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We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

Is anyone else experiencing problems with Transfer via # and the 'T' or 't' flags passed to Dial()? I've tried both the latest CVS and 0.7.1 tarball. If I dial in from a pstn line and then choose an extension that dials a SIP phone with "Ttm" flags, when I press # on the SIP phone, the pstn caller hears the "Transfer" and the SIP phone gets the music on

Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors: Unable to find a path from G729A to GSM Unable to find a path from GSM to G729A What's up with that? I was able to make a call once

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

I've been messing around with a "free line" notification where an extension is dialed for a second when a line becomes available. I can't seem to get the "h" extension to continue when the local party hangs up. I've seen references to other people having the same problem in the list archives, and the solution presented was to use AGI. I've tried dialing

It turns out that the cause of the transfer problem is the Grandstream 1.0.4.39 firmware. I was shipped a bunch of HandyTone-286 devices that contained the 1.0.4.30 firmware. This version had a bug where the phone would sometimes not ring at all. I was told by Grandstream to upgrade to the 1.0.4.39 version. This broke the "Use # as Dial Key" option, and evidently transfer as well. I

The solution to the problems with the Grandstream 1.0.4.39 firmware is to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to work. However, this presents another problem. When I'm using g729 to place a call, I get the warning "Unable to process inband DTMF" because inband is not supposed to work with g729 (although it does seem to work when I've tried it so far).

There seems to be a problem related to the Grandstream HandyTone-286. When a call is placed through the adapter, the call can be transferred. However, when a call is received through the adapter, the call cannot be transferred. The problem does not exist with a BudgeTone-101 (1.0.4.23) using the same Asterisk configuration and Dial() settings (Ttm). I tried all of the firmware on their BETA

Has anyone else had problems registering with Sipphone over the last few weeks? Previously, this had worked fine. I contacted Sipphone technical support, but they're not much help. register => 17471234567:password@northamerica.sipphone.com/123

Hi, I had been using 4 X100P cards in my Asterisk box, but 2 of them were sharing an interrupt. Therefore, periodically I would hear beeps and clicks that I had assumed were a result of this. So, I ordered a TDM400P with 4 FXO modules and installed it in the box last night. Today, we've had nothing but problems with it dropping calls. I installed the latest CVS of everything, and we've

We have a bunch of PAP2s, and using the # to send immediately does not work as described in the manual. The PAP still waits for the "Interdigit_Short_Timer" to expire before sending the dial string. In addition, the dialplan does not cause the string to be sent immediately as it should. Here's the dialplan I'm using:

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

Barton Hodges wrote: > I've been messing around with a "free line" notification > where an extension is dialed for a second when a line becomes > available. I can't seem to get the "h" extension to continue > when the local party hangs up. I've seen references to other > people having the same problem in the list archives, and the > solution

Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to Asterisk via an X100P or through VOIP. His or her voice would then be sent to the streaming

Hi Folks, I''m using a 2.4.x kernel and TC from the iproute2 package so that I can limit traffic through my gateway. I''m using this to mark packets when they leave the LAN: /sbin/ipchains -A forward -j MASQ -i eth0 -s 192.168.1.0/24 -d 0.0.0.0/0 -m 1 When the packets return, I need to have them marked again so that the ingress filter will limit the bandwidth in the opposite