search for: vici

...rectory ZAPTELVERSION="1.2.10" build_tools/make_version_h > version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp /lib/modules/2.6.15-1.2054_FC5/build make -C /lib/modules/2.6.15-1.2054_FC5/build SUBDIRS=/root/vici/zaptel- 1.2.10 modules make[1]: Entering directory `/usr/src/kernels/2.6.15-1.2054_FC5-x86_64' Makefile:486: .config: No such file or directory WARNING: Symbol version dump /usr/src/kernels/2.6.15- 1.2054_FC5-x86_64/Module.symvers is missing; modules will have no dependencies and...

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 fo...

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 --------------...

...'s probably > "context=local_200". Then you put the outbound dialplan in that context > in extensions.conf. Mind you, then 200 is the only phone that can dial > out. 201 can only dial 200 and nothing else. Wait a minute, slow down. I re-installed, same sort of problem: vici:~ # vici:~ # asterisk -rx "sip show peers" Name/username Host Dyn Forcerport ACL Port Status 300/300 (Unspecified) D N 0 UNKNOWN 301/301 192.168.0.24 D N 5060 OK (29 ms) 302/302 (...

...ess 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 perfectly. Normal Hang Up : ----------------------------- Quote: vici*CLI> -- Executing AGI("SIP/cc101-09f44300", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-09f44300", "SIP/VOIP74/17274507674||tTor") in new stack -- Called VOIP...

...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 load_modules: Loading module chan_zap.so failed! [root@vici agc]# Ouch ... error while writing audio data: : Broken pipe what is the problem, any suggestions ? Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061024/00989012/attachment.htm

...o I'm just asking in general. For SIP to SIP peer calling, and by that I just mean "ring" or "beep," some sort of ping, basically, just configure the two softphones to use the IP address for the Asterisk box? also: tleilax:~ # tleilax:~ # asterisk -V Asterisk 1.8.32.1-vici tleilax:~ # tleilax:~ # asterisk -rm Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components li...

...lax message received: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.1.1:55238;branch=z9hG4bK.3e59b63f;alias;received=192.168.1.3;rport=55238 From: sip:sipsak at 127.0.1.1:55238;tag=1e6fe4eb To: sip:123 at tleilax;tag=as7dc4727d Call-ID: 510649579 at 127.0.1.1 CSeq: 1 OPTIONS Server: Asterisk PBX 1.8.32.1-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:192.168.1.2:5060> Accept: application/sdp Content-Length: 0 ** reply received after 0.627 ms ** SIP/2.0 200 OK final received thufir at doge:~$ thufir...

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 =>

...o SIP peer calling, and by that > I just mean "ring" or "beep," some sort of ping, basically, just > configure the two softphones to use the IP address for the Asterisk box? > > > also: > > > tleilax:~ # > tleilax:~ # asterisk -V > Asterisk 1.8.32.1-vici > tleilax:~ # > tleilax:~ # asterisk -rm > Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for > details. > This i...

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 :

...ion isn't the greatest. My LAN uses a wireless bridge to connect to another LAN. It's just a home setup; it is what it is. How do I test a connection? How do check the settings? As far as I can tell, the settings are correct. tleilax:~ # tleilax:~ # asterisk -V Asterisk 1.8.32.1-vici tleilax:~ # tleilax:~ # asterisk -rm Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components l...

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 -fomit-frame-pointer -fPIC -c -o app_voicemail.o app_voicemail.c app_voicemail.c: In function ?sendmail?: app_...

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 a...

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

...OK > Via: SIP/2.0/UDP > 127.0.1.1:55238;branch=z9hG4bK.3e59b63f;alias;received=192.168.1.3;rport=55238 > From: sip:sipsak at 127.0.1.1:55238;tag=1e6fe4eb > To: sip:123 at tleilax;tag=as7dc4727d > Call-ID: 510649579 at 127.0.1.1 > CSeq: 1 OPTIONS > Server: Asterisk PBX 1.8.32.1-vici > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Contact: <sip:192.168.1.2:5060> > Accept: application/sdp > Content-Length: 0 > > > > ** reply received after 0.627 ms ** > SIP/2....

...received: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.1.1:56377;branch=z9hG4bK.0edaada3;alias;received=192.168.1.3;rport=56377 From: sip:sipsak at 127.0.1.1:56377;tag=6b540010 To: sip:devries at tleilax;tag=as02b0fdd6 Call-ID: 1800667152 at 127.0.1.1 CSeq: 1 OPTIONS Server: Asterisk PBX 1.8.32.1-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 ** reply received after 0.844 ms ** SIP/2.0 404 Not Found final received thufir at doge:~$ However, I'm sure you're ri...

...8a4d;received=192.168.0.28;rport=5060 From: "201" <sip:201 at 192.168.0.99>;tag=5fbdd638 To: "201" <sip:201 at 192.168.0.99>;tag=as78b94599 Call-ID: d7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0 CSeq: 4 REGISTER Server: Asterisk PBX 1.8.29.0-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="43b1ba24" Content-Length: 0 <------------> [Mar 23 19:26:0...

...-- Executing DeadAGI("Local/91441425477388 at default-86e4,1", "agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------completed, returning 0 vici*CLI> My agent are NOT getting calls. -- AMD: HANGUP ?? Is that an Issue ? How to solve it ? I have below entry for 8369 : *Code:* ; VICIDIAL_auto_dialer transfer script AMD with Load Balanced: exten => 8369,1,Playback(sip-silence) exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)...

...gt; Via: SIP/2.0/UDP > 127.0.1.1:56377;branch=z9hG4bK.0edaada3;alias;received=192.168.1.3;rport=56377 > From: sip:sipsak at 127.0.1.1:56377;tag=6b540010 > To: sip:devries at tleilax;tag=as02b0fdd6 > Call-ID: 1800667152 at 127.0.1.1 > CSeq: 1 OPTIONS > Server: Asterisk PBX 1.8.32.1-vici > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Accept: application/sdp > Content-Length: 0 > > > > ** reply received after 0.844 ms ** > SIP/2.0 404 Not Found > final received &gt...