Re: [asterisk-users] I can hear my own voice through the headset
On Thursday 04 Oct 2012, frangky robert wrote: > Here is my IP-PBX setupmy setup is : sips softphone <-> asterisk <-> > xorcom PSTN gateway <-> pstn line to telcoi'm using xlite for > windows when I make a phone call (sip - outgoing channel),I can hear > my own voice so clear. it's very annoying mewhen talking a little > loud... any solution? Thanks, We've often faced this problem with SIP soft phones when the computer's sound system gain was set too high. You usually have to play around with microphone gain settings to get to the point where the echo disappears with the other party still being able to hear you. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10.9.0-rc1 : Help with GoSubIf Parsing
At 07:02 PM 10/4/2012, you wrote:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1(${thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
= 2024324321
Try :
same=n,GoSubIf($["${CALLERID(num)}" = "2024324321"]?other,1(${thisexten}):)
The quotes make sure it doesn't fail on an empty callerid.
Ira
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[asterisk-users] username ignored when trying to auth incoming invites
Hello All, I am trying to debug an odd issue. I have two UACs that are sending INVITEs to my asterisk 1.8 server. I want to start authenticating these incoming invite requests with digest auth. The UACs are not registered and I am using host ip to match them with a sip.conf peer. The issue I am seeing is that an incoming invite matches a specific peer (by host ip), but refuses to use the "username" parameter value for digest auth, it will only use the peer name. I see the following error: "chan_sip.c: username mismatch, have , digest has " I have the following sip.conf: [node-a] type=friend disallow=all allow=ulaw context=incoming-context host=XXX.XXX.XXX.XXX transport=udp username=test secret=1234 [node-b] type=friend disallow=all allow=ulaw context=incoming-context host=YYY.YYY.YYY.YYY transport=udp username=test secret=1234 If I auth using "node-a" as the username when sending an invite from that host, everything works. If I auth with "test" as the username from node-a, it fails with the error above. It appears that peer name is always being used for digest auth, rather than the contents of username. Is "username" the wrong place to specify this? Thanks for your help! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 10.9.0-rc1 : Help with GoSubIf Parsing
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1(${thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
= 2024324321
^
[Oct 4 21:53:35] WARNING[11356]: ast_expr2.fl:472 ast_yyerror: If you
have questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [2124531143@from-teliax-sip:3] GosubIf("DAHDI/1-1",
"?other,1(2124531143):") in new stack
I've tried with and without spaces the = sign. Same Result. I've
counted my parens and braces.
Any help really appreciated!
sean
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Re: [asterisk-users] Fax for Asterisk success rates?
2012/10/4 Brett Lehrer > >What is the setup you're talking about ? > >Is it something like this ? > >PSTN nexVortex T.38 gateway - Internet - DSL modem --- > >Asterisk Fax machine > Olivier, > > Sorry, I did a poor job explaining that. That's basically correct, with > the receiving end first and our originating end last in your diagram. For > outgoing faxes only, this is the setup: > > Fax interface (LAN website, in short) -> Asterisk PBX -> DSL modem -> > Internet -> nexVortex trunk -> [recipient] > > Incoming faxes are generally more reliable, but I still get small number > of failures. I've mistakenly overestimated the incoming failure rate. > Don't have clean statistics on that, though. > How many fax and voice calls (which codecs for tha latter ones ?) are on average using your DSL line ? 1. Previously, I experienced failures during the process of converting incoming PDF documents into ready-to-send fax image files while the reverse process (from a fax file into a PDF or whatever document) never failed. I would be curious to check if a greater failure rate for outbound faxing (greater than inbound faxing failure rate) could simply comes from image processing, before any transmission. 