search for: tieline

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 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

I'm connecting to an NEC t1 card via t100p (working great so far!) however I'm having problems dialing from the NEC system to an asterisk extension (sip-grandstream). If I hit the trunck line and dial REAL quick 103 I get the sip extension ringing; if I don't I get an invalad selection message from asterisk - and I can see on the console only one or two digits arrived. How can I

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

...idecallerid=no usecallingpres=yes rxgain=0 txgain=0 group=2 callgroup=2 faxdetect=both pickupgroup=2 amaflags=billing accountcode=qwe_pri_01 callprogress=yes channel => 25-47,49-71,73-96 the purpose of this is to bridge our traditional voice PBX and connected digital phones to our * box with a tieline, as well as allow incoming DIDs to flow through the * box into the traditional PBX using the same tieline. In extensions.conf, I have a dialplan set up for the qw_pri_01 circuit/context for calls coming in to hit the tie line device. This works fine. Going from the PBX to *, I have an issue. W...

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 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 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 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

...record has not been added to the table. Any suggestions for how to resolve are appreciated! Sincerely, **************** Elaine McGovern Jones ************************ ISC Tape and DASD Storage Products Characterization and Failure Analysis Engineering Phone: 408 705-9588 Internal tieline: 587-9588 jones2@us.ibm.com [[alternative HTML version deleted]]

...t? I couldn't find version 0.8.2 on your web site. The legal question is how long has this version been available. And, has the license changed? Thanks much!! I look forward to hearing from you!! Molly Williamson iSeries Access Project Manager IBM Rochester, MN Office: 507-253-5467 Tieline: 553-5467 Internet address: molly@us.ibm.com

Given: Queue(foo|tHnr||bar) where queue foo includes something like IAX2/gw/18005551212 should # transfer work on the remote phone? A read of app_queue.c looks like it ought to work, but all I get is dtmf sent to the caller. (Incidently, I'd really prefer to be able to hit eg * during the announcement to have app_queue continue on as if there were a timeout. Has anyone looked into doing

Hi all: I have question about connection/disconnection information from Asterisk. I want to do some call monitoring selectively, that means I have a list of call cording request, I'd like to send monitoring command to Asterisk when the call connects to a specific agent who is under recording request, I need to have the connection information, similarly I need to notify Asterisk to stop

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?

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

...huge directory: building file list ... done wrote 229184 bytes read 16 bytes 2400.00 bytes/sec total size is 105872465 speedup is 461.92 Sheri French IBM Enterprise Systems Group FRENCHIE@IBMUSM07; Internet: frenchie@us.ibm.com Dept 48Z, 020-3-J204, IBM, Rochester, MN phone: 1-507-253-3099, tieline: 8-553-3099 Build Team, AS400 Development Technologies

I have run accross an interesting problem--one that didn't happen in 95 but that happens in NT. If you create a shortcut to a DOS program and store that shortcut on a samba drive, if the path where the shortcut is stored has a non 8.3 path, the shortcut won't work and says that it cant find the shortcut.(!) I'm guessing it has something to do with the way that MS deals with the long

...y and helpful ------------------------------------------------------- ----- Originalnachricht ----- Betreff: [Asterisk-Users] DTMF after answer Von: john@simlab.net An: asterisk-users@lists.digium.com Datum: 05-08-2004 19:57 > Hi, > I am trying to link up a comdial PBX to Asterisk using T1 tieline E&M. I > have it working for comdial to asterisk but not the other way. Comdial does > not listen for any DTMF before answering the ZAP channel and requires codes > before allowing asterisk to call an outside line or inside extension. > > Does bnyone know how to get Asterisk to...

...t is not commented. If you want to > know > how it functions contact me off-list. > > Regards > > Petr > > > > Sincerely, > **************** Elaine McGovern Jones ************************ > > Phone: 408 705-9588 Internal tieline: 587-9588 > jones2 at us.ibm.com > > > > > >