similar to: * with cisco 7960G

hi! I'm looking for a robust hardware IP phone which supports SIP protocol inorder to implement a call centre. Have anyone used CISCO SIP phones (eg:- 7960G ) with asterisk. From what I know these CISCO IP phones are very robust and feature rich. Yet I'm nervous whether * don't like CISCO at all. Thoughts are most welcome. denzel. -------------- next part -------------- An HTML

hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2.

hi ! I have asterisk with Licensed G.729 codec enabled. Whenever I make a call using this codec a warning apears as, WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames

hi! I wanna do some arithmatic operations (addition and substraction -integer operation) inside extensions.conf. Is there a simple way to do this. If I do yy = ${xx} + 1 // say "xx" is initialized to '0' the resulting "yy" will show "0 + 1" Obiviously not the result I need. Any help !!!!! denzel.

hi We are using snome 200 IP phone with *. It works OK. But after a period of time we can't hear any sounds for any icoming or outgoing calls. I've got two of these phones. Same symptoms occur to both of these( not at the same time ) and the problem remains until the phone is completely rebooted. Don't know whether this's * or Snomes' prob. Any help would be appreciated.

hi! I've got a asterisk system running with around 50 per calls per minute. I've connected * to internal pabx and outside telecom using E1 (ISDN pris). Sometimes calls disconect abnormally. Is this something we have to live with or is it a bug in CVS code ? denzel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, What the support that asterisk has to send/receive Faxes? Can I plug a FAX machine in the a FXS extension and send out Faxes? What's the codec I need to use? g.711? Also can we receive a FAX into a FXS extension in Asterisk PBX? Also I need to know if we can send/receive FSK data from/to an extension plugged into Asterisk PBX? For example if there's a phone model which can send

We have several CENTOS 4.X which inside Firewall.? I can NOT run "yum" inside firewall to upgrade O.S. to new release. ? Does there has way I can download all RPM to otherside firewall server then transfer those files into servers inside firewall and upgrade. ? Anyone have procedure dicuments or website can be reference? ? Thanks.

Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off

I just copied a file system using xfsdump|xfsrestore At least 1 new directory had been created on the source during the xfer (took 9+hours -- 7TB), so I wanted to verify I hadn't missed anything. Using rsync: > rsync --version rsync version 3.1.0 protocol version 31 Capabilities: 64-bit files, 64-bit inums, 64-bit timestamps, 64-bit long ints, socketpairs, hardlinks,

Hi. I am trying to force some traffic that goes to address 203.7.93.94 through a VPN tunnel. I use freeswan 1.98b and Shorewall 1.4.6c in one machine. The 203.7.93.94 is in the DMZ on the other end. (Both ends use the same shorewall and freeswan). I have successfully set up a tunnel between the two network (using a point to point topology, not hub). I added a static routing that redirect

hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the

hi! I'm trying to get zapras working in GSM csd network. Whenever a dialup call is initiated from the mobile to the * gateway the following appears in the log and zapras terminates. Phone gives the error dialup not answered. ==> /var/log/messages <== pppd[2310]: Plugin zaptel.so loaded. pppd[2310]: Zaptel Plugin Initialized pppd[2310]: Using zaptel device 'stdin' pppd[2310]:

Hi rsync team, I thought that rsync would try to overlap computing and IO on both machines. I'm rsyncing a large tree (18G) and am keeping an eye on that. Suddenly my side completely stopped. No IO visible, no CPU time spent. The otherside was doing 100% CPU. Then the other side started to do disk IO. Then suddenly the activities moved over to my side, and I saw things moving again in the

Reindl Harald <h.reindl <at> thelounge.net> writes: > most likely because there is no serious market > > if i want a NAS i buy a NAS > if i want a router i buy a router > > > no struggle with updates and security holes > no struggle with software versions > no struggle with "i can have this and this but not combined with that" > no struggle

I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close

dear list, as a part my problem. I have to estimate some parameters using ML estimation. The form of the likelihood function is not straight forward and I had to use a for loop to define the function. I used "optim" to maximise the result but was not sure of the programme. To validate my results, I tried to write a function to obtain the MLE of a bivariate normal in the same manner. On

Hello, I'm writing from the otherside of the world from where my systems are, so details are coming in slow. We have a 6TB OCFS2 volume across 20 or so nodes all running OEL5.4 running ocfs2-1.4.4. The system has worked fairly well for the last 6-8 months. Something has happened over the last few weeks which has driven write performance nearly to a halt. I'm not sure how to proceed, and

Hi! I get this error when I'm trying to compile wine 0911 release on Red Hat Fedora (with gcc-3.3.1-5) using the tools/wineinstall script. Can someone please tell what is wrong? I could't compile last month's wine either. There was a segfault at the end of the installation process, when it reached some "wine registry". make[2]: Entering directory

Is anyone using a 3com 3CNJPSE to power a 7960G? I have a couple of 7960Gs and 3CNJPSEs but no combination appears to work. Both phones work fine with a cisco power cube. I get a 47.6V reading across pins 5 and 8 on the patch cable coming from the 3CNJPSE. The network still works through the 3CNJPSE, just no power to the phone until I connect the power cube. Using two straight through wired