2. Though your DSL line may have enough bandwidth from your location to its DSLAM, chances are packets are dropped or delivered too late for T.38 faxing. An interesting test would be to use an Asterisk PBX hosted somewhere at "close range" from netVortex fax gateways : that would remove most networking issues out of the equation. > Unexplainable FAX call failures (i.e. not wrong numbers of other >obviously wrong things) should be well below 1%. On a dedicated DSL >line, if everything is set up properly you should be getting that kind >of rate. This is especially true if you are using T.38 and the provider >at the far end uses a decent T.38 platform. Across the open internet >results are much more variable. >Depending what causes your 25% failures, you may get better results with >spandsp than with FFA. >Steve > I see, thanks. All of these faxes are going out to unknown, external > machines. I have no control over anything on their ends, and the > hardware/connection is as variable as you could imagine. I'll definitely > look into SpanDSP. FWIW, the dedicated DSL line is just a 6 Mbps up/768 > Kbps down Internet connection that is solely used by our in-house PBX to > connect to the trunk. > > > >However I'd just suggest that you look at the business case for screwing > around with fax at all. > Oh man, if only... I'd LOVE to just drop fax completely and use email > instead. > > Brett Lehrer > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LDAP Driver and VoiceMail
Hello: I am investigating the possibility of using LDAP for storing certain Asterisk configuration parameters. I have examined res_ldap.conf and see where mailbox can be defined from AstAccountMailbox but I do not see where the password can be stored ? Am I missing something please ? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
From: "Carlos Alvarez" Sent: Thursday, October 04, 2012 1:18 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Fax for Asterisk success rates? On Thu, Oct 4, 2012 at 10:06 AM, Lee Howard wrote: I recognize that you're being a bit facetious in this latter comment No, not really. I stand by it. Useless and *should* be dead. It's dead and people just don't know it.There is no adequate replacement for fax. E-mail doesn't do it Yes, it does. Well, if you were using stand-alone fax machines then that was part of your problem. That was actually the only part of my post that was in jest. -- Carlos Alvarez TelEvolve 602-889-3003 Fax has been a long road in the VOIP arena and asterisk. For T.30 & T.38 to ATA gateways you need the right mix of equipment at both ends. Vendors that support T.38 or PRI's with good T.38 supported hardware gateways work best. On the fax gateway side Steve, and spandsp are god sent. Fax works well when you get your karma in alignment you must set it up correctly. Our systems have processed over 500,000+ faxes this year with very few fax machine compatibility issues. From a technology standpoint I too look forward to the day where we can get rid of faxes, but from a business perspective I am happy to process faxing for our paying customers. Fax + Asterisk can work quite well. Bryant Zimmerman (ZK Tech Inc/interNetGR) (616) 855-1030 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "Call me now" outbound calls in a queue
On Fri, Sep 28, 2012 at 7:42 PM, Mitch Claborn wrote: > I want to put a "call me now" button on the web site that will place the > request into an asterisk call queue and then when an agent picks up the > call in the queue, place the outbound call to the customer. > > The following AMI command works, but it calls the customer first, before > an agent is necessarily available. > > Action: Originate > Channel: SIP/voipms/customer_number_**here > Context: external > Async: true > Application: Queue > Data: sales > Callerid: Company <8005551212> > > How can I get an available agent before the customer call is placed? > > Hello Mitch, Hoping that the Queue application is not automatically Answering the line (till an agent will do this) my suggestion is to switch between "who have to answer" in order to progress to the second call leg. This means that the Queue will be called through a Local Channel and the call to your customer will be made through a Dial application. Below is something to start with - in case it will work you could modify to your needs. [demo] exten => s,1,NoOp(Queue without answer) exten => s,2,Queue(sales) Action: Originate Channel: Local/s@demo/n Application: Dial Data: SIP/voipms/customer_number HTH, Ioan Indreias Modulo Consulting // www.modulo.ro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I can hear my own voice through the headset
> Here is my IP-PBX setupmy setup is : sips softphone <-> asterisk <-> xorcom > PSTN gateway <-> pstn line to telcoi'm using xlite for windows > when I make a phone call (sip - outgoing channel),I can hear my own voice so > clear. it's very annoying mewhen talking a little loud... any solution? Two questions: (1) Does the problem occur when you make a SIP-to-SIP call, without the PSTN being involved? (2) When you hear your own voice in the headset, is it delayed, or is just an immediate louder-than-you-want "side-tone"? If it *does* occur in SIP-to-SIP calls, this would rule out your XORCOM and the PSTN as the cause. If it's only occurring in SIP-to-PSTN calls, then the XORCOM and PSTN (or the interaction between them) is a likely suspect. There are several things which can cause this sort of problem. (A) Direct acoustic feedback within the headset. In this case, you'd