similar to: FWD <> SIP <> Asterisk <> IAX <> Firefly

Hello, I am having horrible echo problems when using the FireFly client on both the caller and callee sides of the call. When I use another IAX soft client like IAXcomm or IAXPhone I do not have the same echo problems. Has anyone else experienced this and do you know what might be the problem? Thanks, dj -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

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

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

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an

Hi Folks, Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call

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

Hi, I was wondering how other people might handle this situation. We have a 10 channel fractional E1 and have so far only allocated about 25 numbers out of a 100 number block (9500 - 9599). If an external caller dials an invalid extension then we would like for them to be put through to our main number. In order to make for a short and simple extensions.conf we have defined our incoming context

I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though

>>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

Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

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

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

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

I've got a setup where we have 100 DID's. Our default dialplan has one line that calls a macro: exten => _22XX,1,Macro(STDEXT,${EXTEN}) The macro is pretty basic: [macro-STDEXT] exten => s,1,NoOp exten => s,2,Dial(SIP/${ARG1},15,Tt) exten => s,3,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(${ARG1}|u) exten => s-NOANSWER,n,Hangup exten =>

Two problems, possibly related: Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4 are FXS, 5-8 FXOs. Here are the config files: /etc/zaptel.conf: -------- fxoks=1 fxoks=2 fxoks=3 fxoks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 loadzone = us defaultzone = us -------- /etc/asterisk/zapata.conf: -------- [channels] language=en usecallerid=yes hidecallerid=no