[Asterisk-Users] Multiple Digium cards?
Would any one recommend building a system with 1 TE110P, 1 TDM40B, and TDM04B? The reason I am asking is that I have a client that is migrating to asterisk and they still have a couple of FXO lines they can't dump yet. I have recommend a channel bank but they don't like that idea because of the cost. -Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy errors on WBEL4
zaptel compiles fine (make linux26) but when I modprobe ztdummy I get the follow error. [EMAIL PROTECTED] zaptel]# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy System info: Linux localhost.localdomain 2.6.9-5.0.5.ELsmp #1 SMP Thu May 19 19:24:44 CDT 2005 i686 i686 i386 GNU/Linux zaptel from June 3 2005 # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a sync source, just use "0" # # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # # The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1 # # Note: "d4" could be referred to as "sf" or "superframe" # # The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1 # # E1's may have the additional keyword "crc4" to enable CRC4 checking # # If the keyword "yellow" follows, yellow alarm is transmitted when no # channels are open. # #span=1,0,0,esf,b8zs #span=2,1,0,esf,b8zs #span=3,0,0,ccs,hdb3,crc4 # # Next come the dynamic span definitions, in the form: # dynamic=,,, # # Where is the name of the driver (e.g. eth), is the # driver specific address (like a MAC for eth), is the number # of channels, and is a timing priority, like for a normal span. # use "0" to not use this as a timing source, or prioritize them as # primary, secondard, etc. Note that you MUST have a REAL zaptel device # if you are not using external timing. # # dynamic=eth,eth0/00:02:b3:35:43:9c,24,0 # # Next come the definitions for using the channels. The format is: # = # # Valid devices are: # # "e&m" : Channel(s) are signalled using E&M signalling (specific # implementation, such as Immediate, Wink, or Feature Group D # are handled by the userspace library). # "fxsls" : Channel(s) are signalled using FXS Loopstart protocol. # "fxsgs" : Channel(s) are signalled using FXS Groundstart protocol. # "fxsks" : Channel(s) are signalled using FXS Koolstart protocol. # "fxols" : Channel(s) are signalled using FXO Loopstart protocol. # "fxogs" : Channel(s) are signalled using FXO Groundstart protocol. # "fxoks" : Channel(s) are signalled using FXO Koolstart protocol. # "sf" : Channel(s) are signalled using in-band single freq tone. # Syntax as follows: #channel# => sf:, # rxfreq is rx tone freq in hz, rxbw is rx notch (and decode) # bandwith in hz (typically 10.0), rxflag is either 'normal' or # 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone # level in dbm, txflag is either 'normal' or 'inverted'. Set # rxfreq or txfreq to 0.0 if that tone is not desired. # "unused" : No signalling is performed, each channel in the list remains idle # "clear" : Channel(s) are bundled into a single span. No conversion or # signalling is performed, and raw data is available on the master. # "indclear": Like "clear" except all channels are treated individually and # are not bundled. "bchan" is an alias for this. # "rawhdlc" : The zaptel driver performs HDLC encoding and decoding on the # bundle, and the resulting data is communicated via the master # device. # "fcshdlc" : The zapdel driver performs HDLC encoding and decoding on the # bundle and also performs incoming and outgoing FCS insertion # and verification. "dchan" is an alias for this. # "nethdlc" : The zaptel driver bundles the channels together into an # hdlc network device, which in turn can be configured with # sethdlc (available separately). # "dacs": The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon # "dacsrbs" : The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon and # also performs the DACSing of RBS bits # # The channel list is a comma-separated list of channels or ranges, for # example: # # 1,3,5 (channels one, three, and five) # 16-23, 29 (channels 16 through 23, as well as channel 29 # # So, some complete examples are: # e&m=1-12 # nethdlc=13-24 # fxsls=25,26,27,28 # fxols=29-32 # #fxoks=1-24 #bchan=25-47 #dchan=48 #fxols=1-12 #fxols=13-24 #e&m=25-29 #nethdlc=30-33 #clear=44 #clear=45 #
[Asterisk-Users] Polycom IP600 Questions
1. How do you set the music on hold to work with asterisk. Right now when I place a call on hold the caller hears nothing. MOH works with all my other IP phones. 2. Ringer Volume. How do you set the ringer volume? So that it's set on reboot. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Live Voip
