similar to: Re: Asterisk-Users digest, Vol 1 #3402 - 17 msgs

Hi everyone, Maybe someone could help me. I have Asterisk in production with TE410P connected to PSTN. When I call from internal phones, either voip or connected via other PRI trunk, to PSTN and if the called phone is busy I don't hear anything!?! I should hear tone indicating that called number is busy. I have played with busydetect and callprogress in zapata.conf, but I didn't find

Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a

Can someone tell me what is wrong here: Feb 2 19:45:44 ERROR[1074441696]: cdr_addon_mysql.c:381 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. The database is created, cdr table also, the username and password is right. I have tried configuring cdr_mysql.conf to connect via localhost mysql.sock or via tcp port, but in both cases I got this error. Thanks!

Hi all, We have been fighting with telco for a entire week. Today they came here with a LITE3000 to analyze what is going on. When I configure zaptel with no external clock, E1 gets aligned/synchronized with bit rate in 2048000 bps, both me and telco. span=4,0,0,ccs,hdb3,crc4 But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in

Hi everyone, I want to have a group and dial multiple phones/lines simultaneously. If I use this Dial command: exten => 222,2,Dial(${TRUNKBP}/246&SIP/258&${TRUNKBP}/243,20,tTr) ... all phones ring just once, after that only the first one continues ringing and only that one can answer. Can anyone tell me why? thanks! Tomica -------------- next part -------------- An HTML

Hi all. We are getting our first PRI line to use with Asterisk and one of the technical specifications is about framing, structured or unstructured. The main difference about them is almost clear for me: http://ckp.made-it.com/g704.html says: "G.704 is the framing specification for G.703. A carrier can 'steal' a 64kbps time slot (TS0) from a 2.048 Mbps line and use this to

Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica ---- This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr

Is there any script guru on the list that can help me. I'm trying to start asterisk with nice -5. Normally the command would be: nice -5 asterisk but asterisk start from the scrip on Gentoo as -U asterisk -G asterisk Here is the script: =============== depend() { need net use zaptel } start() { local OPTS USER GROUP if [[ -n "${ASTERISK_NICE}" ]];

A customer contacted us today concerning getting a VoIP to PSTN system with a few IP Phones setup. Asterisk should fit his needs. It is not a big job, but I think that this customer is going to need onsite work. Please contact me off list if you are an interested reseller in the Washington, DC area. Sean _______________________________________________ Sean Robertson NETXUSA p. 800-289-6389

I've configured asterisk to connect a MySQL database for CDR, Voicemail and SIP/IAX2 peers. - CDR are reccorded - Voicemail config is readen directly in the database but SIP/IAX2 peers still have to be declared in sip/iax2.conf to make calls... However, when I restart Asterisk: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found ==

I'm trying to get running an ethernet with two machines: a samba server running suse linux 6.4 and a client running windows 98. I follow the steps in the DIAGNOSIS.txt file, but I get stucked when executing 'net view \\SERVER' in the client. I have laready done everything it says in that file: 1) Set a lmhost file in the client. Just with this line: 192.168.0.1 ASTERIX #Asterix is

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227

Hi Ted there was a drift from the initial topic. worth a new thread! 2014-07-27 9:13 GMT+02:00 Ted Mittelstaedt <tedm at mittelstaedt.us>: > On 7/26/2014 12:18 PM, Arnaud Quette wrote: > >> Hi Eric, >> >> sorry for the lag, summer time... >> >> I'm first seconding Charles comments >> >> 2014-07-09 12:31 GMT+02:00 Eric S. Raymond <esr

I understand that softphone are the answer in fact I deploy a ton of the Ip comm version every week. I am under contract with the phones so I can't sell them and there no easy way out of the contract. As for 79XX's I have several office that have them working over a VPN backed in to our main office where the CCM's and GW's are with managable problem and for the most part they

Dear All, I would like to fit some parametric survival models using left truncated, right censored data in R. However I am having problems finding a function to fit parametric survival models which can handle left truncated data. I have tested both the survreg function in package survival: fit1 <- survreg(Surv(start, stop, status) ~ X + Y + Z, data=data1) and the psm function in package

has anyone seen this? -- Starting Skinny session from 192.168.13.102 -- Starting Skinny session from 192.168.13.102 triton*CLI> Oct 15 13:44:05 WARNING[213019]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 15 13:44:05 NOTICE[213019]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Success Oct 15 13:44:05

Hi Timo, I have another problem, this time with user shared folders: User "markus" shared the folder "ForTest" to test: SETACL "INBOX/ForTest" test akxeilrwts dovecot-acl and shared-mailboxes.db have been successfully updated. As user "test" the folder (and "#User" namespace) is not visible. When I configure the shared namespace with

When sending calls to my Long Distance Provider I've come across this problem. After about 3 or 4 seconds into a call, it gets cut off. This is what I have concluded after doing a trace. 1. An invite is sent to the Asterix PBX 2. Asterix sends back a 100 trying. 3. Asterix then sends a 200 OK, with session description. 4. They ACKnowledge the Asterix 200 OK 5. Asterix then sends a 183

I have an APC SMX3000RMLV2UNC UPS, with installed network card model AP9631. http://www.apc.com/resource/include/techspec_index.cfm?base_sku=SMX3000RMLV2UNC http://www.apc.com/resource/include/techspec_index.cfm?base_sku=AP9631 I planned to use network connectivity with this unit and apcupsd, but have not this far. The UPS is shared by multiple servers, so a USB or Serial connection via the

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log