similar to: HT488 doesn't disconnect FXO

Hi, spandsp-0.0.2pre18 works fine for txfax with HT488(version-1.0.1.2), but rxfax doesn't work. After some FAX sounds, it hangup! Could someone tell me how to debug? The following is the * CLI> log to 192.168.0.161:43222 -- Executing NoOp("SIP/4881-bde9", "") in new stack -- Executing RxFAX("SIP/4881-bde9",

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

Hello, Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. TIA. /wai-sun

Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences?

Hi I noticed that my system has three sets of data regarding telephony behaviour in different parts of the world: 1. libtonezone , part of zaptel, and the data is from the source file zaptel/zonedata.c . Zaptel seems to use it for generating some tones. 2. /etc/asterisk/indications.conf . Asterisk uses it to play tones of busy, congestion, dialtone, etc. (PlayTones). The format is pratically

channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 100000000 or 'never' to disable *entirely*. If you are in Israel, the following is important: As Bezeq in Israel doesn't like the B-Channel resets happening on the lines, it is best to set the resetinterval to 'never' when installing a box

Hello, I don't make VoIP-to-PSTN call from Grandstream HT488, but I do PSTN-to-VoIP and no problems. Somebody can help me? Wendell

Hi List, I know that the zaptel modules have echo cancellation, but is this possible to do this on VoIP <-> VoIP traffic as well? I'm toying with a SIP gateway which has apparently a terrible call quality and would like to know if there is any way asterisk can help with this. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP

Hi All, I have a GSM box, which needs to connect to a analogue phone line. I've plugged the GSM box to a Grandstream ATA (386). This ATA has extension number 600. Now what I want to accomplish is the following: - If a mobile-number is chosen by a user, asterisk needs to call the ATA (600), wait for a few seconds, and then send the mobile-phonenumber. Or, if it's possible, define the

For Whom the Gaza Bell Tolls -- Part 1 By Edmund Connelly for The Occidental Observer January 16, 2008 ?The Israelis can kill whomever they want whenever they want.? --Paul Craig Roberts I sometimes think that it?s pointless for Americans to talk much about recent events in Gaza because we know how it will play out ? America will do absolutely nothing to interfere with the ongoing massacre.

For Whom the Gaza Bell Tolls -- Part 1 By Edmund Connelly for The Occidental Observer January 16, 2008 ?The Israelis can kill whomever they want whenever they want.? --Paul Craig Roberts I sometimes think that it?s pointless for Americans to talk much about recent events in Gaza because we know how it will play out ? America will do absolutely nothing to interfere with the ongoing massacre.

hi, after upgrade from 1.0.x to 1.2.x i cannot send faxes my topology: PSTN<-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500 fax log: Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 20d700003cb20000@192.168.1.209 - INVITE (With RTP) Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Feb 13 23:50:35

Hi, I've sent the following message 3 weeks ago, yet have recieved no reply at all. I don't want to nag, but could anyone please help me? Thanks, Dror ------------------ Hi, I have a UPS of the Israeli company Sustainer, model S-1000C (http://www.sustainer.co.il/data/files/specifications%20smart%20en.pdf - don't worry, it's in english...). I've been trying to connect to

On the Bombings Noam Chomsky The terrorist attacks were major atrocities. In scale they may not reach the level of many others, for example, Clinton's bombing of the Sudan with no credible pretext, destroying half its pharmaceutical supplies and killing

Hi, I have a UPS of the Israeli company Sustainer, model S-1000C (http://www.sustainer.co.il/data/files/specifications%20smart%20en.pdf - don't worry, it's in english...). I've been trying to connect to it from Debian to no avail for a long time now. The company has attached to it a program called Commander Pro (for windows) and RPMs of UPSmart for old versions of Redhat and

What is the best way to pick a prebuffering length for a streaming audio application using UDP transport? I'm using Speex in a VoIP application with RTP transport, currently with a fixed 500ms prebuffer on the playback side. However, I'd like something a bit more adaptive to accomodate high-jitter connections. For example, in one test configuration there is a very low average

I need to reopen this discussion because it's impossible to run spandsp (and VoIP) under these circumstances. With zaptel unloaded, I see the following "vmstat 1" output: no swapping, an occasional disk output, +/- 1003 interrupts/sec., less than 10 context switches/sec., CPU idle 100%. A very quiet system. I load modules zaptel and wcfxo, and the system utilization stays the

Hello! I encounter the following problem (asterisk 11 and 13) with Teconisy as trunk provider with enabled qualify and default t1min (100ms): Teconisy most often doesn't answer the first invite before asterisk default t1min ended. Therefore asterisk sends one more invite. This second invite is answered by Teconisy with status 486 - Request terminated - Channel limit exceeded. (The second

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Ok, this is a silly question, but what does the jitter buffer do? I'm really new to audio, so please bear with me. From what I gather (primarily from the list archive), the jitter buffer is a wrapper around the Speex decoder. I give it the packets I receive, in whatever order I receive them, and then it gives me back a clean stream of audio samples. But what I don't entirely