similar to: Digium TE405P and Cirpack Switch

Hi, I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch (www.cirpack.com). <IP Network>--<*>-[TE405P]-<Cirpack>-<Public PSTN Network> ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack is Network, * is Terminal/User. As I encountered some pb with Sip to Zap transcoding (* to Cirpack way poor quality, the other way fine), I

I'm installing TE405P card. This is my zaptel.conf. -------------------------------------- span=1,0,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 loadzone = us defaultzone= us -------------------------------------- When i modprobe wct4xxp, -------------------------------------- PCI: Found IRQ 11 for device 02:0b.0 PCI: Sharing IRQ 11 with

Hope this is the right place to report/ask for help... Have have a 1.2.7.1 installation running reasonably happily for a while. Thought we might give 1.4.0b3 a go. Ran it on a local test machine (that has the single port card) and all was well. However, when I run it on a machine with the 4-port card, it has trouble with Zaptel. I attach some diagnostics and welcome any suggestions (including

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

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

Hi all, I've a problem in DTMF dialog between * and a Cicpack Transit Node Class 4-5. The call initiat by a mgcp phone pass by the cirpack and arrive in SIP on *, everything is ok (negociation and phone call) but when we try to use the voicemail, Asterisk don't understand DTMF. Here are some logs (SIP debug on) on a DTMF '2' receive :

Folks, I really hope you can help me here - I'm beginning to tear my hair out! About 10 days ago my company moved to a new office. As a result of this, we've plugged our PBX box, which has happily been running for the last three years, into our new E1 line. Since then, I've been seeing intermittent yellow alarms. Obviously, since this was working fine in the old office, the thing to

Hi Everybody, Is there anyone who already uses Asterisk with TE405P card? Is this card going well with the old E400P or T400P drivers? Are there a lot of differences? In the card's data sheet writes this sentece: "Digium has designed the TE405P to be fully compatible with existing software applications..." Is this meaning the the old drivers are also OK for this card? Thank you

Hello Im from Denmark and i've just got my Digium TE405P. But i have some problems when i connect it to my E1 connection (ISDN30). My telco delivered a alcatel box witch have a G.703 120ohm (DB9 with a serial to rj45) and a 75ohm coax connection. I've tried to connect using the 120ohm with rj45 and a ordinary utp cable. But it dosent seem to work. I've tried several zaptel.conf

Salut Guy, I have the same problem with a Cirpack (B3G carrier) What I see is that you use sip info to detect DTMF. The problem is that there is no normalisation on the content of the sip info frame for dtmf detection. First, asterisk try to detect the header "application/dtmf-relay" and you have the header "application/dtmf" see line 6069 of /channels/chan_sip.c function

Salut Guy, I have the same problem with a Cirpack (B3G carrier) What I see is that you use sip info to detect DTMF. The problem is that there is no normalisation on the content of the sip info frame for dtmf detection. First, asterisk try to detect the header "application/dtmf-relay" and you have the header "application/dtmf" see line 6069 of /channels/chan_sip.c function

Hi, I am new with asterisk. I am setting a Wildcard TE405P. E1s in Italy come in on a pair of RG-59 coax cables with BNC connectors. So I need an adapter/balun http://www.allcomtlc.com/al_g703n3.htm . I have It but I am not sure that It works. I have configured my asterisk in this way: zaptel.conf span=1,1,0,ccs,hdb3,crc4 dchan=16 bchan=1-15,17-31 loadzone=it defaultzone=it zapata.conf

Hello, We have finished our tests of the new Digium firmware on the quad T1 cards(TE405P/TE410P). Overall it is a big improvement over the version 1 firmware. Here's the review: http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html MATT--- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Can please someone help me. I have successfully installed Asteriskathome 2.5 on a server with a Intel Server Board SE7525RP2. May issue is after placing the TE405P in the server, it is not booting anymore. Has anyone in here have the same experience? Can someone please point me to the right direction. Thanks in advance, Leonimar __________________________________________________ Do You

lspci dont show Digium card. The motherboard is Intel 945gnt. 00:00.0 Host bridge: Intel Corporation 945G/P Memory Controller Hub (rev 02) 00:02.0 Display controller: Intel Corporation 945G Integrated Graphics Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 1 (rev 01) 00:1c.2 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 3 (rev

Hi Group, I have an asterisk running as media gateway with a Digium TE405P 2nd Gen rev 2 with echo cancellation. It is interconnected to a telco carrier via ISDN Pri. The voice quality is clear except that sometimes a hear a beep sound that occure around 5 to 10 secs in the middle of the conversation. When I check the logs in the asterisk, I found this. Nov 30 00:48:38 DEBUG[27705]

Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as caller number. 2. A call made from a SIP

Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob.

Hi list, After a recent upgrade to Asterisk v1.4.14, my message log is now filling up with the following error messages: <-------------> [Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645 determine_firstline_parts: Bad request protocol Packet --- (1 headers 0 lines) --- bitis*CLI> <--- SIP read from 82.101.62.99:5060 ---> Cirpack KeepAlive Packet <-------------> Seeing

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