similar to: Case studies for 120 simultaneous calls on IVR

Hi, We have a working asterisk with 1 E1 and a TE405P. During conversation, when one side is speaking, the other hears sometimes (every 5 to 20 seconds, it is random) a little deformation of the voice/sound. It sounds like "scricth" and not during more than 0.5 seconds. We have exactly the same problem when asterisk give messages to caller by an IVR. No need to be lot of

Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

I tried to upgrade to 1.0.5 for libpri and all asterisk stuff to the last version. But now, my spans stays in RED Alarm. All seems ok, no error. Just no more links up I use a TE405P. Any idea ? I think it?s something from the old release that make conflict. So I tried to get back to previous version but I got the same problem. Not more SPAN UP Is there a script or something to

I've been testing both T400P and TE405P boards and I'm running into some kind of hard limit on the number of simultaneous calls. This is on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1. Everything is fine up to 190 channels, but the 191st call fails every time with errors like: Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1 Dec 14 15:44:00 WARNING[1215]: Failed

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Hola Colegas: Me reintegro al estudio de R y quisiera me pudieran resolver el problema que planteo en el archivo adjunto. Gracias anticipadas. *MANOLO MÁRQUEZ P.* ------------ próxima parte ------------ Se ha borrado un adjunto en formato HTML... URL: <https://stat.ethz.ch/pipermail/r-help-es/attachments/20130511/b0c9f15d/attachment.html> ------------ próxima parte ------------ An embedded

Anyone care to elaborate on the differences between the T400P and the TE405P? -Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050908/e817f8bb/attachment.htm

I have my peers registered to SER.asterisk seems to be sending mwi for the peers seen in the sip show peers CLI command. i have my ser server registered with asterisk as a type=friend and all clients register to ser.how do i get mwi to work for these clients registered to SER. Thank you, -AA

Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much

I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) "three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz)" I cannot get the card working in any of the slots.

Hi, I am an asterisk newbie and looking around for information . I wish someone could take their valuable time off to answer my query in detail. I wish to set up an IVR system that can allow user authentication and therefter accept 2-3 inputs from users ..generate a key and transmit the same in voice back to the user . The system will intially have small load but if the whole package in future

I am getting ready to purchase my first Digium card to start experimenting with Asterisk. Before I make my purchase, I wanted to make sure I'm not going to have issues with these cards (need to see what the specs are on my box, 5V or 3.3V PCI ). I will be using Asterisk @ Home, so will be Asterisk v1.0.9. I took a quick poke at the lists, and it appears several people have been having

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Hi, I would like to realize a voip testbed that should simulate the scenario in which two companies have an asterisk PBX connected through a PRI-ISDN to the Telco operator. I have no experience of T1/E1 connection but I think that the above could be relized with 3 asterisk boxes equipped with Digium TE405P cards. One of the box will represent the Telco, the other two, the two companies PBX. I

Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31

My Asterisk server believes that a Digium TE405P and sound are incompatible. Basically, no matter what else I do to the machine it terms of hardware, if the TE405P is installed, none of the playback/background/etc commands work. MOH works fine. So far, I have tried: 1) Seven different PCI soundcards with different chipsets. (Go go Computer Junk Store!) 2) Disabling each and every device

This works fine. Message: 26 Date: Tue, 7 Jun 2005 17:22:23 +0800 From: "Ma Zhiyong" <mazhiyong@sohu.com> Subject: [Asterisk-Users] te405p and dell poweredge To: <asterisk-users@lists.digium.com> Message-ID: <004501c56b42$6a12fcd0$7301a8c0@hylaclient> Content-Type: text/plain; charset="gb2312" Hi, I want to install * and te405p on Dell Poweredge 1850. Can I

We originally purchased a TE405P so we could connect our Asterisk server directly to the T1 PRI from our provider, however due to all the problems reported with T1 PRI's interacting directly with Asterisk, we scrapped that idea and decided to stick with our Cisco router making the conversion to SIP. Anyways, we found a new potential use for it. We have an old location with an existing PBX,

We recently migrated our install from an ancient fedora 11 install of samba and openldap to a centos 6.3 setup with its openldap and samba. The domain has been totally recreated from scratch as the person that did the previous setup has not been employed here in many years. After fighting with shares for a while we mostly got them fixed and working. However the biggest issue now is when our GIS

I have no experience with GIS whatsoever. I really know squat about it. I'm currently working on a project to plot locality data for wildlife on a map of Shasta County. This is for a new herpetological club that hopes to track strength of our local herp populations, since it seems evident that in the last 20 years we lost one species of frog and are close to losing another (probably less