similar to: Digium TE405P dtmf issue

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

Good Morning, I've recently gotten Asterisk installed and configured our IVR using FreePBX. Things seem to be going well except a few of our inbound callers are ending up in the wrong place when trying to connect to a specific extension. The example I had this morning was someone trying to call extension 212 and getting connected to the Sales queue which is option 2 on the IVR. I looked in

I just upgraded my asterisk install from january 2005 CVS HEAD to current CVS HEAD : * zaptel * libpri * asterisk My asterisk have one TE405P with one span (the clock source) pluged into a telco PRI E1, a second span is a PRI E1 to another PBX and the third one is a T1 to a rhino channel bank. Everything went smoothly except that the two PRI span restart every ten seconds after some HDLC

Hi, we have a E1 pri from Citylink, (they are using Ericsson Engine exchange), that are restarting after 5 - 15 minutes, before and after that we can make calls in and out w/o problems. The cards have been tested in two computers (Atholon XP 2200+ and Celeron 2.6Ghz), are on there own IRQ, not showing any IRQ misses in zttool. It's all correctly configuered - zaptel and zapata (attached). We

I have TE405P span connected to a patton/digi 2977 T1/PRI port. Zaptel starts up fine without any errors. Libpri is installed as well. But I get the following when trying to load asterisk ------- Jul 3 19:30:26 ERROR[8192]: Signalling requested is PRI Signalling but line is in Unknown signalling 896 signalling Jul 3 19:30:26 ERROR[8192]: Unable to register channel '25-48' Jul

Hi, I'm really lost on the directory functionality of the asterisk. I am currently setting a ACD using asteriskathome. I want to use the # feature where in caller can access the companies directory. My issue is how to setup the entry on the directory. Sorry for my silly question Can somebody guide me please. Regards, Leonimar Cape __________________________________________________ Do You

Hello, I'm running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem I'm having is that when I pick up any

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, 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:

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, I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch (www.cirpack.com). <IP Network>--<*>--<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 tryed to

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, I tried so many but can't find the right setting for my problem. What do i have to configure so that the complete number including extension is displayed at the called party. At the moment the called party only sees the number 7837-0 not the 7837-134. Everything works fine. Incoming and Outgoing calls. Is there someone who configured a german pri with that digium card? I really

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

Hello all, I really checked voip-info.org but it still seems to be not very easy and I just hope that there is anybody with a simular config. We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper than a channel bank :-) Carrier ----S2M------ * -----S2M------Siemens | |

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

Hello, Can somebody please assist me with my problem. Currently I am using a Asterisk@HOme version 2.4 with a TE406P digium card. One the E1 is connected to a telco switch via an ISDN. May issue is that may incoming calls in the zap channels gets disconnected or release after 1 ring. I really dont know what setting should I change to increase the timeout of the ring. I have even tried upgrading

I understand from the archives that * does this occassionally, but I'm trying to figure out why. * didn't do this at all for two days, and then it's gone and done it 3 times in the past hour. It does not seem to be affecting calls, I'm just curious as to the reasoning behind the B channel resets and why they are so erratic. Regards, Andrew

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

The Panasonic KX-TA624 series PBXes (and similar models) support a DTMF integration feature that can be enabled for dedicated voice mail ports. What I want to do is connect an X100P FXO port to a jack on the Panasonic and make use of the Panasonic's DTMF call progress tones that it provides in DTMF integration mode. The integration works well when a Panasonic extension is forwarding into