similar to: T1 signaling configuration

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate => yes' without success, but it

Hello, I'm trying to set the 3rd span of a new digium quad card as a E&M T1 for Faxes to a Hylafax server. The 1st and 2nd spans are working as PRIs. When I start asterisk, the logs show a signaling error and chan_zap.c dies. I also get an error that it can't read the gains but they are the standard shown below. 2.6 kernel, Debian Stable, * 1.2 svn from feb 2007 my procedure: make

HI, I'm trying to config a span from port 2 of a DIgium T410p to a Brooktrout TR114-P8V-T1 card. I have a T1-PRI from the TELCO in port 1 (thru first port) working just fine with Asterisk 1.0.3 - been working fine for some time now. No problem with dialplan PSTN calls. Now I'd like to "route" specific DID numbers in thru the TELCO T1-PRI and out thru the second port (span) on

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system

SBC installed a T1 ADTSe (Digital Trunking Service Enhanced) e&m wink start with 24 1 way trunks. The CO says they dial pulse DP the seven digit dnis number. The channels work now but take long time to answer and get these messages repeating until I guess the CO stops Pulse dialing the number. Nov 23 19:08:58 WARNING[1827865]: chan_zap.c:4718 ss_thread: getdtmf on channel 8: Operation now

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

Howdy, Just installed a new switch in a new location (Ubuntu, 2.6.24-24 kernel, zaptel 1.4.12.1 built from source, libpri-1.4.10.1 built from source, asterisk 1.4.26 built from source, wanpipe 3.5.4 built from source, Sangoma A104d with firmware that is probably a year old). I plugged in an RBS T1, ESF, B8ZS, wink start, and MF signalling. I stuck with the defaults that the wanpipe build

Hi, I am trying to address some error messages that are hitting the log files for two 4.9.5-Debian file servers in our all-Samba AD domain. Most prominently "connect to service Profiles initially as user MYDOMAIN\tc-mj00y2ps$ (uid=11128, gid=10515) (pid 1634)" "../source3/smbd/uid.c:453(change_to_user_internal)" "change_to_user_internal: chdir_current_service()

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

On 02/01/2020 05:30, Mike Ruebner via samba wrote: > Hi, > > I am trying to address some error messages that are hitting the log files > for two 4.9.5-Debian file servers in our all-Samba AD domain. Most > prominently > > "connect to service Profiles initially as user MYDOMAIN\tc-mj00y2ps$ > (uid=11128, gid=10515) (pid 1634)" >

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

Hi All, I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its connected directly to the PSTN. But I am unable to make outbound calls on the zap channels. The light on the card is green. Asterisk CLI shows all 24 channels when I give the command 'zap show channels'. I also noticed that Asterisk CLI shows an incoming call every few seconds on the 24th channel. This must be

Hello, I have a T1 (not PRI) plugged into my Asterisk server. Everything is working well, except I only get the first digit of the 4 digit DID in Asterisk. The T1 provider (Eschelon) tried switching to 7 digits, and I only got the first digit of the 7. Can anybody help? We're adding another DID and I need to trap it correctly. Thanks. Here are snippets from the relevant files: --

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

I have a system setup with two T1 circuits, the customer complains of call quality issues. I've checked logs and don't see anything too strange, but when I run "dahdi show status" I see: Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO T2XXP (PCI) Card 0 Span 1 OK 0 0 19 ESF B8ZS YEL 0 db

Could anyone with any experience with * over a non-PRI T1 help this newbie? I have a fractional T1 that is working fine through a channel bank, but I can't get any response on * using a T400P. My analog line and the extensions work fine, but the T400P port 1 shows a red led (what exactly does that tell me?). I brought up zttool which shows a condition red also, but ztcfg shows nothing amiss.