similar to: rxfax(spandsp-0.0.2pre18) and HT488

Hi, I have HT488 with it's FXO connected to Israeli PSTN (bezeq) when dialing to that PSTN line asterisk see gets the call and direct it to the right extension but if the extension doesn't answer and the dialer is hanging the call the extension will keep on ringing. I'm not an expert but it seems like my asterisk doesn't recognize the hangup signal from the HT488 -or it's the

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

Hello, I don't make VoIP-to-PSTN call from Grandstream HT488, but I do PSTN-to-VoIP and no problems. Somebody can help me? Wendell

Hello, Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. TIA. /wai-sun

Hello,I have single linux router ( fedora core 1 ), 2 ISP, 1 internal network,1 IP space from every ISP My scenario: eth0 1.0.0.2 netmask 255.255.255.252 -> ISP 1 eth1 2.0.0.2 netmask 255.255.255.252 -> ISP 2 eth2 1.0.1.1 netmask 255.255.255.0 -> IP space from ISP1 eth3 2.0.1.1 netmask 255.255.255.0 -> IP space from ISP2 Config I try: /etc/iproute2/rt_tables: 10 isp1 20 isp2 ip

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

Hai, Can you post /etc/krb5.conf /etc/resolv.conf /etc/hosts Output off: samba-tool dbcheck --cross-ncs And last question, are there any DC removed from the domain. 50abc2a4-574d-40b3-9d66-ee4fd5fba076/0x00000005,localaddress=> 192.168.50.40] My guess here the UUID is a removed DC. You samba config looks fine to me. Greetz, Louis > -----Oorspronkelijk bericht----- >

about that systemd part.   systemctl daemon-reload systemctl samba-ad-dc stop   clear the logs in /va/log/samba   systemctl samba-ad-dc start   and and extra check on the dns part. sudo /usr/sbin/samba_dnsupdate --verbose and check your logs again to make sure your all ok.   Greetz,   Louis   Van: Sergio Belkin [mailto:sebelk at gmail.com] Verzonden: woensdag 12 december 2018 14:16 Aan:

Hi, I use the Van Bell repo, I've upgraded from samba 4.7 to samba 4.9 but now it fails, these are the errors: dic 12 09:14:49 samba4 samba[4881]: task[dnsupdate][4881]: [2018/12/12 09:14:49.372290, 0] ../lib/util/util_runcmd.c:327(samba_runcmd_io_handler) dic 12 09:14:49 samba4 samba[4881]: task[dnsupdate][4881]: /usr/sbin/samba_dnsupdate: Failed to bind to uuid

Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences?

Thanks Louis; /etc/krb5.conf [libdefaults] default_realm = EXAMPLE.COM dns_lookup_realm = false dns_lookup_kdc = true /etc/resolv.conf search example.com nameserver 192.168.50.40 /etc/hosts 127.0.0.1 localhost samba4.example.com 192.168.50.40 samba4.example.com samba4 ldap.example.com Output off: samba-tool dbcheck --cross-ncs NOTE: old (due to rename

I'm getting the following when sending to a specific fax machine. Any ideas? File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif' Changed from phase 0 to 2 Slow carrier up Slow carrier down Slow carrier up <<< NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56 49 4e 47 54 00 67 00 80 80 80 0c 01 02 NSF without final frame tag The remote is made by

hello, running asterisk 1.4.11 on CentOS 4.5 I am getting no response on function STRPTIME() the system just hangs, STRFTIME() is working fine as seen below. Same thing happens whether I called in from a softphone or via teliax. While executing the following code : ; exten => s,n,Set(v_ts=) exten => s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)}) exten =>

Ok, shaping on Linux is new to me.. so bear with me if i am just stupid. curtain:/etc/shorewall# grep TC shorewall.conf | grep -v ^# TCP_FLAGS_LOG_LEVEL=info TC_ENABLED=Yes CLEAR_TC=Yes TCP_FLAGS_DISPOSITION=DROP curtain:/etc/shorewall# So it should be enabled, right? ---- tcrules ---- 1 eth0 0.0.0.0/0 all 2 eth1 0.0.0.0/0 all 2 eth2 0.0.0.0/0

Hi all, I hope someone here can help me with the following problem: we have a router/firewall machine which connects our local network to the internet. As there are also some servers in the local network working as mail- and webservers, we are using DNAT to make this machines reachable from the internet. The firewall-machine is connected to the internet with two connections, one ADSL connection

We recently upgraded from Dovecot 1.2.15 (on Debian squeeze) to Dovecot 2.1.7 (on Debian wheezy/stable). For now, we continue to use FTS with Squat indexes. We'd like to eventually switch to Lucene but it's not yet supported in Debian stable. Since upgrading to 2.1.7, we've seen some problems with IMAP search results when an FTS Squat index is present (i.e., when the

hi, after upgrade from 1.0.x to 1.2.x i cannot send faxes my topology: PSTN<-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500 fax log: Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 20d700003cb20000@192.168.1.209 - INVITE (With RTP) Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Feb 13 23:50:35

Hi all: I have problems to setup my zaptel E100P hardware. When I start * after receive the "Asterisk Ready" I see this: *CLI> Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 1 Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 2 Up to channel 31. anfter this: Jun 22 20:37:55

Hello list, We're currently facing some issues concerning T.38 gateway faxing. This is a device used almost exclusively for receiving faxes. Calls are incoming to asterisk on a SIP trunk (sangoma netborder) using G711A. Gateway mode is activated in the asterisk dialplan towards a Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0 with the T.38 gateway patch applied (I know I

I looked at this a bit a while back when you first asked but I didn't see anything obviously wrong. Benjamin, at least one of the referenced failures was on BE powerpc64. I don't remember if that one (http://lab.llvm.org:8011/builders/clang-ppc64be-linux-lnt/builds/4883) failed on stage 1 or 2 and the results pages are no longer available. BTW, I tried testing the same revision many