similar to: Digium Card: Echo, Echo and more Echo

I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [root@asterix root]# modprobe zaptel [root@asterix root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel

On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's

Helo, I have problem with Digium TE110P connected to CISCO 3640 (port on NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I tried with real PBX problem was exactly same. I have this messages in Asterisk conole and log sometimes: NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Usually 2-5 such messages in series, can be repeated after 10

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/0efacc79/attachment.htm

We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS

Hi, I have a problem I will describe. I have PAP2 connected to the internet to an asterisk box with 2 TDM cards, one TE100P E1 with PRI and one TDM400P with 2 FXS an one FXO. When I call to the TDM400 cards from the PAP2 eveything is OK, sound quality is perfect. When I call to terminate the call in PSTN through E100P I hear clicks which aparently are RTP packet looses. This clicks are only heard

Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with "lspci -vb" command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same

Hi it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply a E1 link to a old PABX ? Thanks

Hello, i've googled and can't find a definite answer, so here goes: I have purchased the Digium TE100P, and am setting up the connection, however the telco i'm supposed to work with does not support PRI/ISDN E1 connections. They only support E1/R2 lines. Is there a way i can make the TE100P work with this? I've not seen any zaptel.conf that supports this. Any workarounds?

Hi all, So I'm trying to work on this complex fax server setup, and part of it involves connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100% of the way there. When dialing the fax extension from my

Hi all, Can someone give me a simple rational explanation why a $5 analog handset gives me no echo whatsoever on an analog PSTN line, but PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require software-based echo cancellation. Surely a $5 analog handset does not have an "echo canceller". The echo I mean is when I hear myself while talking to another party. I have heard

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Hi, I have a Gateway running in "TE" (terminal equipment mode as "slave" that I need to connect to my asterisk server using a TE100P card. Can anybody give a quick run up of how to run the TE100P's in Network Termination mode to have this working sucessfully? Cheers! Regards, Sahil Gupta VoiceValley

On Fri, Apr 27, 2018 at 11:45:02AM -0400, Kevin Easton wrote: > The struct vhost_msg within struct vhost_msg_node is copied to userspace, > so it should be allocated with kzalloc() to ensure all structure padding > is zeroed. > > Signed-off-by: Kevin Easton <kevin at guarana.org> > Reported-by: syzbot+87cfa083e727a224754b at syzkaller.appspotmail.com Is this patch going

On Fri, Apr 27, 2018 at 11:45:02AM -0400, Kevin Easton wrote: > The struct vhost_msg within struct vhost_msg_node is copied to userspace, > so it should be allocated with kzalloc() to ensure all structure padding > is zeroed. > > Signed-off-by: Kevin Easton <kevin at guarana.org> > Reported-by: syzbot+87cfa083e727a224754b at syzkaller.appspotmail.com Is this patch going

I've been playing with the outgoing call spooling feature a bit lately and it all works as it should with the exception of one irritation. I'm mostly using SIP to talk to the phones and using G.723.1 I copy the call file into the spool/outgoing directory and the originating phone rings. I pick it up and the remote phone rings. However there is dead silence from the originating

From: "Michael S. Tsirkin" <mst at redhat.com> Date: Fri, 27 Apr 2018 19:02:05 +0300 > There's a 32 bit hole just after type. It's best to > give it a name, this way compiler is forced to initialize > it with rest of the structure. > > Reported-by: Kevin Easton <kevin at guarana.org> > Signed-off-by: Michael S. Tsirkin <mst at redhat.com>

From: "Michael S. Tsirkin" <mst at redhat.com> Date: Fri, 27 Apr 2018 19:02:05 +0300 > There's a 32 bit hole just after type. It's best to > give it a name, this way compiler is forced to initialize > it with rest of the structure. > > Reported-by: Kevin Easton <kevin at guarana.org> > Signed-off-by: Michael S. Tsirkin <mst at redhat.com>

When talking to me, people are complaining the volume was not high enough. The phone only allows to change the volume of the speaker/earpiece. Is there an alternative solution? Is it possible to increase the volume in asterisk? Frederic

Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)