I personal would not touch LiveVoIP with a 10 foot pole. They don't provide very little customer service. The service they do offer has been very bad or unusable in the past. On Wed, 2005-05-11 at 10:53, Sean Kennedy wrote: > Hi all, > > Before I setup an account with them, I'd like to hear other people's > impression of LiveVoip. I'm considering using them for 800 numbers, and > I'd like to feel comfortable that others here on the list have had good > experiences with them. > > Thanks, sorry if this is the wrong list for this. :) > > Sena > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: LiveVOIP
David, The fact of the matter is that LiveVoIP has no customer service. They don't care about small users or asterisk users. Other providers have higher prices but offer real customer service. Go look at teliax they care about customers service. On Mon, 2005-05-02 at 22:17, David Josephson wrote: > Luki writes about choppy audio with LiveVOIP. We have an almost > identical situation except that we were switched from the San Diego > gateway to the Van Nuys gateway. Some improvement but still not usable > for real customers. I have an open trouble ticket with them and no > progress. Doesn't matter whether it's MOH, IVR audio or calls; incoming > audio and DTMF dialing is fine, outgoing audio to the PSTN is choppy, at > best one dropout every 10 seconds, usually one short dropout every one > to three seconds. The comments from their tech support and CTO were that > they were aware of the problem and it was "a capacity issue" that they > were working on. There is a separate problem in that ringback tone (or > any other audio sent without answer supervision being active, > apparently) is not played to the PSTN side. This is not unique to > LiveVOIP and has been discussed (with its workarounds) before. I don't > mind their brusque attitude or the lack of user-level support, but we > won't be able to use their service if they can't fix the dropouts. There > is a lot of clatter here on the list about them not being a "real > provider" but a lot of this is sour grapes from people reselling more > expensive service. We'll see ... they don't have to be 100% facilities > based to provide good service, but they do have to fix this issue. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVOIP troubleshooting
switch to real provider On Mon, 2005-05-02 at 20:21, Luki wrote: > Hi everyone, > > I need some ideas to troubleshoot this issue: I recently got an 800 > numbers from LiveVOIP and it works but on most calls the caller gets > hears choppy audio (one drop out per 10 seconds or so). > > I know this isn't LiveVOIP's support forum but I'm sure some here use > their 800 service and I'm interested in their feedback and ideas. And > don't get me wrong, LiveVOIP's support has been quite good -- > cooperative, fast response, action taken as requested -- but I do not > want to try their patience. At this point I am not blaming them for > this issue either. > > Here's the summary: > > * I'm connected via IAX2 to > * The server is in a datacenter with plenty of bandwidth. > * Using ulaw with "standard" 20 ms frames. > * I hear the caller perfectly fine, caller hears choppy audio. > * tcpdump shows incoming and outgoing packets right on time, > every 20 ms in each direction. > * I'm not using trunking for now. > * Pings to LiveVOIP are about 35 ms. > * iax2 show channels shows 1 ms jitter, 42 ms lag. > * Drop outs occur on IVR (or audio generated on the server itself) or > during normal conversation with a SIP client (ATA or phone) connected > to the server remotely. Connection between server and phones is well > tested and working fine. > > I have asked LiveVOIP to switch me from their Vancouver node to their > New York node, which reduced ping times from 50 ms to 35 ms. Less > chops but still not perfect. > > Note that the same server is already connected to several Broadvoice > accounts, which work flawlessly. > > Anyway, if anyone has some ideas of what I can try, please let me > know. I do not want to keep trying all their nodes to find one that > works for me. I do not necessarily want to use a different codec > either since I have the bandwidth and I may be receiving faxes, so I > need ulaw. > > Thanks and sorry for the long-ish post. > --Luki > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 one way audio
Upgraded one of my asterisk servers to the latest cvs head version last nigh now I get one way audio on IAX2 channels when calling other asterisk servers. Anyone seeing the some problems? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unbelievable...
