similar to: SNMP instrumentation and/or talk path health monitoring?

Displaying 20 results from an estimated 3000 matches similar to: "SNMP instrumentation and/or talk path health monitoring?"

2005 Mar 22
1
Is there a way to get inserted into an LEC's CLIDB?
> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the
2004 Sep 19
2
Effectively using a telco Type 102 Milliwatt Test line with ztmon itor -v to set txgain/rxgain in zapata?
I am trying to obtain optimum gain settings for a bank of analog lines connected to a channel bank. My telco has provided a 'Type 102' test line to use for incoming level calibration. This is functionally equivalent to app Milliwatt(), but provides tone from the CO inwards. Question is, how should one use this a 0dbm test source with ztmonitor? Am I correct in understanding that a 0dbm
2004 Aug 31
1
Why is it called 'Comedian Mail?
Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District
2005 Aug 11
1
PRI dropped calls w/ asterisk dropped betweenpstn & norstar
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Gary Reuter > Sent: Thursday, August 11, 2005 12:59 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped > betweenpstn & norstar > > > I poured over my logs most of
2004 Sep 07
3
Maximum tollerable lag/jitter for IAX2 w/o jitterbuffer enabled?
I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from
2004 Aug 19
6
How to run different codecs between the same endpoints on an IAX trunk?
Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like
2004 Aug 27
3
Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?
Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is
2004 Aug 27
0
Hangup() doesn't always when talking to Nortel Norstar over CT1 E &M wink-start trunk line?
I've noticed a problem with calls to Hangup when talking to my Norstars over channelised T-1 E&M trunk lines - it's been present since I started to fiddle with Asterisk last December and it's still present in 'Asterisk CVS-HEAD-08/13/04-10:37:13'. Specifically, when a call is connected to Asterisk from the Norstar DTI card to my T100p I get the following conditions
2004 Sep 16
2
No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS?
I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)
2004 Aug 19
1
Inband announcement of parking slot from app _parkandannounce?
Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from
2004 Sep 07
4
Maximum tollerable lag/jitter for IAX2 w/oji tterbuffer enabled?
> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >
2004 Aug 29
1
Bridging audio in cmd_dial() before connect completes?
Is it possible to make cmd_dial() bridge the audio going out to the network back to the calling party as soon as dial() starts? Put another way, is it possible to have the caller hear the outside dialtone and subsequent DTMF digits? I notice that there is an option 'r' to dial(), thus: r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one
2004 Sep 07
1
Monitored outbound dialing via Zap interface?
I'm using a T100p to interface to a channel bank and from there to analog PSTN lines. Because of my particular setup I have to do post-connect inband DTMF dialing - which takes up to 5 seconds for a 10 digit number, assuming 0.5/sec per digit (ie. using "zap/g1/31|5|D(6045551212)". Even with an 'outside transfer' voice prompt before commencing dialing my users are getting
2004 May 28
0
Problem with digits blending on inbound pulsed digits?
I have a situation where I am receiving DID calls using Immediate Start Pulse signalling on a Loop Start trunk. The line terminates on a Newbridge Mainstreet 3624 channel bank, which provides battery etc. The channel is converted and routed to Asterisk. The lines are configured as follows: /etc/asterisk/zapata.conf ; Channels 1-24 service MainStreet 3624 channel bank context=infrom-did group=1
2004 Jun 03
0
Preserving received digits during a fax match?
I have a set of analog DID lines coming into my Asterisk box, via a channel bank. The numbers in the DID bank route to various places, including voice lines of various staff. I am using the fax detection engine to intercept faxes accidentially sent to numbers on the DID bank and reroute them to a physical fax set up in the office. I would now like to preserve the received digits and pass them
2004 Jul 16
0
Transmitting a hook-flash down an E&M DS-0?
I'm trying to access feature codes remotely over a channelised T1 between a Norstar MICS (rev 4) and Asterisk. The timeslots are configured E&M and have been working fine under most circumstances except this one. There is mention of accessing the facility by calling Flash() from within extensions.conf, but I can't get it to work... Right now I can't tell if it's because
2004 Aug 11
0
Inband announcement of parking slot from app_parkandannounce?
I'm trying to use Asterisk app_parkandannouce to build a global parking pool from within a couple of Norstar PBXes. Right now I can blind transfer calls into the parking lot, but the slot announcement relies on calling back the 'transferee' after the call is parked and I can't pass enough callerid data out from within the PBX to be able to route the call back in (ie. no PRI
2004 Aug 27
0
'set verbose 3' or other way to get '-vvv' level debugging out of running background asterisk?
I'm having a dialplan problem on one host where trunks get pinned up flapping between 't' and 'i' states and start eating lots and lots of CPU (loadavg > 4.00). I haven't been able to pin down the problem reading through extensions.conf and test calls haven't caught it yet either. Unfortunatly the offending trunks are FXO immediate start DID trunks so subsequent
2004 Sep 08
0
T100P calls with playback starts speaking be fore pickup
> -----Original Message----- > From: Jerry Geis [mailto:geisj@pagestation.com] > Sent: September 8, 2004 2:19 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] T100P calls with playback starts speaking > before pickup > > > Hi > > I am using a T100P connected to a panasonic phone switch using T1 and the > switch has 4 analog lines
2004 Oct 06
0
Can Asterisk provide Answer Supervision signalling to a channel b ank via T1?
I have an older Newbridge Mainstreet 3624 upon which I'm terminating some analog DID lines. They are effectively loop-start trunks with battery supplied by me (ie. FXO) and consumed by the serving central office. One major part of DID is the requirement for providing Answer Supervision in the form of battery polarity reversal on the analog trunks. Without it wierd things start happening, like