probably hear it even if the headset was unplugged entirely. The only cure is to buy a better headset. (B) Incorrect audio-mixer settings in your PC. To the PC audio infrastructure, a headset usually "looks like" a microphone and a separate speaker. The audio mixer (hardware and software) usually has an ability to mix some of what the microphone "hears" into the speaker output. If this "knob" is turned up too high, you'll hear your own voice too loudly. If too low, you won't hear your own voice at all when you speak into the headset, and many people find this lack of side-tone to be confusing. The cure here is to adjust the audio side-tone level, either in your Windows audio-mixer control panel, or in X-Lite (if it has such an adjustment). (C) Electrical "reflection" from an analog impedance discontinuity in the analog telephone-line system. This can result from a mismatch between the telephone wiring, and the PSTN interface device, and can occur at any point in the analog transmission. If the loud side-tone you hear is *not* delayed noticeably, then the impedance mismatch might be at your XORCOM/PSTN interface. The XORCOM may have a software adjustment or jumper setting, to match its audio impedance to that of your local phone line... try fiddling with these settings to see if they reduce the excessive side-tone level. If the loud side-tone you hear is delayed (it sounds a bit like an echo) then it may very well be at the "far end" of the phone line, outside of your own physical control... it might be at your local phone office, or anywhere between you and the far end of the phone connection. Not much you can do about this. (D) Audio feedback at the far end of the call, in a cheap phone handset. Sometimes, audio from the "back side" of the speaker in a handset travels through the body of the handset and is picked up by the microphone, and results in an audible delayed "echo" of the voice from the far end of the line. Using a better handset, or stuffing the handset full of audio damping material (cloth or cotton or fiberglass) is the cure here. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect calls : known reasons
On 04-10-12 19:50, Chad Wallace wrote: On Fri, 28 Sep 2012 11:03:05 +0200 Jonas Kellens wrote: On 28-09-12 10:57, Administrator TOOTAI wrote: Le 28/09/2012 10:40, Jonas Kellens a écrit : Maybe I need to explain a bit further : the call is send to the IP-phone and answered. The call lasts for about 1 à 2 minutes and is then disconnected. We had this problem with some PSTN call termination providers, sometimes only against some destination. I don't know if your incoming calls are 100% VOIP, I would start to see with providers. You may also check hangupcause and dialstatus variables. Pure SIP. Hangupcause 16 Dialstatus Answer It has nothing to do with the provider-side. You could narrow it down by inspecting the SIP packets for the call in question (using wireshark or Asterisk sip debugging) and seeing which end issues a BYE packet--if either one does. Also, typically you only have contact with one end of a call (your users) so it's very hard to say that something didn't happen on the other end (somewhere out in the wild, where people drive through tunnels). It is Asterisk that sends the BYE. I wouldn't know why. It has nothing to do with tunnels and so... Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect calls : known reasons
On Fri, 28 Sep 2012 11:03:05 +0200 Jonas Kellens wrote: > On 28-09-12 10:57, Administrator TOOTAI wrote: > > Le 28/09/2012 10:40, Jonas Kellens a écrit : > >> Maybe I need to explain a bit further : the call is send to the > >> IP-phone and answered. The call lasts for about 1 à 2 minutes and > >> is then disconnected. > > > > We had this problem with some PSTN call termination providers, > > sometimes only against some destination. I don't know if your > > incoming calls are 100% VOIP, I would start to see with providers. > > > > You may also check hangupcause and dialstatus variables. > > Pure SIP. > > Hangupcause 16 > > Dialstatus Answer > > It has nothing to do with the provider-side. You could narrow it down by inspecting the SIP packets for the call in question (using wireshark or Asterisk sip debugging) and seeing which end issues a BYE packet--if either one does. Also, typically you only have contact with one end of a call (your users) so it's very hard to say that something didn't happen on the other end (somewhere out in the wild, where people drive through tunnels). PS, Sorry for the late reply... I haven't checked the list in a week. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call extension play sound file then connect caller
