similar to: E1 problems

Hi, I'm installing my first E100P. My zaptel reads the following: Span=1,0,0,ccs,hdb3,crc4 E&m=1-31 My Zapata.conf reads the following: Signaling = em_w Channel =1-15 Channel =16-31 After starting the zapter service I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) ??? PauloHM -------------- next part -------------- An HTML attachment was scrubbed...

Hi, I'm trying to build a call center application that allows attendants to come in the morning and dial a certain extension to make their extension available. I wouldn't like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM -------------- next

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>

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

Hi all I have a server that has two TE412P (T1/E1+DSP) cards installed. One of them configured as an E1 PRI connected to PSTN and another one configured as a T1 E&M connected to Avaya PBX. Each card only uses two ports, so there are 2 E1 lines and 2 T1 lines connecting to this server. The purpose of this server is as a TDM trunk gateway that gets call from E1/T1 and then forward to an IP-PBX

Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=...

Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland

Hi All, Can anyone tell me what are the <dev1> <dev2> parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM

Hi all, We are about to make our first channel bank install. This will be a one PRI outside connection and up to 70 extensions. As the schedule (and the budget) is pretty tight, I would like to learn a little bit more about general experiences with channel banks, like echo cancellation problems, Caller ID usage, etc. TIA, Paulohm

I want to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk. Diagram Telco E1 ===>Proprietary PBX========(CAS)===>IVR

Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM

Hi, I try to find some information about CAS E1 signalling and how it's handled by Asterisk. My customer wants to connect to a BT ITS Netrix by CAS E1 E&M. The system is intended to take the channels and mix them (meetme / confbridge) and send the audio back mixed to each. The layout: BT ITS Netrix: CAS E1 E&M <-> MUX - WAN - MUX <-> Digium TE220, Asterisk I've

Hi All, Does anyone have any experience with the ArrayCom VoIP phone? I bought one a couple of weeks ago, it used to work quite well with * until I misconfigured one option. I now cannot make it work anymore, because the phone boots up, doesn't find a valid SIP gateway, resets itself and keeps rebooting indefinetely ;-( Their technical support refuses to answer my questions. Any hint on a

Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas

After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief R&D engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe E&M. I have done E&M wink but have no idea about E4 Superframe E&M and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve