similar to: Nextel phone and mute on Asterisk?

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'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

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

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

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'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 =>

Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others.

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

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):

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 have a customer that is complaining that any call coming in from Nextel gives a fast busy. We are running Asterisk 1.4.7.1 with Zaptel 1.4.3 and all the MFC/R2 patches and libraries. All other calls go out and come in, just Nextel seems to have this problem. The phone company technician connected a PBX emulator on the line and that one could receive the calls from Nextel. The E1 is provided

Sorry, repost --------------------------------- Hi, I'm trying to build a vpn between two hosts with valid IP address. One maquerades the private network 192.168.1.0/24, the other masquerades 10.101.0.0/24. Configuration files and routes seem to be ok. I'm getting the following on beta: Could not set up a meta connection to alpha: Ago 5 17:56:24 Viatel01 tinc.nextel[16355]: Trying to

Hi, I'm trying to build a vpn between two hosts with valid IP address. One maquerades the private network 192.168.1.0/24, the other masquerades 10.101.0.0/24. Configuration files and routes seem to be ok. I'm getting the following on beta: Could not set up a meta connection to alpha: Ago 5 17:56:24 Viatel01 tinc.nextel[16355]: Trying to re-establish outgoing connection in 245

What I'm looking to achieve: Incoming calls to me extension will ring for 15 seconds. After that, I want the calls to forward to my cell phone and attempt to get through for another 30 seconds. After 30 seconds, I would like Asterisk to timeout the call, and goto my Asterisk voicemail. I've got the call forwarding down, but I'm using Nextel for cellular service, which loves give

I have a Vonage acct that uses the Cisco ATA186. Currently, I have the ATA186 plugged into a SPA3000 to act as the FXO port. I installed a X100P card with the idea of replacing the SPA3000. Now, when I plug in the ATA186 into the X100P card and make a call into the system (from cell phone) and hangup when the IVR is playing, Asterisk is not detecting a hangup and keeps looping the IVR. If

Hello: I have problems sending dtmf signal to an ATA186 my configuration is: ATA186 --> asterisk --> ATA186 --> FXS to FXO Converter --> PSTN The ATA186 are set to send dtmf RFC2833, but it seems that the ATA186 can't generate dtmf so I can dial to a PSTN number. Is there a setting that can fix my problem, inband dtmf does not work because I'm using G729 codec Thanks

Dear R-helpers, I am using confint() within a function, and I want to turn off the message it prints: x <- rnorm(100) y <- x^1.1+rnorm(100) nlsfit <- nls(y ~ g0*x^g1, start=list(g0=1,g1=1)) > confint(nlsfit) Waiting for profiling to be done... 2.5% 97.5% g0 0.4484198 1.143761 g1 1.0380479 2.370057 I cannot find any way to turn off 'Waiting for. .." I tried

I have a CyberPower PR3000LCDRTXL2U with a BP48V75ART2U expansion chassis, which I am monitoring using NUT 2.8.0 (on Gentoo Linux). TThe UPS appears to be telling me that the batteries need replacement due to age. CyberPower support has confirmed that for me and told me how i should be able to mute the alarm from the front panel until I can replace the batteries, but it doesn't appear

I tried using the iLBC codec, and whlie I like it, I ran into a strange issue. I did a few searches on Google and haven't found anyone with the same issue as this. Anyhow, I was using a Plantronics analog headset and box plugged into a Digium TDM card, dialed out over my VoIP provider's IAX channel to the PSTN. I was in a conference call which is running on an Avaya PBX (which

We're beginning to test MultiTech's CallFinder CDMA Units, one for Sprint PCS and one for Alltel. They sell both. Our intent is to use them as a backup line for our main office (which has a PRI) and a backup/911 line for our remote offices which are all connected via * over a VPN with no local trunks at any of them. In the interest of maximizing use of the lines, I'm putting