similar to: MeetMe over IAX2 Test

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'd like to integrate a few peoples cell phones into our asterisk system as we're walking around a data center some days and carrying another cordless phone just doesn't seem to make sense. Forwarding is easy, however once forwarded I want to be able to flash and transfer back to another party / voice mail / etc. How can this be done? As far as I know there's no way to generate

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.

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

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 =>

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).

hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof -- mail:

Hi all, I'm experiencing a problem with SIP channels going ZOmBIE after the following sequence of events: - IAX2 client calls SIP client - SIP client consultive transfers (using sip REFER) the call to a MeetMe extension, and hangs up. At this point, the IAX2 client will indeed be in the meetme room, but a 'show channels' at the * CLI reveals that the SIP channels that were involved

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

We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP

Hi there I am using IAX2 softphones dialing into meetme conferences. I also have jitterbuffer=yes, with typical jitterbuffer settings. The problem I am having is that as soon as there is a delay from a participant, then the delay continues until the participant hangs up and dials in again. When dialing in again the delay seems to go. It seems to me as though as soon as the server registers

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

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 --------------

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):

I'm trying to use the password entered with Authenticate to create dynamic meetme conferences with the following dial plan: exten => _XXXXXXXXXX18467,1,Authenticate(/etc/asterisk/meetme.pw|a) exten => _XXXXXXXXXX18467,n,MeetMe(CDR(accountcode)) ; 281-8467 However CDR(accountcode) is always being set to 1022 no matter what password is used. The passwords are stored in a file so they can

hi guys: recently I want to buinding a meeting confence on asterisk and use the meetme application. I have a ztdummy kernel afteri the lsmod commond: ztdummy 5768 0 zaptel 182660 28 zttranscode,ztdummy crc_ccitt 3008 1 zaptel I also configure the meetme.conf conf => 1000; my extensions.conf [default] exten =>

Hey Guy! I have following dialplan for meetme and i want if someone type wrong meetme extension it should say invalid extension. But look like following doesn't work. its just hangup if i type wrong number. how to fix this code.. ;Conference rooms/lines: exten => 7580,1,Goto(ivr-meetme,s,1) [ivr-meetme] include => meetme exten => s,1,Answer() exten => s,n,Wait(1) exten =>

One of the features that I thought would be popular with the Web-MeetMe suite is the ability to start all non-admin callers in a muted state and selectively unmute them. For example any large conference that is of an announcment nature with a Q&A session. It's probably a feature I should have tested better, but I just discovered that a caller that is joined to a MeetMe with the |m flag

Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen