similar to: remote phones, no audio to PSTN

While pondering several issues, poor quality PSTN POTS lines, potential cost savings with multiple cell numbers, the FAX problems over TDM400p, etc, I wondered about: Cell phone "Base stations" to replace POTS lines. Devices to "cradle" cell phones and connect to TDM400p, for instance, to mimic PSTN. Are there such beasts, how do they play with asterisk? Will FAX work over

Experiencing one way audio using IAX2. I did see some other posts on this, and see there may be some internal issues with asterisk and one way audio. Can this be a widespread problem? So many seem to be using IAX, I find it puzzling. Some information points to this being a problem on asymmetrical connections. This is a decidedly asymmetrical connection, with 1.5 Mbs download and 256 kbs,

I'm researching an asterisk implementation for a client. Originally, they wanted a T1 (as other vendors had quoted such). Now tho, they are asking about "just doing VOIP", cause fortune 500's seem to be so successful at it. That questionable assertion aside, I see there are a lot of outfits (Asterisk2PSTN, for one) that seem to offer what I think it required, a means for

Hi! I want to get audio from the PSTN before the call is answered so I don't miss when the called phone is busy or if there is some error (like the phone is unavailable or is wrong, etc) and hear the ringing from my telco. I have polarity reversal in my telco for incoming and outgoing calls. If I set answeronpolarityswitch=yes then I get no audio until the call is answered. If I set it to

Try this subject line if you will. On 11/14/06, joe a. <joea@j4computers.com> wrote: > > Did not know how to make up a subject line for this. > > I have a dial plan that allows a caller can try my cell phone. And that's > fine. If the call cannot be made, it sends caller back to voice menu. > > However, I'd like a way for the caller to elect to go back to the

My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a.

This is, hopefully, just a case of brain fade. With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and all LEDS on the card lit, I get no dial tone, plugging an analog phone into ports 1 or 2, only a buzz and click. zaptel.conf - defaultzone=us loadzone=us fxoks=1,2 fxsks=3,4 zapata.conf [channels] signalling=fxo_ks language=us context=phones-1 group=0

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

Since the beginning (of my Asterisk life) I have an install that is, supposedly, set up for call waiting. Using a TDM400p, with FXO and FXS modules. On the Analog phones, I can hear the Incoming call (call waiting) tone, but the system does not respond to a "hook flash", to place the current call on hold and answer the incoming call. I have not attempted, nor research how/if this can

I installed Asterisk and a digium wildcard (X100P). Using the extensions.conf with a few changes and a sip.conf file that includes two extensions, I can place calls between the SIP phones. I also can call in to the SIP phones from the PSTN using the X100P. On incoming calls I can hear the default demo announcement and call the digium IAX line. The main problem i'm having is calling out to the

I have a setup with a polycom 601 and an act p160s. All on local segment, no NAT. Can call the act p160s, from the polycom, rings, connects, and a conversation can take place. The reverse is not true, Dialing from the act to the polycom does not work. SIP debug shows, at the end, "Incoming call: got sip response 416 "unsupported URI Scheme" back from 192.168.0.xxx. Which is

Hi all, Newbie quest here. I have 3 PSTN lines (2430-2432) setup by the CLEC in a hunt group coming in to a TDM04B and 3 Grandstream gxp-2000's (say a,b and c) and Asteriskathome installed. I want all phones (a,b,c) to be able to take calls from the hunt group. Does this mean I must make 9 SIP extensions on * ? 3 for each IP phone? and group them in trtiplets (ring groups?)

Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called

Was messing with a polycom 501 and changed the IP from dhcp to static. Working with a user remotely. Now, the user says the phone does not show anything on the LCD and does not respond to any buttons. When rebooting, there is text shown as it proceeds. ?? Is there a way to reset this to a default? Does not respond to ping on the address we set. joe a.

Hi All, Everytime I make outgoing call, the channel at TDM card doing hungup after might be a second the destination number get ringing.The call is from sip phones to PSTN phone. The sip phones was completely registered to asterisk. here is my conf : sip.conf : [1234] type=friend username=1234 secret=aaaa host=dynamic context=sip-ph extensions.conf : [sip-ph] exten =>

Take a look at freepbx 2.2 beta. We have made both ringgroups and follow-me have a call confirmation option. When used, the ringgroup/follow-me extensions that are outside lines (like your cell phone) must confirm they want the call (press 1 to accept, 2 to decline). All the while the caller hears ringing (or MoH if chosen). If no answer, they are sent on to where ever else you want them to go

Hi, I have a question on VoIP adapters. As far as I understand, those adapters are usually used to connect DSL/Cable access to a normal phone (Internet to Adapter, then to PSTN phones). I want to know if you can use those adapters to do the opposite: connect a few lines (1-4 let`s say) to the adapters, then deliver via SIP to an Asterisk box. (I know I could use a TDM400 and Asterisk, but I

Hello, I'm facing problem where as whenever there are incoming call from pstn, there will be no audio coming in. User at the other end also could not hear my voice. This happens few days back. Im using asterisk 1.6.1.2 with dahdi tool 2.2.0. I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and asterisk 1.6.2.5. However, it does not help at all. My current config as follows :-

Getting "no service" display on aastra 480i. Sip debug shows an "unathorized" blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out.

Hello, I am having a problem with my current SIP over VPN setup. We have a server running asterisk at our office. All the phones in the office are on the same network / local to this server. We also have two employees with home offices using SIP phones over VPN to connect to the asterisk server. These phones have no problem with calls to the phones in the office, however there is no audio