On Mon, 2005-04-18 at 12:49, Rusty Shackleford wrote: > > -Original Message- > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Rich Adamson > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Unbelievable... > > > > > > Having worked as a senior > > manager in a technical organization, a large number of > > tehcnical people simply do not comprehend some words (or read > > other words into whatever they happen to be reading), or, > > jump to conclusions based on their technical > > knowledge that are unreasonable (contractually or otherwise). > > > > The wording is very obviously oriented toward those types, > > and I'd bet a fair amount they _still_ receive calls that are > > clearly answered on their web site. > > I'm sure this is true. Users, which is to say "CUSTOMERS" can be > maddeningly clueless at times. However that is still no excuse for > bullying and threatening. Qwest and others have learned over the last > several years, and much to their dismay, that even simple indifference > to customer concerns will result in a wholesale exodus as soon as other > alternatives become available. Treating customers with the outright > contempt that LiveVoIP displays with the statement in question is, > again, staggering in it's short-sightedness. Companies like livevoip that can't / will not provide "good" customer service need to go away bottom line. It give the whole VoIP industry a bad name. > > > Regardless of what their web site says, they've provided me with the > > best service of the half dozen itsp's that I've worked with > > directly. And, I don't work for them or represent them. > > My experience with them has been likewise positive, which proves that > they are at least capable of providing good service, on occasion. The > fact that some users are frustrated to the point of posting here in this > list in order to get the attention of the company's principals, SHOULD > strike those principals as a clanging alarm that something in their > customer service system is broken. Sadly, the lessons of "Customer > Service 101" appear to have been lost on them. And that's a shame, > because as we both know, they are doing a largely good job, and it is in > everyone's interest (theirs and their cusomters', at least) that they > continue to do so. Glad to here someone has had a positive experience with them. My experience with livevoip as a resell has been less than positive. Their service does not work. Their DID's don't work right. (caller ID, DTMF, ringback, etc...) They have NO customer service. On top of that they called my customers that had DID's though them. Trying to recruit them. Just my 2cents... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Livevoip responds to DTMF via IAX issue
Brandon Patterson, Okay at this point it should be know that Livevoip. Does not support DTMF over IAX. Why not save the time and trouble and stop selling Level 3 DID's? After all the trouble that you have had with Level 3 DID's why would you even sell an unstable product? Like this. On Thu, 2005-04-07 at 20:10, The Phone Guys wrote: > LiveVoip Supports the every changing and improving Asterisk Code > for many many customers on a daily basis. In the case of the DTMF > issue we have people working on it. No estimated time to a solution. The > work continues. This looks like you are one of the 5% that we may not be > able to support. So we are happy to save you all the list time and approve a > refund. > In the future we expect to have a fix for this issue - just no date given. > > Brandon Patterson > LiveVoip LLC > > - Original Message - > From: "Brian Litzinger" <[EMAIL PROTECTED]> > To: > Sent: Thursday, April 07, 2005 6:51 PM > Subject: [Asterisk-Users] Livevoip responds to DTMF via IAX issue > > > > > > [EMAIL PROTECTED] writes: > > > >>Level 3 provides DTMF inband - IAX works out of band. > >>5% of our customers have this issue. We do not control Asterisk > >>development and we are not going to change the Level 3 setup. > > > > I'm not sure I understand what livevoip is saying here. > > > > When I ordered the service I told them I was going to be > > running an asterisk server. I even selected their 'Asterisk Plan'. > > > > So they are saying their 'Asterisk Plan' doesn't work with Asterisk? > > > > Confused. > > > > -- > > Brian Litzinger > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Liveviop problem
That is how livevoip deals with problems > /dev/null On Wed, 2005-04-06 at 15:23, Andrejus Stavickis wrote: > Hi, > > I'm just curious if someone had/has a problem with livevoip. When I try > to make an outgoing call, I receive: > -- Called :@217.160.244.186/x037378896 > Apr 2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call > rejected by 217.160.244.186: No authority found > > The , and first 5 digits of the phone is modified in > this log. > > I tried to call Livevoip, they said send us an e-mail and I did, but no > response whatsoever for about a week now. > > Sincerely, > > --Andy > x6722 > > "Outsourcing is akin to making a skyscraper taller by taking material > from its lower floors." > --Byron Katz > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Livevoip still no DTMF?