> >> I am trying to setup a context to take a inbound call, hold the > >> call, > >> connect to > >> an external number, play a sound file to the external number, then > >> connect > >> the inbound caller to the external number. > >> > >> My thought was to accept the call and place them in a parking lot. > >> Then > >> call > >> the external number, play the sound file and connect the inbound > >> caller > >> to > >> the external number. > >> > >> > >> Is this even possible and if so, is this the best approach? > >> > >> > >> Thank you in advance. > >> > > > > You might look into FollowMe, especially if you want the external > > number > > to have a choice of whether or not to accept the call. > > > > A very high level overview is here: > > http://answers.oreilly.com/topic/2705-asterisk-how-to-use-followme-to-call-a-series-of-phone-numbers/ > > (though that gave me enough to get started) > > > > > Thanks for the reply. I tried using FollowMe as it seemed like the > perfect > solution, however I was unable to play the sound file then connect > the > caller. I would like to bypass the need to press the 1 to accept the > call. Have you tried the Dial application M or U options? Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On Thu, Oct 4, 2012 at 10:06 AM, Lee Howard wrote: > I recognize that you're being a bit facetious in this latter comment > No, not really. I stand by it. Useless and *should* be dead. It's dead and people just don't know it. > There is no adequate replacement for fax. E-mail doesn't do it > Yes, it does. > > Well, if you were using stand-alone fax machines then that was part of > your problem. > That was actually the only part of my post that was in jest. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
>What is the setup you're talking about ? >Is it something like this ? >PSTN nexVortex T.38 gateway - Internet - DSL modem --- >Asterisk Fax machine Olivier, Sorry, I did a poor job explaining that. That's basically correct, with the receiving end first and our originating end last in your diagram. For outgoing faxes only, this is the setup: Fax interface (LAN website, in short) -> Asterisk PBX -> DSL modem -> Internet -> nexVortex trunk -> [recipient] Incoming faxes are generally more reliable, but I still get small number of failures. I've mistakenly overestimated the incoming failure rate. Don't have clean statistics on that, though. > Unexplainable FAX call failures (i.e. not wrong numbers of other >obviously wrong things) should be well below 1%. On a dedicated DSL >line, if everything is set up properly you should be getting that kind >of rate. This is especially true if you are using T.38 and the provider >at the far end uses a decent T.38 platform. Across the open internet >results are much more variable. >Depending what causes your 25% failures, you may get better results with >spandsp than with FFA. >Steve I see, thanks. All of these faxes are going out to unknown, external machines. I have no control over anything on their ends, and the hardware/connection is as variable as you could imagine. I'll definitely look into SpanDSP. FWIW, the dedicated DSL line is just a 6 Mbps up/768 Kbps down Internet connection that is solely used by our in-house PBX to connect to the trunk. >However I'd just suggest that you look at the business case for screwing >around with fax at all. Oh man, if only... I'd LOVE to just drop fax completely and use email instead. Brett Lehrer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On 10/04/2012 09:27 AM, Carlos Alvarez wrote: However I'd just suggest that you look at the business case for screwing around with fax at all. As a society, if we had decided to stop supporting this dead technology years ago, with all the time and money we've collectively wasted we could have completely eliminated world hunger. I recognize that you're being a bit facetious in this latter comment, but the argument that you're making here is unfounded. I believe that if you were to look at the Davidson Consulting reports about the fax industry for as long as those reports have been available you'd find this. The technology is not dead and has enough momentum to propel it forward for many years to come. Maybe this is understood in your acknowledgement of "society" supporting it, but the reason why it's supported is because the technology is sound and fills a very valuable purpose in business and other activities. There is no adequate replacement for fax. E-mail doesn't do it, and most other reliable document communication mechanisms are locked-up in proprietary patents and interests that will invariably prevent them from becoming standardized at all. I'm not a big T.38 fan-boy, although I do applaud the ITU for that attempt to get fax working on IP networks. Unfortunately, it's fundamentally flawed because it needlessly perpetuates the tether between fax and telephony. In an IP network there is no reason whatsoever for fax to be saddled on top of a telephony layer. Fax is data communication, and IP networks are quite effective at data communication. I can envision a future fax system which truly uses modern IP network designs such