search for: vicis

Hi iam installing zaptel 1.2.10 on my FC5 when i make iam getting following error any one suggest me whats wrong, i have installed source also in the same server. grep: /lib/modules/2.6.15-1.2054_FC5/build/include/linux/autoconf.h: No such file or directory ZAPTELVERSION="1.2.10" build_tools/make_version_h > version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \

I've got the VICI predictive dialer runnning over IAXs to another asterisk server. It dials fine. I can make phone calls manually with no problem. When VICI dials a new number it rings the other end once and I get the error: Jan 26 13:53:10 NOTICE[10206]: Dropping incompatible voice frame on IAX2/VOIP3/5 of format slin since our native fo rmat has changed to gsm I have set ALLOW=ALL in

Hello; As I am using vicidial and its asterisk version which is 1.8, I need to know the required channels to be existed so the asterisk will support fax, SMS, gtalk, Jaber? In other words, how I can know that it is enabled in this asterisk (actually it is 1.8.21-vici)? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 15-04-09 12:06 PM, Chad Wallace wrote: >> but don't know where to put those lines. I have BABY defined as >> >channel variable: >> > >> >BABY = SIP/babytel_out >> > >> >but that seems circular, somehow. > You put them in the context for your clients... From what you show > below, I'd say they go in the "local_200"

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works

Hi all I have installed 1.2.12.1 in FC5 with libpri.1.2.4 when i start iam getting the following error and it quits == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared object file: No such file or directory Oct 23 16:16:07 WARNING[11084]: loader.c:554

I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a starfish on it. In some ways, astonishing that it's not really that definitive, it's more general -- and it only clocks in at one ream of paper! In any event, I'm having some port problems on my home network: http://security.stackexchange.com/questions/81752/ I need to open ports for

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

I want to do something like: exten => _NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _1NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _011.,1,Dial(Dial({TOLL}/${EXTEN}) exten => _9NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _9Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _91NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten =>

It looks as if that is more of a question/issue with your router, rather than Asterisk. I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut" You SHOULD be able to communicate between devices on the

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk the definitive guide", 4th ed. While I don't have the page handy, I was reading the suggestion to try SIP to SIP before proceeding to outside connectivity. I'm aware that SIP trunking is a construct, but am, obviously, learning the system. What I'd like to do is from the CLI "ping"

Hi all I have downloaded 1.2.13 installing on my FC5 when iam making, iam getting the following error could some one suggest me the what is the problem make[1]: Entering directory `/root/vici/asterisk-1.2.13/apps' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=k8

One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0. Today, when they downloaded , the CDR from the carrier site for 26th June 2010 , they see 50% calls are NEVER dialed by Dialer but it appears in CDR. Amazingly, all the call durations are of 29-30 secs. When we checked the status of the same in Dialer, lead is present there but its marked as NEW which means Dialer has ever dialed

Dear all, I've recently switched from EViews to R with RMetrics/fSeries (newest version of july 10) for my analysis because of the much bigger flexibility it offers. So far my experiences had been great -prior I had already worked extensively with S-Plus so was already kind of familiar with the language- until I got to the fSeries package. My problem with the documentation of fSeries is that

On 2/20/15 6:15 AM, thufir wrote: > What's the difference between user "123" and "devries"? Based on the > output here, they seem the same..? > > tleilax*CLI> > tleilax*CLI> sip show users > Username Secret Accountcode > Def.Context ACL Forcerport > 201 password 201 > default

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at

In the Asterisk log I see: --- [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] <--- SIP read from UDP:198.38.7.34:5065 ---> SIP/2.0 200 OK To: <sip:16046289850 at sip.babytel.ca>;tag=sd3D4swKRc From: <sip:16046289850 at sip.babytel.ca>;tag=as07c833c5 Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK13c68eb7;rport Call-ID:

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script

On 2/20/15 2:29 PM, thufir wrote: > On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > > >> A "sip set debug on" will give you more info on why you are getting the >> 404. It probably has to do something with your context/dialplan. > > on tleilax: > > tleilax*CLI> > tleilax*CLI> sip set debug on > SIP Debugging enabled > tleilax*CLI>