similar to: Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid="MaxTNT" <maxtnt> context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing

Hello, I'm currently testing Asterisk over a T1 cross connect to a MaxTNT chassis that we have. It is working fine switching the calls through, but there is about a 10 second delay from the time Asterisk initiates the call until the TNT accepts it. It appears to be a ANI issue, I've changed several settings and formatting options on the T1 between the two, as well as turning on/off the

Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past

Is there a way to dial DTMF after bridging the call. The current option D() in Dial will dial DTMF before the call is bridged and this doesn't do the job. I need to dial DTMF after the call is bridged and the message is played with "Background" -- #Joseph

I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

The port is determined with port=nnn setting. You can't return per-protocol port like that, you need to look at the protocol requested by user and return port based on that, or you can omit port to default into "standard port". not using ssl/starttls is default. > On 17 September 2018 at 16:35 Alexander Chekalin <alexander.chekalin at gmail.com> wrote: > > >

I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @<proxy-addr>... So the callerID on the phone looks like: 2145551212@10.10.10.10 which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? 7.5

auth process receives the protocol requested when performing authentication as variable %s (see https://wiki2.dovecot.org/Variables) You can use this to choose the value you return for port. Aki > On 17 September 2018 at 16:56 Alexander Chekalin <alexander.chekalin at gmail.com> wrote: > > > Seen that URL but port= is strange due to there is no protocol connection. > So

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

Hi, I try to set up dovecot as a proxy server, to proxy requests to several dovecot-based backend servers. I wand external clients who connects to this proxy Dovecot to use TLS (this is easy to set up) while want to have unsecured (plain IMAP/POP) connections to backends. You see, links to backends are over LAN so no TLS needed, and these backends are poor old machines (with old Docecots like

Thank you! Ok, so I can omit ssl=no and startssl=no, and this results in default settings for ssl which is 'off'? Or the defaults are 'on' anyway? Can I somehow specify ports on remote hosts that proxy will use to connect to? Like (just image): 'proxy host_imap=10.1.1.1:143 host_pop=10.1.1.1:110' or somehow? On Mon, Sep 17, 2018 at 4:33 PM Aki Tuomi <aki.tuomi at

Seen that URL but port= is strange due to there is no protocol connection. So if I set port=12345 then what proto will I see there? Misleading setting this is why I mentioned (non-existing) per-proto port setting above. May I please ask for any example on how to pass port per proto? It is a bit fuzzy for me to figure it out but I do believe you used to use it somehow. On Mon, Sep 17, 2018 at

Hi All, I'm getting a strange warning message when I perform a MYSQL data lookup. The operation performs fine, I retrive the data I'm looking for and continue on through the dial sequence without an issue. I'm wondering if this warning message is something to be concerned about, can't find any info about it. warning message: Mar 28 15:55:40 WARNING[27481]:

Hi, I have several Grandstream phones connected to Asterisk, some behind NAT and others not. If I reboot all the phones, everything is fine. Should the connection go down, and then come back again, those behind a NAT are still able to make calls, but are unable to receive calls. -- Executing Dial("SIP/1239-ba74", "SIP/1242|60|t") in new stack Jan 12 23:45:19

http://bugzilla.mindrot.org/show_bug.cgi?id=1239 Summary: OpenSSL ENGINE support broken on Solaris using Sun Studio compiler Product: Portable OpenSSH Version: 4.4p1 Platform: Other OS/Version: Solaris Status: NEW Severity: normal Priority: P2 Component: Build system AssignedTo:

Hi. I'm using samba to authenticate squid users in Windows AD, and to provide 'em some statistics on the Internet usage. As this requires the existence of windows users in Unix environment, I use nsswitch.conf and nss_winbind.so to map users in Unix environment via samba as domain member (security = ads) and its winbindd daemon. As of samba 3.5.6 I'm expiriencing some major

Dear all I''ve gotten into the habit of including SVN headers in my templates, etc so it is easy to see where the file installed into /etc/puppet/ came from. Furthermore, we use svn cp to create release branches. Therefore, you''ll see something like this: # $Id: dumpadm.conf 1239 2007-10-23 16:04:06Z sa_dewha $ # $URL:

I am trying to configure Asterisk to receive an inbound call on a Zap channel T1 and Dial a SIP UA registered to Asterisk. SIP Debug and pcap output shows that asterisk is sending an INVITE, followed by an immediate SIP Cancel message. I hear one ring on the called party and then an immediate disconnect - don't know why. The phone actually sends back a 100 Trying, 180 Ringing, 200 Ok,

Hi, After I successfully dumped the GPO policies on my working domain controller I would like to reuse it on a different domain server, but when I use the following command: samba-tool gpo restore B59E0B93-8226-40CA-A5C8-58A7AA1D139E /var/tmp/samba_gpo/policy/\{B59E0B93-8226-40CA-A5C8-58A7AA1D139E\} I got this error message: Using temporary directory /tmp/tmpo7huf4c0 (use --tmpdir to