone can help.. this is the deal. I have setup a linux box to serve as a "gateway" for our internal lan. Running 2.0.34 slakware. I have 2 nic cards.. One for our "internal" lan (192.168.2.0) and the "external" lan..I have gotten the OS to see both of them, but when trying to get DHCP to work with both, it kept barfing..

Hello, dovecot-1.0-test32 continuously fails on one of our customers mboxes with the following error: Sep 11 16:27:15 web dovecot: POP3(xxxxxxxxxxxxx.de1): file mbox-sync-rewrite.c: line 360 (mbox_sync_read_and_move): assertion failed: (need_space == -mails[idx].space) About 200 other mboxes are working perfectly on the same server. Where should I start looking for the cause of this?

I have SIP (asterisk 11.23.0) running on port 5060 just fine. udp. I have another SIP trunk thats wants to run on port 5068 (long story). I have enabled tcpenable=yes in sip.conf and defined port=5068 in my trunk definition. It does not seem that anything is listening on 5068? How can I run SIP tcp on port 5068? telnet localhost 5068 Trying 127.0.0.1... telnet: connect to address 127.0.0.1:

Right now it reply 401 Unauthorized with message in log "No matching endpoint ..." on Content 0 should reply 200 OK I guess <--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 ---> OPTIONS sip:10.30.100.27:5080 SIP/2.0 Via: SIP/2.0/UDP 10.30.100.41;branch=z9hG4bKf5eb.1ac76487000000000000000000000000.0 To: <sip:10.30.100.27:5080> From: <sip:vprx00 at

Hi, during our usual work I found it anoying that one can not easily see who logged in using public key authentication. In newer versions of SSH the fingerprint of the public key gets logged, but who can tell which key belongs to whom from his head? So I wrote a little ad-hoc patch (vs. 3.5.p1) so that the comment field on the keys in the authorized_keys[2] files get logged to make life

Hi folks, AFAIK CentOS uses RedHat's source RPMs for building the next CentOS release. I am not sure about the bootstrap procedure and the infra- structure packages, so lets put these corner cases aside. RedHat's "regular" binary and source packages are based on open source (GPL2, GPL3, Apache license, whatever). For building the binary RPMs other open source RPMs with

Hi folks, I've got a moderate number of keys in my ssh config file. Problem: Very often I get an error message like Received disconnect from 2001:db8::8077 port 999:2: Too many authentication failures Authentication failed. AFAIU the ssh-agent is to blame here, trying out all keys he has ever seen. This conflicts with MaxAuthTries 6, set by default on the peer. The solution seems to be to

Hello! There's the "g"-option for the Dial-cmd that allows to execute the next extensions in the current context when the callee hangs up. I would need the same for a call where the caller hangs up, concretely i have to inform a agi-application of the end of a call. Does someone know a way to do this from the dialplan? thanks Christian

Hi folks, I have configured the imap folder separator to '/' (see below). AFAIU the separator character does not affect the internal folder representation managed by Dovecot, but just the communication between client and imap server. Question is: What is supposed to happen if I try to create a folder named "a.b" in this configuration? Since '.' is not a separator