[Asterisk-Users] Txgain Rxgain

Hi, I currently have a TE210P with 2 E1 lines, one of them goes to the Telco which is fine and the other one goes to a Siemens HiPath 3750 PBX. The problem is that signal that the HiPath return is to HIGH and generates a lot of echo even when talking with a PAP2 on the same subnet,

[Asterisk-Users] Asterisk and Siemens HiPath 3750 issues

Hi, I'm currently facing some issues regarding echo between the asterisk box and the 3750, here is my scenario: TELCO -- Asterisk -- HiPath 3750 (E1) (TE210P) | SIP PHONES When I dial from a SIP

Re: [Asterisk-Users] 2 SIP phones on Y data connector on 1 ethernet

A_ Navone, You cannot use a Y connector on a data (ethernet) connection, you must use a switch or and older hub to accomplish this. Regards, Humberto 2 SIP phones on Y data connector on 1 ethernet - will that cause problems ? thx in advance

Re: [Asterisk-Users] PAP2 and ringing issues

Hi Aaron, I tried the progressinband=no and it worked great. Thanks for the tip. Humberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Humberto Aicardi Sent: 01 November 2005 17:17 To: Asterisk-Users@lists.digium.com Subject: [Asterisk

[Asterisk-Users] PAP2 and ringing issues

Hi, I currently have several PAP2-NA units configured to an Asterisk box, everything works fine except from the fact that after dialing a number I can hear ringing tones. When I connect to the same Asterisk box using XLite or EyeBeam I hear only one, any ideas on what may be wrong on the

Re: [Asterisk-Users] zaptel.conf config for CAS signalling

Even better, share the whole zaptel.conf Humberto would you please share line 213 with us? On 10/18/05, Matt Hess [EMAIL PROTECTED] wrote: I have a customer that needs to do cas signaling across a t1,esf span.. it looks like this can be done but I'm not sure how as the documentation is

Re: [Asterisk-Users] Dial plan questions

Neil, When you use the Dial command you must specify the device to use for dialing, so you cannot use Dial(2201,20) you must use Dial(SIP/2201,20) which informs to use the the SIP device 2201. Regards, Humberto Aicardi I'm afraid I'm quite confused by what I've found on the Wiki. I have

[Asterisk-Users] Error when answering CAPI

Hi, I've a Fritz card which was working fine, recently I changed hardware and my nightmare started. Now when I call someone through the chan_capi (0.3.5 or 0.4.0) it works fine but when I receive calls I always get hungup. Can someone please give some help? Here are the logs: *CLI --

[Asterisk-Users] Segmentation fault

Hi, I have just got to the office and now * is giving me the following error: -- Executing Macro(SIP/204-df20, dial|CAPI/111222:b130) in new stack -- Executing DBput(SIP/204-df20, RepeatDial/204=130) in new stack -- DBput: family=RepeatDial, key=204, value=130 --

RES: RES: [Asterisk-Users] chan_oh323 gatekeeper

Discussion Assunto: Re: RES: [Asterisk-Users] chan_oh323 gatekeeper That's correct, just send FXO calls to the Asterisk box. Calls from H.323 sources will go into the context specified in the oh323.conf file. Adi On Wed, 5 Jan 2005, Humberto Aicardi wrote: You're right it works, but how about

RES: [Asterisk-Users] asterisk addson

I'm having exactly the same problem, I'm currently using * HEAD. If anyone can help please let us know what is the issue. Regards, Humberto -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de rizwan Enviada em: Thursday, January 06, 2005 4:34 AM Para:

[Asterisk-Users] Asterisk and SER security doubts

Hi, I have configured * with a x100p and a E100 E1 card and everything is working fine, now I have setup a SER which the UA would connect, I will be using the * box as a E1 gateway and Voicemail. Anyway, I was alarmed after I tried the integration, when the SER forwards any call to the

[Asterisk-Users] chan_oh323 gatekeeper

. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? Thanks, Humberto Aicardi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

RES: [Asterisk-Users] chan_oh323 gatekeeper

You're right it works, but how about receiving calls, how can you register so the FXO gateways knows where to send the calls? Or I just setup the FXO gateway with the IP address of the * box? Humberto Aicardi -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de

[Asterisk-Users] Echo problems

to eliminate the echo problem using a BRI line. Would a BRI gateway solve the echo problem? Is there a PCI ISDN BRI card with echo cancellation? Please send some help. Thanks in advance, Humberto Aicardi ___ Asterisk-Users mailing list Asterisk-Users

[Asterisk-Users] ISDN HFC cards

. If switching to HFC works better can someone point out where to buy them (online)? Regards, Humberto Aicardi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

RES: [Asterisk-Users] Fax detection CAPI (doesn't work!)

[contrX] supports DTMF but still the extension for fax does not get executed, any toughts? Regards, Humberto Aicardi -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Carl Sempla Enviada em: Tuesday, December 14, 2004 8:15 PM Para: Asterisk Users Mailing List - Non

[Asterisk-Users] Fax detection CAPI (doesn't work!)

,Background(fn-intro) exten = fax,1,capiAnswerFax(/tmp/${UNIQUEID}) ; --- never gets executed?! Can someone provide me with further information if need to setup anything else? Best regards, Humberto Aicardi ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] Best option for FSX: IAXy or TDM400P or Voip phone?

Hi, I currently have a * server with a IAXy adapter and a Voip phone. My doubt is: which is the best option? I personally find IAXy to be very effective, except from the fact that they don't support G729. The other option would be to use the TDM400P, which I have heard that it has some

[Asterisk-Users] Problems with CDR and the dst field!

cdr_addons_mysql.so, is there any way to prevent this from happening? Regards, Humberto Aicardi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

RE: [Asterisk-Users] Question about e1/digium

) ; All calls routed to 4000-0001 would be dial SIP extension 1000 Exten = 0010,1,Dial(SIP/1002SIP/1004) ; In this case all calls to 4000-0010 would ring two extensions Hope this helps! Humberto Aicardi [EMAIL PROTECTED] -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em

[Asterisk-Users] CAPI Newbie

to know if the Fritz PCI card is talking to the NT device? Any help would be welcomed. Thanks in advanced, Humberto Aicardi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE