similar to: Integrating cell phone into Asterisk Extension..

We were just testing forwarding one of our numbers to a VoIP CLEC and ran into an issue that we've seen before but never figured out the cause of. It seems if you call us and immediately hang up - or if the call is forwarded by Verizon (which stills rings about ? of one time) the Zhone detects the ring and answers the call.. The Zhone however DOESN"T detect that it's no longer

I notice most things say to open ports 10000-20000 for UDP for SIP, however from time to time this range isn't where Asterisk is opening the ports: We're at xxx.xxx.xxx.xxx port 8542 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) This call has no audio, presumably because port 8542 is firewalled in the iptables on the server.

We want to test capacity of our MeetMe room. The thing that is distinct about this is that the incoming line is being delivered IAX2 to our server across the net - so Telephone -> VoIP Gateway -> MeetMe. We want to test both the VoIP Gateway and the MeetMe room performance. You can reach our MeetMe room directly at 1-301-561-9229 If you want to test with us we're thinking maybe 9pm

Hey all, Any suggestions on how to estimate how much data rsync would synchronize on average in a given installation? Assume that a full rsync has already run and the only data being updated is just the daily diffs. Thanks, noam Noam Birnbaum http://maccentricsolutions.com/ 877.luv.macs x89 ð Apple Certified Technical Coordinator ð Apple Certified Help Desk Specialist --------------

I am looking into supporting around 20 SIP clients (ATAs, IP softphones, etc) distributed in around 10 different end points (in South America). For the most part they all have narrow band connections 64kpbs, 128 at most and I?d like to use g729 all around (don?t have too many alternatives) To start with, I will have one * with no gateway to the PSTN and eventually a few * boxes with termination

I'm guessing you aren't on a digital line (i.e. POTS). If that's the case you'll have problems sometimes that the dial tone isn't there as fast as it dials. I forget the pause character (p? It seems like that was pulse not pause). I'd break this into 2 steps, answer and announce something - make sure you are answering fine. Step 2 - Dial via ZAP and hopefully then you

I see in NexentaStor''s announcement of Community Edition 3.0.3 they mention some backported patches in this release. Aside from their management features / UI what is the core OS difference if we move to Nexenta from OpenSolaris b134? These DeDup bugs are my main frustration - if a staff member does a rm * in a directory with dedup you can take down the whole storage server - all with

Greetings - I have not been a long time follower of this list, but I have scanned through the last year or so of archives, after not finding much from google searches. I am hoping someone here can inform me if what I want to do is feasible, and give me some general guidance to follow so that I can continue my research and complete this task. I admin a RH3 system that is primarily a Samba

I've asked this question in the IRC Channel, and have had no happiness yet :-( I have incoming lines hooked to asterisk using X100P's. Unfortunately, when we cal forward our lines using the phone company, the line still rings about a half of a time. This is enough to get * to start 'simple switch' and after my 2 second wait, answer the line. Unfortunately, * doesn't see the

Hello, I have a really irritating issue that I haven't had time to investigate much - I hope someone has encountered it and can tell me a solution. I didn't see anything in searching archives / sites.. When my Nextel i90c phone gets a page (2 way text message via the internet option) it has an irritating tone to get me to hear it. However this tone seems to mute asterisk (reproducible).

I'm having a problem with Caller ID matching. The call is coming in via IAX2 to our system, the caller id doesn't seem to parse right. I just got the latest CVS version an hour ago or so. Relative extensions are pretty simple: [disaid] ; ; Check caller id for disa access ; exten => s,1,Wait,0 exten => s/7031234567,1,goto,disa|s|1 exten => s,2,congestion [main] exten =>

I am seeing this issue posted a lot in the forums: A zpool add/replace command is run, for example: zpool add archive spare c2t0d2 invalid vdev specification use ''-f'' to override the following errors: /dev/dsk/c2t1d7s0 is part of active ZFS pool archive. Please see zpool(1M). (-f just says: the following errors must be manually repaired:) Also, when running format and

Here's what I am trying to do... First I'll have a list of 4 digit numbers like: Code:OtherCode 1234:4321 9999:4444 3333:1111 People will call our 800#, Have the number they are calling from read to them (ANI?) & then enter in the code (let's say 1234). If the code matches one on the list, then the OtherCode (4321 for 1234) will be read/spoken to them. With the exception of the

I have a Dell R710 which has been flaky for some time. It crashes about once per week. I have literally replaced every piece of hardware in it, and reinstalled Sol 10u9 fresh and clean. I am wondering if other people out there are using Dell hardware, with what degree of success, and in what configuration? The failure seems to be related to the perc 6i. For some period around the time

Try : killall -s HUP /usr/sbin/smbd is causes that smbd rereads its config. At 11:20 10.09.2002 +0200, Kurt Pfeifle wrote: >Hi, > >I have a question regarding the visible list of printers in the network >neighbourhood of my Samba server, and how to force it to become updated. >Maybe one of my settings is wrong? Maybe it is a bug? > >My problem (short):

We have following problem. On some of the extentions I call cell phone after 10 seconds or so.Or, like this one below- we call cell and office phone at the same time ;Eric on extension 105 exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)         same => n,VoiceMail(105 at default,u) Where problem comes in - if person not at the desk - his cell phone shows call from OFFICE number and

Hi, being a total newbie in shellscripting and similar stuff, I am suffering from a brain freeze around the following problem: * I need to print (or transfer otherwise) to a WinNT box; * the files are send from inside a shellscript; * the script gets the filename on the commandline when started; * so the only knows it works on "$1"; * the problem is, that the original filename needs to

Hello All, I apologize if this question has already been answered but how do you transfer a call to a cell phone or another land line outside the PBX? Setup I have two pots lines into my current Asterisk Box. I have an outsides sales guy who wants to work off his cell phone or transfer his calls from his extension and the main sales extensions. How can I do this right? -- Otis

I've set up asterisk with my X100P as a home answering machine. Works great so far - answers the phone after 20 seconds, runs the phone tree, emails voicemail, etc. However, the one feature traditional answering machines have that I haven't been able to figure out is how to listen in on the call. Ideally I could just route through Console/dsp and hear it on my speakers. I've tried

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you