similar to: error 488

I have a whole bunch of remote devices connected to my Asterisk box, including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only rolled out recently and I am having a problem that is intermittent and inconsistent. It happens to some users but not other users on the same ISP. It happens to users in 2 different countries where the Internet setup (NAT issues) are completely different. It

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

Has anyone tested a grandstream 488 FXO gateway on an Asterisk machine? I read that the 488 has a FXO port on it, can I use the grandstream 488 to pass traffic to the pstn from Asterisk. I would use this at home to pass traffic into a foreign country's PSTN over the internet. Thanks.

Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my

Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 --------INVITE--------> --------INVITE--------> <-------200OK---------- <-------200OK---------- --------ACK-----------> --------ACK-----------> --------INVITE

Hi all, I have an issue with my system and the phone company (Alltel). If someone goes into any mailbox on the server to leave a message, once they hang up on their side the system is not detecting this and will sit there and record until you get the standard message from the phone company "If you'd like to make a call, please hangup and try again", which is preceded by the fast

First of all, I'm not knocking Sipura/Linksys. I have heard very good things about their products. I'm just wondering if they are the only quality shop on the market. I know about the zoom 5801 where you can't dial out the FXO from SIP, only from the FXS port. And I have heard similar about the HT-488 also. I want to know if anyone else makes ATAs where all of the features work

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the

Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=488 netfilter@linuxace.com changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |netfilter@linuxace.com ------- Additional Comments From netfilter@linuxace.com 2006-06-24 19:10 MET ------- Not

I am trying to build PWLIB to get OH323 up and running. I am not an expert in linux but can someone help telling me how I can do the following: How can I add a directory to LD_LIBRARY_PATH?! Thanks in advance ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- ---------------------- For

I have a daily time series and have two questions to get some help with. Firs,t I have dates in simple numeric values. e.g. ymd [1] 20050104 20050105 20050106 20050107 20050110 20050111 20050113 20050114 [9] 20050118 20050120 20050121 20050124 20050125 20050126 20050127 20050128 [17] 20050201 20050202 20050203 20050204 Now, I'd like to compute statistics, e.g. acf, by business days. So, I

Hi, I'm having real difficulty in getting calls to go through with Asterisk. I've managed to check that my SIP connection is made to my provider. Below is an email I received from them: ----------------snip--------------------------------snip--------------------------------snip---------------- I am not certain of the reason for rejection but it has to do with the SDP, it does not

can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. i have been told that asterisk@home has this built in to just a button hit, but i dont want to

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?

Are there any known problems with Cisco routers (Cisco 837) and SIP sessions? I have been trying to track down a problem for about 3 hours now and I think the Cisco router is the culprit!!! I keep getting "488 Not acceptable here" messages, which are apparently normally the message you get when a common codec can't be found. I'm also getting "chan_sip.c:3434 process_sdp:

Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk

I'm new to asterisk and need some help with getting a SIP connection working. I am trying to establish a termination point/DID number in another country. I am currently running Asterisk CVS-HEAD. My foreign provider uses SIP and authenticates via IP address. I am not required to register my SIP connection in order to send or receive calls. Can someone help me with how to understand the

Hello, Here's what I'd like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. I've done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), I'd like to do it with H323. Problem is, how to present a

Hi I have a analog phone connected to my Gateway Handytone and registered to Asterisk 1.4 I have configured my HandyTone 488 (in the section FXS Port) for make and receive calls, however I can not transfer a call when it come via PSTN. But, when a call come from via IP I can transfer it. [phoneanalog] type=friend secret=XXXXXXX context=local nat=no qualify=yes host=dynamic dtmfmode=rfc2833