as DNS, encryption, security, and rides on a very effective communication protocol and yet continues to operate on the fundamental communication protocol defined in ITU T.30 which makes well-implemented faxing so dependable. I can't count the hundreds of hours I've wasted on fax support just to prop up this stupid and unnecessary technology. Many others have felt exactly the same way, and I don't mean to be rude, but invariably the reason why they feel this way is because they repeatedly tried to do it the wrong way. We just made the decision this week to outsource it all and never deal with it on our network again. I am slowly re-gaining my sanity because of that decision. And until the new technology comes along that is and will be *precisely* the right decision for most of the people who move to a virtual environment or who completely detach themselves directly from the PSTN. Now I'm going to take a fax machine out to the parking lot and shoot it, even talking about this awful waste of time makes my blood boil. Well, if you were using stand-alone fax machines then that was part of your problem. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On Thu, Oct 4, 2012 at 6:29 AM, Brett Lehrer wrote: > I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking > service over a DSL line solely dedicated to VoIP usage. For both incoming > and outgoing faxes, I'm getting a failure rate of just over 25%, and over a > handful of reasons. > > Is it natural to have this many problems on a completely digital > configuration? I'm trying to cut our analog phone line (because it's so > expensive), but some fax machines just don't seem to ever accept a fax. > Many of the failures are on the same numbers, forcing me to fall back to > an old analog fax machine just to make sure it actually gets through. > > Has anyone else had any similar experiences, or is this indicative of a > failure in the setup on my end (or even the trunking service)? > I'm not going to address the tech issues, as others already have. And if you didn't know, Steve Underwood is THE fax guy so whatever he says is gold, listen to him. However I'd just suggest that you look at the business case for screwing around with fax at all. As a society, if we had decided to stop supporting this dead technology years ago, with all the time and money we've collectively wasted we could have completely eliminated world hunger. I can't count the hundreds of hours I've wasted on fax support just to prop up this stupid and unnecessary technology. We just made the decision this week to outsource it all and never deal with it on our network again. I am slowly re-gaining my sanity because of that decision. Now I'm going to take a fax machine out to the parking lot and shoot it, even talking about this awful waste of time makes my blood boil. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
2012/10/4 Brett Lehrer > I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking > service over a DSL line solely dedicated to VoIP usage. For both incoming > and outgoing faxes, I'm getting a failure rate of just over 25%, and over a > handful of reasons. > > Is it natural to have this many problems on a completely digital > configuration? I'm trying to cut our analog phone line (because it's so > expensive), but some fax machines just don't seem to ever accept a fax. > Many of the failures are on the same numbers, forcing me to fall back to > an old analog fax machine just to make sure it actually gets through. > > Has anyone else had any similar experiences, or is this indicative of a > failure in the setup on my end (or even the trunking service)? > > Brett Lehrer > What is the setup you're talking about ? Is it something like this ? PSTN nexVortex T.38 gateway - Internet - DSL modem --- Asterisk Fax machine > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Why are alarm transmitter said to be avoided with DSL ?
Hi, 1. I've got this question bouncing in my mind for a long time: why are alarm transmitters often said to be avoided with DSL lines ? The kind of alarm transmitters I'm thinking about are those having two analog ports: one connected to Telco analog line, the other to a fax or a terminal or whatever. The typical setup is: Telco line ADSL splitter - Alarm Transmitter - Fax machine | | ADSL Modem - LAN For me, this setup MUST work as I'm not aware of any Alarm Transmitter using the higher frequencies used by modems. Is this correct ? Opinions ? 2. Another question, from a location I've got trouble with, is: what may happen if an alarm transmitter is incorrectly installed before the ADSL splitter like this ? Telco line - Alarm Transmitter ADSL splitter - Fax machine | | ADSL Modem - LAN Would it explain why ADSL traffic is momentarily cut when a fax is received, for instance ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On which kernel version the DAHDI-2.0 release will work ??
