similar to: Asterisk Digium FXS

i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I found a LSI161SCREV2 Dialogic board in one of my drawers, and i was wondering if by any luck, i could make some magic happen with asterisk ... If asterisk does not support it, is there any PSTN to H323 or PSTN to SIP gateway that support this dialogic card and that can be connected to an Asterisk Box? Digium, I PROMISE that I will buy my cardf rom you once my tests are conclusive

Hello, I just downloaded and installed the latest version of asterisk under Fedora. (had it under FreeBSD but was having TOOO many problems) After my installation i noticed that the channel H323 was not included ( I remember that i did not have to install it under freeBSD) but I have seen that SIP and IAX are supported though. So i am wondering: Does asterisk consider H323 so achaic that it does

Hello, Digium support kindly proposed to ship a TE120P card to help resolve the issue. I plugged in the card, and introduced the loopback plug. I cleared the red alarm for a while and then i started seeing alarm switching from Yel/Recovering to Blue/Rec with a lot of IRQ Misses. I call Digium that assisted us, and we noticed IRQ sharing with the VGA adapter and the Ethernet port. I changed

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

Hello All, I'm looking to get some feedback on which solution of providing FXS is going to have the best results with faxing. I'm only looking to see what method is going to provide the best digitization into Asterisk, not for transmission from Asterisk to else where. Any recommendations of specific channel banks are welcome. I will need to provide approximatly 216 FXS Ports and need

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

I have 2 * boxes. 1 has 2 PRI's from the Telco, and a PRI to the 2nd * The other has ZAP channels to Channelbanks for endusers. If someone on the second box calls a Toll Free number (it probably doesn't matter that it is toll free) that is auto answered by an auto attendant (QVC, a Bank, the Airlines, Credit Card Companies....) then the call gets dropped with in a couple of seconds of

Hello all, Has anyone use the Supermicro PDSME+ in combination with the TE110P successfully? My experience so far is not very good. I am running trixbox 2.0 but: 1) with zttool I am getting IRQ Misses. Don't seem to have IRQ conflict, but I am now running my SATA HD in DMA. And I am not able to set it in DMA(HIO... operation not permitted) 2) With zttool the Alarm is RED 3) When I do the

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

Hi! Is there anyone that are using a E1-channelbank and have any tips about some type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I think we're using some slightly modified version here in Sweden, but I'll check that tomorrow) and connect one port to a channelbank for 30 analogue telephones. It would also be great to get callerid on the analogue phones, so it would

Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get "a

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar

Hi all, We have been fighting with telco for a entire week. Today they came here with a LITE3000 to analyze what is going on. When I configure zaptel with no external clock, E1 gets aligned/synchronized with bit rate in 2048000 bps, both me and telco. span=4,0,0,ccs,hdb3,crc4 But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in

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