similar to: Performance on VoIP

Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

I have two loop-start phone lines connected to an Adtran TA 750 channel bank. When a call comes into asterisk thru the first phone line and the call is redirect (in transit) back thru the second phone line, the channel bank will leave one or both lines opened (ie hangup with spawn a connection again). does any body have a suggestion: 1) I checked the cables into the channel bank from both phone

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/

Hi. I am having trouble with something that should be simple. I am unable to get sprintf using escape sequences: > sprintf("a\nb") [1] "a\nb" > sprintf("a\"bc\"d") [1] "a\"bc\"d" But it seems to need them any way: > sprintf("a\"bc"d") Error: unexpected symbol in "sprintf("a\"bc"d"

Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank Thanks Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/1131cad4/attachment.htm

hello, I would like to realize a VoIP Gateway, with some extra-features. The aim is to get the phone number of the caller, to make research in our database, and to put him automatically through the good employee. The company is equipped with a VoIP network : software : - PSTNgw - Ohphone - OpenGatekeeper hardware : - Quicknet phonejack - gateway : Voicetroniw OpenLine4 Is it possible to

Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B.

All, I need help. I have an (actually 2) Adtran TA750's with 8 FXO ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put an analog phone on each incoming line, I have no buzz. I also have 2 Carrier Access Access Bank I's with 12 FXO ports. When I plug the same analog lines into either one of those, no noise or buzz whatsoever. I went so far as to move the TA750

I have a T3 coming from my carrier. From there I want to use an Adtran mx2800 T1 Mux to break the T3 into 28 T1/PRI which feed into seven quad T1/PRI equipped servers. Everything seems very straight forward with the exception of the D channels for the T1/PRI. I am not very familiar with large circuits such as T3s. I know that I can use one D channel per set of quad port on each server. So if

I have a test install set up as follows Grandstream 102 ---- Asterisk ---- X100P ---- Adtran Express 3000 --- ISDN line to PSTN Most things I want to do work fine except I do have some intermittent problems with an echo. I am assuming that this is created in the link between the X100P and the Adtran terminal adaptor. I am not sure of the proper terms but on the PSTN side of the Adtran there is

Rich Hi, After Qwest pronounced my circuits as within spec. (yes, disconnected from the house) I listened on the lines with my butt set. Clear of all noise and hum. I then got my box of Cat3 and laid a circuit around the outside of the house and into my lab. Still clear of all noise and hum. I then connected the circuit to the Adtran and made a call. Lots of hum and buzzz. As I said,

Hello, I have just received Adtran TSU 600 with 24 FXS ports. I have installed sucessfuly T100P card. Adtran is connected to t100p with crossover T1 cable. On T100P card I have a green light and on Adtran I do not get any errors or alarms. But I do not get dialtone on FXS ports. Adtran is configured: For Network Timing, fxs ports ore fxs_ls on Adtran. In zaptel.conf: span=1,1,0,esf,b8zs

I have following setup Asterisk - T100P -> Adtran TSU600 P + FXOcard -> PSTN line When PSTN line is plugged directly in to analog X100P caller id is received by Asterisk but when I plug it into adtran I'm not getting caller id. Any ideas what kind of setup Adtran TSU600 requires to pass caller id to T100P ??? regards m.

Hi, Maybe someone has seen this before... I've installed a new T100P, but it doesn't seem to work. I've attached the T100P to an Adtran 750 using a crossover cable. The Adtran shows a red alarm on the T1 interface. The Adtran has been set to factory defaults with FXS cards in 1-3 and an FXO card in 5. The T100P shows no signs of life--the leds are not lit. Should I be seeing

I am having a problem using the Adtran 750 FXO quad card with a Groundstart trunk line. I am able to receive calls on the trunk line, however dialing out is not working. The Adtran does not seem to be doing the signaling. Has anyone used the 750 FXO card in Groundstart mode? Any special configuration issues that I should be aware of? Syed Akbar Alico Systems Inc www.alicosystems.com Tel:

Hello. I have been going thru the wiki and asterisk related sites and have not been able to find any documentation about how to configure an Adtran TA750 channel bank. The remote disconnect supervision doesn't seem to be working, when the remote caller hangs up asterisk takes up to 30-45 seconds to hangup the call. Can somebody help? Thank's

I am trying to do my first asterisk install on a SuSE 9.1 box, using the asterisk-update script mentioned a few days ago on this list. I did read the 'quickstart' document on onlamp.com, and made sure the following packages were installed via yast: bison, cvs, gcc, kernel-source, libtermcap-devel, ncurses-devel, newt-devel, openssl096b, and openssl-devel. The SuSE 9.1 DVD contained

Jay, I have had a lot of trouble with the FXO ports on Adtran TA750. Unless the incoming POTS lines have a balance impedance, they will buzz very bad. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Sunday, April