On Thu, Oct 04, 2012 at 07:40:57PM +0530, upendra wrote: > Hi, > > > Can any one tell me on which linux kernel version i can compile and run the > DAHDI-2.0 release and test it . Probably 2.6.18 is your best bet. I'm genuinly curious as to why you would want to run a version of dahdi from four years ago? -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On 10/04/2012 09:29 PM, Brett Lehrer wrote: I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking service over a DSL line solely dedicated to VoIP usage. For both incoming and outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful of reasons. Is it natural to have this many problems on a completely digital configuration? I'm trying to cut our analog phone line (because it's so expensive), but some fax machines just don't seem to ever accept a fax. Many of the failures are on the same numbers, forcing me to fall back to an old analog fax machine just to make sure it actually gets through. Has anyone else had any similar experiences, or is this indicative of a failure in the setup on my end (or even the trunking service)? Brett Lehrer Unexplainable FAX call failures (i.e. not wrong numbers of other obviously wrong things) should be well below 1%. On a dedicated DSL line, if everything is set up properly you should be getting that kind of rate. This is especially true if you are using T.38 and the provider at the far end uses a decent T.38 platform. Across the open internet results are much more variable. Depending what causes your 25% failures, you may get better results with spandsp than with FFA. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AVAILSTATUS always 0
On 04-10-12 16:59, Danny Nicholas wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, October 04, 2012 9:48 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] AVAILSTATUS always 0
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten => s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten => s,n,NoOp(availstatus = ${AVAILSTATUS})
${SIPPEERNAME} = sip username from realtime Mysql database.
Kind regards,
Jonas.
Could be a realtime issue. What do the other 3 variables return?
-= Info about application 'ChanIsAvail' =-
[Synopsis]
Check channel availability
[Description]
This application will check to see if any of the specified channels are
available.
This application sets the following channel variables:
${AVAILCHAN}: The name of the available channel, if one exists
${AVAILORIGCHAN}: The canonical channel name that was used to create the
channel
${AVAILSTATUS}: The device state for the device
${AVAILCAUSECODE}: The cause code returned when requesting the channel
Thank you for your answer. Below the information :
[Oct 4 17:03:26] -- Executing [s@sub-CheckNetworkProblems:6]
ChanIsAvail("SIP/SipAgenT01-54e1", "SIP/tech0") in new stack
[Oct 4 17:03:26] -- Executing [s@sub-CheckNetworkProblems:7]
NoOp("SIP/SipAgenT01-54e1", "AVAILSTATUS = 0") in new stack
[Oct 4 17:03:26] -- Executing [s@sub-CheckNetworkProblems:8]
NoOp("SIP/SipAgenT01-54e1", "AVAILCHAN = SIP/tech0-54e2") in new
stack
[Oct 4 17:03:26] -- Executing [s@sub-CheckNetworkProblems:9]
NoOp("SIP/SipAgenT01-54e1", "AVAILORIGCHAN = SIP/tech0") in new stack
[Oct 4 17:03:26] -- Executing [s@sub-CheckNetworkProblems:10]
NoOp("SIP/SipAgenT01-54e1", "AVAILCAUSECODE = 0") in new stack
Kind regards,
Jonas.
--
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Re: [asterisk-users] AVAILSTATUS always 0
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, October 04, 2012 9:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AVAILSTATUS always 0
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten => s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten => s,n,NoOp(availstatus = ${AVAILSTATUS})
${SIPPEERNAME} = sip username from realtime Mysql database.
Kind regards,
Jonas.
Could be a realtime issue. What do the other 3 variables return?
-= Info about application 'ChanIsAvail' =-
[Synopsis]
Check channel availability
[Description]
This application will check to see if any of the specified channels are
available.
This application sets the following channel variables:
${AVAILCHAN}: The name of the available channel, if one exists
${AVAILORIGCHAN}: The canonical channel name that was used to create the
channel
${AVAILSTATUS}: The device state for the device
${AVAILCAUSECODE}: The cause code returned when requesting the channel
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AVAILSTATUS always 0
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten => s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten => s,n,NoOp(availstatus = ${AVAILSTATUS})
${SIPPEERNAME} = sip username from realtime Mysql database.