LiveVoip service sucks. Period. They don't have any real customer service. Their service is unreliable a best. They seems to change their pricing weekly. After the problems I had with them. I will never use them again. On Fri, 2005-04-01 at 13:44, Brandon Patterson wrote: > Level 3 does DTMF inband DTMF. Period. > > > > On April 1, 2005 01:44 pm, Brian Litzinger wrote: > >> Made the suggested changes. Called in via SIP and Cell Phone. Still > >> no response to DTMF. > > > > It's time to get lowlevel. > > > > iax2 debug and look for "received DTMF digit '3'" or something. tethereal > > will also show you the IAX2 IEs for DTMF. > > > > If you do not see this, the far side is not sending DTMF, and you need to > > complain to livevoip. IAX2 DTMF is *always* out of band. > > > > -A. > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-841 Call waiting?
I don't believe the SPA-841 has call waiting On Wed, 2005-03-30 at 12:41, Steve Clark wrote: > Paul Dugas wrote: > > On Tue, March 29, 2005 12:24 am, Darren Wright said: > > > >>I can call out just fine on all 4 lines, however, if I am on the line, > >>another call coming in does not ring the 2nd line...it just goes to busy > >>/ VM. > > > > > > DO you have the phone registering 4-separate time with your server? I > > initially had my 2-line unit setup that way and was getting what you > > describe until Sipura told me that both of the buttons needed to be using > > the same single line. I can get you the names of the fields to set > > tomorrow when I've got access if you'd like. > > > > Paul > > > Hi Paul, > > Were you able to get the fields you changed? I have an 2 line 841 and can't > get > call waiting to work. > > Thanke, > Steve > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small qos switch
Linksys makes a VPN router with Dual WAN interfaces and QoS http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-31672629504.htm On Fri, 2005-03-25 at 11:13, Bob Knight wrote: > I have multiple locations running * where all the phone are > on their own lan and all the data is on a separate lan. > The problem is they are sharing the same dsl connection. > The locations are IAX2 trunked together, but it only takes > one data down/up load to just kill the voice. > > What I am looking for is a small switch with QoS that I > can stick in ahead of the dsl modem. Plug in one connection > from the voice lan and one from the data lan. > > I have found quite a few 24 or 48 port switches that will do > this, but I really do not need anything that big. There are > already switches in place. > > Any recommendations please? > > thanks, bk... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wanted: sample config' using GOTOIF's for all features for a roll-out
Bon soir, * People, You guys are fantastic. This list is hyper-helpful; thanks for that. I have an organization that needs to roll-out * for each of its 30+ locations across the country & we want all features in a DB where they can be easily toggled on & off, as needed. I have done some research in the usual places & got a bit of it, but not all of it, & am hoping someone else out there might be generous enough to share with me (on or off list) a skeleton type config' that would achieve this. I have a few notes to add: + that the internal DB can handle up to approx 10 calls per second, beyond which I might start investigating using an external DB. I dont think we'll have that level of calling. + that a DB via ODBC might not be required until we have serious strain on the DB + need sample DB tables & fields + sample skeleton config' (so we dont flood the list, either offlist or www.pastebin.ca) + that AGI might be overly-complicated + that it might make sense to use GOTOIF's everywhere possible in the dialplan to be able to flexibly enable features for each caller, each destination, each client, each + that there are approx' 20 main features (ie: call waiting, moh, intl dialing, local dialing, n-digit dialing, voicemail, forwarding, etc, etc, etc) any client might want. (I havent really counted. Any one know of a good list of core features that a given client might want.) + does any one have a list or URL that lists the core features that a given client might want in a typical intl business type setting. I count 65 features on asterisk.org features page. Any and all replies & pointers & sample config's & URL's are so very much appreciated. I hope I can return the favor to the list soon. Spread peace. Bobby Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
