similar to: unclear about IAX

Hello Everyone, What kind of hardware setup would I need to do if I want a T1 connection to PSTN and have 48 users in office with analog phones. Will something work if I have a T410P card in asterisk and have one T1 going to PSTN and other two to a channel bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks. Deepak

I have the following setup: AnalogPhone1--TDM400P-ASTERISK---via SIP---Softswitch--------POTS Phone2 When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well. When making a call from Phone2, I get a dial tone but after dialing the number I hear nothing (no ringing tone). On Asterisk console it says that a call is coming in and that it is ringing Zap/2. I can also hear the

I have following setup: AnalogPhone_1--TDM400P--Asterisk---SIP---[Softswitch]----POTS-AnalogPhone_2 I can call from AnalogPhone_1 to AnalogPhone_2 and all is fine. When I call to AnalogPhone_1 from AnalogPhone_2, AnalogPhone_1 rings BUT I hear no ringing tone AND when someone picks up AnalogPhone_1, there is no "sound" and parties on both end cannot hear each other. Seems that no

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

When I get a SIP call, I get the following error: *CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 5147771111, 1) == Spawn extension (incoming,

I have been running af Asterisk server Version 0.7.2 for a while now But I I wanted to upgrade my version to the new 0.9.0 or the CVS 1.0 Stable. But when I install one of the new asterisk servers I having lots of troubles with the IAX connection between my servers. When I start the 0.7.2 asterisk server it shows me something lige this == Parsing '/etc/asterisk/iax.conf': Found ==

Hi All, I am having trouble with receiving calls from FWD via IAX. I know this isn't a FWD support forum, but I suspect the problem is my asterisk setup. The problem is that I can dial out to fwd subscribers, even myself but they can't dial me using my FWD number. I don't know much about IAX, but it would seem to me like a registration problem, but I get no errors or warnings in

HI ALL: I am newbie to IAX, my iax.conf is as follows: [general] port=5036 ..... but I donot why it doesnot listen on UDP PROT 5035, instead it listens on 4569 Asterisk CLI debug says: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Nov 30 11:52:12 WARNING[1076220544]:

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Hi, I am in the process of planning a dial plan, In regards to the requirement, I am confused how to go about the dial plan. The scenario is like below. BRANCH - A - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 BRANCH - B - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 Now what I need is that if a user in Branch - A wants

Please can someone help me with the following two error messages: Error 1. I have loaded the Zaptel dirvers and everything is ok with ztcfg. I have configured Zapata.conf and everthing looks good but it apears the Zap channels dont load when starting Asterisk. When I make a call to one of the fxs port I get the following error message. -- Executing Dial("SIP/39-b204",

We have been, unsuccessful, trying to implement inter-asterisk communications using IAX. I was wondering if anyone is having similar problems, or would share information about their succes/ Our environment consists of two servers with public IP addresses and DNS entries that resolve. (for the example we use "domain.net") Based on the information available to us, we have tried the

I did not get any answers to my earlier posting. I hope I have better luck this time- The question: Is it possible to use Asterisk as media gateway controller? I know * supports MGCP, but does that also imply that I can use * to control any third party media gateway (such as one providing media conversion from E1 to AAL2). - DL

Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks, Geo ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

Count me in too. -----Original Message----- From: sip [mailto:sip@intology.com] Sent: Friday, October 17, 2003 1:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in ----- Original Message ----- From: "Paulo Mannheimer" <paulohm@instant.com.br> To: <asterisk-users@lists.digium.com>

Has anybody has issues with the new Native MOH and IAX trunking when placing a call on hold? My scenario, Call is placed on a Definity G3 via PRI to Asterisk. Gets trunked over to another Asterisk system via IAX2. Call is answered by operator and placed on hold. At that point, audio is very broken and feedback pulsing is heard. Bad enough that the caller hangs up. If the call is

Evening, I've got * asterisk up and running with nufone account for inbound and outbound calls from the world... everything works quite nicely except that I noticed last night that when I have a call active from * to nufone, web surfing on the network slows to a crawl... additionally, I have a VPN tunnel to another office (using CIPE) and while inter-office (i.e. - SIP to SIP) calls

Hi, I have a wildcard x100p. I just installed asterisk by following step: # cd ../zaptel # make clean ; make install # cd ../libpri # make clean ; make install # cd ../asterisk # make clean ; make install # make samples When I test Asterisk typing # asterisk –vvvvc I find one error and one warning: [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers ==

i tried with the option encrpted password= yes , security = user in my smb.conf file.. but when i did following on my root prompt #killall -HUP xinetd #service smb restart #smbclient -L localhost so i get the problem session setup failed: ERRSRV - ERRbadpw (bad password name/password pair in a Tree Connect or Session Setup are invalid.) i'm not gettin why this is coming. i create the