similar to: Question about SIP connection and disconnection events on Asterisk

Hi List: I can not find out an example how to store "include => context name" statement into Realtime static. Please help me on this one. Thanks, Tielin

Hi List: The main concern I have is that multiple accounts of the Asterisk servers could be configured in the phone, any one server of group of registered servers (2-3) down, the phone still works with existing registered servers, or the sip phones automatically register to next server referencing the account settings when current registered Asterisk server is down. Does anybody know some

Hi List: Can someone hlep to provide one or two examples to data entry for sip.conf under the table structure? CREATE TABLE `sip_conf` ( `id` int(11) NOT NULL auto_increment, `cat_metric` int(11) NOT NULL default '0', `var_metric` int(11) NOT NULL default '0', `commented` int(11) NOT NULL default '0', `filename` varchar(128) NOT NULL default '',

Hi All: I'd like to test a pure VoIP call center set up under Asterisk, Can I use existing IP routers to get VoIP traffic from service provider to Asterisk with good quality of voice? In other words, do I have to do any hardware upgrade to make VoIP work in existing enterprise environement, we have 10g Ethernet LAN? Many thanks, Tielin

Hi list: Has anyone done to set up two servers in different remote offices through VPN in order to get the VoIP communication? Thanks for your information. Tielin Xu

Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed

Hi List: Please someone help me to point out where I can get the idea to configure Asterisk for mysql server running on different Linux. Many thanks, Tielin

Hi All: I'd like to use Realtime Static in terms of the performance concern about dynamic realtime. Assume that I create a table: as following: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20)

Hi All: I got questioned about QoS of VoIP, I am thinking that I can configure Asterisk server in this way: I install FXO cards to talk to PSTN as carrier side, I configure SIP phones for our call center agents, that means I can get good quality from PSTN, internally connecting IP phones to Asterisk for agents, please someone confirm me if this arrangement could provide good quality of voice?

I updated to 2.3.0 today, and now I'm getting these entries in my log: Jul 7 14:26:20 mail-cbf dovecot: anvil: Error: connect limit: disconnection for unknown pid 17687 + ident lmtp/backup at backup.invalid Jul 7 14:26:20 mail-cbf dovecot: anvil: Error: connect limit: disconnection for unknown pid 17659 + ident lmtp/backup at backup.invalid Jul 7 14:26:21 mail-cbf dovecot: anvil: Error:

Hi all, I'm interested in buy a Astribank-32 but i never heard about detection of far-end disconnection. Does anyone have some experience about that functionality in this hardware ? Thanks in Advance, Gleidson Antonio Henriques

Dear Folks, Im looking for a way to detect if an analog line is connected to card or not (Im using Sangoma A200). Im using the dialtone detection when dialing but need a way to detect the disconnection of the line when it actually happens. Anyone have any hints or tricks for this? Regards. -- Mohammad Sh. -------------- next part -------------- An HTML attachment was scrubbed... URL:

> Hello, > Here is the context during the problem occurs : > > We have a new machine running on centos 8.From this machine, we restore a > postgresql dump on an other machine runnning on centos 7.After several > hoursof running, restore fails due to a disconnection (no route to > host).But, if we disable the firewall on centos 8, restore succeed. > Before having this new

I will check with firewall-cmd. Regarding hardware problem I have doubt as we use VMWare (but I keep in a corner of my mind).At this moment, we have compiled kernel and have installed it and have tried 4 restore passed.We need to do more tests and understand why 4 restores passed. Thanks Thomas Poty Le jeudi 28 mai 2020 ? 23:58:22 UTC+2, hw <hw at adminart.net> a ?crit : On

Hi all, My scenario is this one: LandLine------------------E1---------------|-------------| |-------------------| |OLDPBX|-------E1-----------|Asterisk1.2.5|-----VoIPusers GSMGateway---------Analogue------ |-------------| |-------------------| What is happening: 1- SipUserAgent "A" Dials a call to a Local Extension "B" in the OldPbx 2- "B" , the called party

Help! I'm having an odd problem that I'm not seeing in any of the list archives and thought I'd ask and see if anyone can help. I've got Asterisk behind a NAT and an SPA-841 SIP phone behind a different NAT. Everything works fine (incoming calls ring, outgoing calls work, audio in both directions, etc.) except for one thing -- when I hangup the SIP phone, Asterisk never

Hello, Here is the context during the problem occurs : We have a new machine running on centos 8.From this machine, we restore a postgresql dump on an other machine runnning on centos 7.After several hoursof running, restore fails due to a disconnection (no route to host).But, if we disable the firewall on centos 8, restore succeed. Before having this new centos 8 machine, we had a centos 7

Thanks Simon, Of course we are not sure but we have a strong feeling : - We tried the restore in loop (14) and all worked fine when firewall is disabled.- We tried the restore several times but no more 2? succeed restore at a row when firewall is enabled. We also tried : - - iptables avec nftables en backend - - firewalld avec nftables en backend - - nft avec nftables en backend

All: I tested echo test by dialing *43 under Asterisk configured by FreePbx by using x-lite softphone. I could not figure out how the call is routed to context from-internal. In sip_additional.conf, I have three extensions defined as 2826, 2800 and 2801, which all are defined context as from-internal. FreePbx doesn't define any entry for *43 as an extension in sip related config files. I

Hi I'm running Samba 3.0.24 both on a Debian etch and a Ubuntu 7.04 both as PDC's and I'm experiencing some really strange disconnection issues. i have net mapped a share to p: and after doing som copying of files all of a sudden it says that the p: drive and i can't acess it for a couple of seconds and then it shows up again. Browsing through the logs i can't really find