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
Brett Lehrer wrote: Hola, I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking service over a DSL line solely dedicated to VoIP usage. For both incoming and outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful of reasons. I've never heard of that service so I unfortunately don't know the underlying equipment they are using for their service but it makes a *huge* difference. T.38 (which I hope you are using as that increases the chances a bit more) implementations wildly differ in interoperability and how well they generally work. This is one of the big problems with doing fax over VoIP and why for some individuals it works great and why for others it just falls apart. Is it natural to have this many problems on a completely digital configuration? I'm trying to cut our analog phone line (because it's so expensive), but some fax machines just don't seem to ever accept a fax. Many of the failures are on the same numbers, forcing me to fall back to an old analog fax machine just to make sure it actually gets through. In a perfect world, no. In reality it depends as I mentioned above. Has anyone else had any similar experiences, or is this indicative of a failure in the setup on my end (or even the trunking service)? Without more information (like logs/etc) it's hard to isolate things and point fingers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
I had the same problem for a while. I found replacing fax machines with a scanner and either an email-to-fax program or just web-based faxing had better results. I don't want to tell you the gateway I used because they turned out pretty badly in the end. But there is hope! - Logan On Oct 4, 2012 8:29 AM, "Brett Lehrer" wrote: > I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking > service over a DSL line solely dedicated to VoIP usage. For both incoming > and outgoing faxes, I'm getting a failure rate of just over 25%, and over a > handful of reasons. > > Is it natural to have this many problems on a completely digital > configuration? I'm trying to cut our analog phone line (because it's so > expensive), but some fax machines just don't seem to ever accept a fax. > Many of the failures are on the same numbers, forcing me to fall back to > an old analog fax machine just to make sure it actually gets through. > > Has anyone else had any similar experiences, or is this indicative of a > failure in the setup on my end (or even the trunking service)? > > Brett Lehrer > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] On which kernel version the DAHDI-2.0 release will work ??
Hi, Can any one tell me on which linux kernel version i can compile and run the DAHDI-2.0 release and test it . *Regards Upendra.* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax for Asterisk success rates?
I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking service over a DSL line solely dedicated to VoIP usage. For both incoming and outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful of reasons. Is it natural to have this many problems on a completely digital configuration? I'm trying to cut our analog phone line (because it's so expensive), but some fax machines just don't seem to ever accept a fax. Many of the failures are on the same numbers, forcing me to fall back to an old analog fax machine just to make sure it actually gets through. Has anyone else had any similar experiences, or is this indicative of a failure in the setup on my end (or even the trunking service)? Brett Lehrer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer blocking CDR and recording?
I am sorry, all my fault ): I used click2dial and set a wrong outgoing line - snom lines count there lines starting at 1 instead of 0. how could they only dare to do so... 2012/10/3 Tim Nelson > > - Original Message - > > No idea? ): > > How about showing your dialplan, and the log or console output from when > you make the call? I have a hard time believing this number is special in > any way... > > --Tim > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call extension play sound file then connect caller
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Gary Carr Sent: Wednesday, October 03, 2012 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call extension play sound file then connect caller I am trying to setup a context to take a inbound call, hold the call, connect to an external number, play a sound file to the external number, then connect the inbound caller to the external number. My thought was to accept the call and place them in a parking lot. Then call the external number, play the sound file and connect the inbound caller to the external number. Is this even possible and if so, is this the best approach? Thank you in advance. You might look into FollowMe, especially if you want the external number to have a choice of whether or not to accept the call. A very high level overview is here: http://answers.oreilly.com/topic/2705-asterisk-how-to-use-followme-to-call-a-series-of-phone-numbers/ (though that gave me enough to get started) Thanks for the reply. I tried using FollowMe as it seemed like the perfect solution, however I was unable to play the sound file then connect the caller. I would like to bypass the need to press the 1 to accept the call. Thanks Again! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
