similar to: res_snmp

--- hgaillac-sip@yahoo.fr a ?crit : > hello, > > How asterisk could support res_snmp even this module > don't help to monitor all asterisk features? > > monitoring asterisk with snmp would be a good > thing. > Which solution ? > > Harry > --- Kristian Kielhofner <kris@krisk.org> a ?crit : > > > hgaillac-sip@yahoo.fr wrote: > > > I

Just updated two fc3 systems running svn trunk. One updated, installed properly, and is working fine. The second box failed during the 'make install' process with: /usr/lib/libnetsnmp.a(parse.o)(.text+0x275a): In function `unload_module': : multiple definition of `unload_module' res_snmp.o(.text+0x310):/usr/src/asterisk/res/res_snmp.c:102: first defined here /usr/bin/ld:

I'm having trouble building res_snmp (under Asterisk 1.4.0) on a box running Debian Sarge. res_snmp says its dependencies are netsnmp but Debian doesn't seem to have a netsnmp package. I've tried installing pretty much every package available related to snmp and no luck. I'm just wondering if anyone has successfully built the res_snmp module under Debian Sarge stable. Any

Hi, I have problems compiling asterisk 1.4.11 with res_snmp. I do 'make menuselect', and I see that this resource module depends on netsnmp. I am using centOS 4.5. I do: > yum install net-snmp net-snmp-devel net-snmp-utils net-snmp-libs I don't know if i am missing something. I go to the source directory and I do: ./configure but still does not work: > ... > checking for

Yes, I can call the phones, they ring, etc, and call call out, just no outbound audio. Is their any difference in the inbound & outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc? Thanks, Ross ---------- Original Message ---------------------------------- From: "Carlos Alperin"

Ok, With everything restore on rtp.c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at one time. Carlos -----Original Message----- From: Carlos Alperin [mailto:calperin@senecacom.net] Sent: Wednesday,

Hello, I 've installed both cacti and res_snmp for monitoring. Does res_snmp is able to send snmp traps when hardware is out of service or others status ? Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos

I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64 very good, but since FC keeps updating, I tried to follow newer kernel versions. I can't pass the zaptel compilation. Everything is OK, but when I finished, and tried to load it, allways got module not found when I run modprobe zaptel, and modprobe ztdummy. I already tried to modify is with the sed 1 option but

I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual processor machine. The install lookups on the search for the Sata drive, since however it loads the sata_sil driver it doesn't work. Did someone knows what version of Linux is using on Asterisk Now? Thanks, Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL:

> I don't see anything obviously wrong with your configs. > You don't want relaxdtmf. That can cause the problem, not fix it. POST 2 --> got no response Hi Eric, at the begining -> Thanks for your help. relaxdtmf is not written in my config, so it should be at the default, i guess i remember default is yes ? However, the dtmfmode should be the same, i think so, too, but

I upgrade 1.2.7-1 to 1.2.9 but asterisk is not stable I 've lost call SIP<->ZAP. channels. i can't hear sound because of res_snmp.so . Is it a b?ta release ?? I downgrade to 1.2.8 or 1.2.7 I do hope 1.4 will be a real stable realease Harry __________________________________________________ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it

This issue may sounds trivial I need to build a Router for send Internet + VoIP traffic. The computers are in a different network that the Phone Gateway. The Computers are going to be send to a 3 Mbps connection using OSPF, in the meantime the phones are going to be send to a T1 using OSPF too. The routing software is going to be Zebra. I need to switch the outgoing in case that the T1 or

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules /

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

HEllo list ! When user A calls user B via Asterisk (Users A and B are registered on the same Asterisk server ) and an ISDN PRI, user B phone always shows Asterisk server telephone number. How to hide it and how to forward user A number ? We tried usecallerid, callerid, hidecallerid, restrictcid, usecallingpres in zapata.conf but we always see Asterisk server telephone number ! Thanks

Dear All, I need to bind two different ports at the same time for SIP. 5060 and another port number. Is it possible ? It would be something like port=5060,5062 Isamar

Hello Everyone, This is my first post, and this is my problem :-). I have a asterisk@home, work excellent (only internal users), but i need outbound calls. One person give me an access to his "Cisco 5300 Media Gateway", he give me a dial rule and the router ip address. I've created a SIP Trunk, and a outbound routing, with all the info (the rare thing, the