similar to: Sip registration Timers

Hello, Does anyone know what may be causing this? Asterisk was built from cvs tonight. Ztcfg also says us is an invalid tone zone. Anyone got some information on what this is, why is it happening, and possibly some solutions? [root@anistonetech zapata]# asterisk -f -d DEBUG[1024]: File config.c, Line 653 (__ast_load): Parsing /etc/asterisk/asterisk.conf DEBUG[1024]: File config.c, Line 653

Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

Like most others on this list I also have some really annoying echo whenever a call goes out to the PSTN from a SIP phone... SNOM/Budgettone -> Asterisk -> X100P -> PSTN I have tried every echo canceler in the makefile and turned on and off aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and I can get it reduced to only a few seconds on the intro of the call and

Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

Everyone, Just thought I would drop a line telling everyone here I have the software RxFAX/TxFAX up and running without any real problems. I did have to..... RH 9.0 1) Install an audio devel rpm 1) install libtiff from source, and copy over a bunch of include files to /usr/local/include 2) build/install spandsp 3) move app_rxfax.c and app_txfax.c to apps/ dir in asterisk source tree. 4)

Hello, I resolved my echo issue using grandstream/estara etc etc sip phones and wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl motherboard for an asus p4pe? i845? based motherboard and the echo has completly gone away along with aggressive suppressor option in the makefile. I hope this helps others. Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH

Hi, We are looking at consolidating our lines with PRI. This will allow the elimination of many fax lines. Some of them will be replaced with this type of config ... PRI * IAX * Channel-Bank FAX We will have daggressor suppressor enabled. Is anyone doing this and should I expect smooth operation? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.

Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody

I'm wondering if anyone knows of ISPs with service that has QoS features that would be good to use with VoIP stuff. Granted, the QoS would only be supported as long as you stayed within their network, but it might be better than nothing. --Eric

I have a simple problem with sialing a SIP device. I'm SURE it's a syntax problem, but I dunno what it might be. Here are the debug messages: == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) -- Executing Goto("Zap/1-1", "2111|1") in new stack -- Goto (default,2111,1) -- Executing Dial("Zap/1-1",

To the list ... I got the IAX2 stuff simplified & working (for now). See my earlier posting to the list. Now, here's a question for you all. I found a posting by J Todd where he gives BW utilization for various IAX2 codecs with trunking on. Now, the number of calls I can sustain over an IAX channel, obviously is going to be determined by the capacity and state of the physical pipe.

Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I

Now that Solaris 10 11/06 is available, I wanted to post the complete list of ZFS features and bug fixes that were included in that release. I''m also including the necessary patches for anyone wanting to get all the ZFS features and fixes via patches (NOTE: later patch revision may already be available): Solaris 10 Update 3 (11/06) Patches sparc Patches * 118833-36 SunOS 5.10:

Folks! I discovered some serious problem with several Sipuras 3000 but I don't know if the problem is with them or Asterisk. Basically, if I call a Sipura PSTN line, when there is a call already in progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I am able to get through and connect to dialed number. The other call gets disconnected but the originator of the

Responses should be copied to r-help using ReplyAll. You are still sending html formatted emails. If you are using Microsoft Outlook, click the Format Text tab and select ?Aa Plain Text?. No one has asked you to reveal the data set, only to create one with a similar structure. Is the data I sent reasonably close? What should it look like after it is transformed? David C From: nguy2952

No html!, Copy the list using Reply-All. The data frame group_PrivateLabel does not contain variables called Product_Name or Region. David C From: nguy2952 University of Minnesota <nguy2952 at umn.edu> Sent: Friday, June 1, 2018 2:13 PM To: David L Carlson <dcarlson at tamu.edu> Subject: Re: [R] Regroup and create new dataframe Hi David, your example is perfect! I am still

Hi Derek, Here''s the latest email I''ve received from the zfs-discuss alias. ------------- Begin Forwarded Message ------------- Date: Mon, 18 Sep 2006 23:55:27 -0400 From: Jonathan Edwards <Jonathan.Edwards@sun.com> Subject: Re: [zfs-discuss] ZFS and HDS ShadowImage To: Eric Schrock <eric.schrock@sun.com> Cc: zfs-discuss@opensolaris.org, Torrey McMahon

Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN

Hi, The code in Support/Timer.cpp has strangely inconsistent behavior for printAll vs printJSONValues. The former can work multiple times, while the latter calls prepareToPrintList(), which stops all timers, hence making all further attempts to print timers crash. Would it be possible not to call prepareToPrintList on printJSONValues, or at least make it optional? I am trying to serialize

Hi all, I've installed the current debian nut package, i.e. version 2.0.4. I've configured upssched with timers as follows: AT ONBATT black at localhost START-TIMER runonbattery 5 AT ONLINE black at localhost CANCEL-TIMER runonbattery AT ONLINE black at localhost START-TIMER backonline 5 AT ONBATT black at localhost CANCEL-TIMER backonline The idea is to send alerts in case of