similar to: Dial app does not indicate ringing to calling party

Source: xen-api-libs Version: 0.5.2-3 Severity: serious Tags: jessie sid User: debian-qa at lists.debian.org Usertags: qa-ftbfs-20130620 qa-ftbfs Justification: FTBFS on amd64 Hi, During a rebuild of all packages in sid, your package failed to build on amd64. Relevant part: > make[2]: Entering directory `/?PKGBUILDDIR?/stdext' > ocamlfind ocamlc -I ../xml-light2 -I ../stdext -I

Package: xen-api-libs I cannot backport xen-api-libs to an oldstable system. It fails with: $ dpkg-buildpackage -rfakeroot -us -uc dpkg-buildpackage: source package xen-api-libs dpkg-buildpackage: source version 0.5.2-3.1 dpkg-buildpackage: source changed by St?phane Glondu <glondu at debian.org> dpkg-buildpackage: host architecture amd64 dpkg-source --before-build xen-api-libs-0.5.2

how would one dial multiple multiline sip phones (cisco 7960) and making sure that all the phones ring on the next available line appearance? I'm currently using the local channel to accomplish this but I'm having some trouble. Here is the configs: each cisco 7960 phone has six registrations in sip.conf, 1XX1 thru 1XX6, normaly when an extension is dialed the following happens: exten

Hello, In my sip.conf I have: ;Register and forward FWD numbers to internal extensions register => FWDNUMBER:PASSWORD@fwd.pulver.com/9500 Which should register Asterisk at FWD and then when any calls are made to FWDNUMBER those calls should be forwarded to extension 9500 as specified in the extensions.conf. What I am getting is it is trying to dial the 9500 (IAX Firefly) client twice when

We just upgraded Cent from 5.3 to 5.5 on a bunch of servers with little to no issues (kudos to you guys) except on a few servers where it appears we are having a problem w/ the Tomcat upgrade. Previously our application server was working fine under 5.3 with Tomcat/Java installed as such: java-1.4.2-gcj-compat-1.4.2.0-40jpp.115 java-1.6.0-openjdk-1.6.0.0-0.30.b09.el5

Well.. I cannot see where stdext was being used, but with your pointer to where this is setup I was able to start using stdext and fix the problem. Patch attached. On Tue, 16 Nov 2004 10:25:50 +0100 Morten Ofstad <morten at hue.no> wrote: > >>I've also determined why VC++ complains about deprecated destructors > >>when using hash_map. Because it's not ANSI

Source: xen-api-libs Version: 0.5.2-3.1 Severity: serious Tags: jessie sid Control: block 718767 with -1 Hi, xen-api-libs FTBFS when binNMUed against ocaml 4.01.0. >From the amd64 build log: > ocamlfind ocamlopt -I ../xml-light2 -I ../stdext -I ../uuid -annot -g -package xmlm -c -o rpc.o rpc.ml > ocamlfind ocamlc -I ../xml-light2 -I ../stdext -I ../uuid -annot -g -c -package camlp4 -pp

>>I've also determined why VC++ complains about deprecated destructors >>when using hash_map. Because it's not ANSI (yet), Microsoft decided to >>move it from the std namespace to the stdext namespace. Use of >>std::hash_map is therefore deprecated. Similar shenanigans have been >>committed by gcc from one version to another. I see where this is

Le 02/12/2013 18:05, Mathieu Malaterre a ?crit : > I cannot backport xen-api-libs to an oldstable system. It fails with: > [...] > ocamlfind ocamlc -I ../xml-light2 -I ../stdext -I ../uuid -thread -c > -o threadext.cmi threadext.mli > File "threadext.mli", line 63, characters 9-15: > Error: Syntax error: 'end' expected > File "threadext.mli", line

Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-000000: Append Cat-000000: default Var-000000: 127 Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do ActionID:

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

I am running 1.4.10.1. I have a macro that is called from default for a certain extension (both below). I added NoCDR to s to try and stop extra CDR records, but I am still getting them. Any idea how to stop them? extensions.conf: [macro-STDEXT] exten =s,1,NoCDR() exten =s,2,Dial(${ARG1},30,Tt) exten =s,3,Goto(s-${DIALSTATUS},1) exten =s-NOANSWER,1,Voicemail(${ARG2}|u) exten

I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each phone as already there own mailbox because if someone know there extension

Hi. Has anyone experienced hangup detection problems with the VoiceMail2 app? I have a console phone on the FXS port. When I call a SIP phone, and get its voicemail greeting, I can enter the VoiceMail2 app, leave a message, and then hit # to stop message recording. Recording does stop, but the channel stays up inside the VoiceMail2 app (as shown by a "show channels" command) for about

Many thanks to Martin Pycko and Mark Spencer. Mark's suggestion below was correct: "Maybe it's stuck trying to send the e-mail notification. If you take the e-mail address out of /etc/asterisk/voicemail.conf does that speed it up?" Indeed it did! The problem turned out to be a 60second delay while invoking mail, caused by a mis-configuration of my hostname and

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Adams, Gavin Is there any additional information I could provide to start tracking this down? I was thinking about looking into the various applications source to see how they access the data elements for callerid. I know where the values are pulled

When someone calls in on a zap channel with FXO and presses an extension, and another user picks up using (*8) I changed it to 888, after a few minutes ( I think 2), the call gets dissconected. The users all use Cisco 7960. I didn't yet have a chance to test it when not using Call Pickup (*8)888. Please help. Here is the screen shot in asterisk: +++++++++++++++++++++++++++++++++++++++

Due to some XML issues, we installed *1.6 from Sun jdk-6u20-nb-6_8-linux-ml.sh on Centos. * ** *below is the output from *[root at www2 tmp]# alternatives --config java There are 3 programs which provide 'java'. Selection Command ----------------------------------------------- 1 /usr/lib/jvm/jre-1.4.2-gcj/bin/java * 2

In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID "1000" and line 2 as SIP ID "2000". Basically I have this set up so that 1000 and 2000 are "lines in hunting" on incoming extension "555". I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here

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