jnovak@logisticshealth.com
2005-Feb-11 08:28 UTC
[Asterisk-Users] RE:mandrake linux install of zaptel
Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of asterisk-users-request@lists.digium.com Sent: Friday, February 11, 2005 1:37 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 7, Issue 168 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-owner@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: Searchable Mailing Lists & NooB Question (Rich Adamson) 2. RE: TelIAX troubles (Scott Bussinger) 3. Re: Asterisk not acceptingmultiple SIP phone logins (Juki) 4. Re: Asterisk not accepting multiple SIP phone logins (Juki) 5. RE: dtmfmode and IAX protocol (Rich Adamson) 6. RE: dtmfmode and IAX protocol (Michael Giagnocavo) 7. Re: Why echo occurs (Rich Adamson) 8. RE: dtmfmode and IAX protocol (Rich Adamson) 9. Re: Why echo occurs (Steven Critchfield) 10. RE: Searchable Mailing Lists & NooB Question (Ed Guy) 11. Re: Why echo occurs (Steve Underwood) 12. Re: Why echo occurs (Steven Critchfield) 13. RE: Zombie SIP channels (Florian Overkamp) 14. Re: Why echo occurs (Steve Underwood) ---------------------------------------------------------------------- Message: 1 Date: Thu, 10 Feb 2005 23:32:01 -0600 From: Rich Adamson <radamson@routers.com> Subject: Re: [Asterisk-Users] Searchable Mailing Lists & NooB Question To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <Chameleon.1108100640.adar0@vegas> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 Looks like your numbers add the transmit and receive data rates together, which is not a realistic way to discuss bandwidth consumption. An IAX link consumes about 22kb/s (round it to 30kb/s, who cares) in the transmit direction, and another 22kb/s in the receive direction. (There's your 60kb/s.) When comparing my numbers to things like 256,000 bits/sec of DSL bandwidth, you truly are comparing apples to apples. So, if you could orchestrate all IAX calls to be just exactly perfect across the 256,000 bits/sec DSL bandwidth, that DSL circuit could supposedly handle about eight simultanous gsm calls (256,000 divided by 30,000). However, there are lots of other real world issues that would preclude it from actually supporting anything close to eight calls. Four to six might be realistic if nothing else is using the DSL circuit. ------------------------> > > > > >Very rough numbers: iax-gsm consumes about 22kb/s, > > > > I see about 60kb/s > > > g711 about 80kb/s on > > > > > I see 155kb/s > > Is that normal? This is an IAX link to voicepulse. I see all these lower > numbers posted around but fail to see that on my connections. Using G711, > Its only possible to have one connection at anytime, do to my upload > capped at 256kb/s. So I use GSM, sounds fine anyway. Just wondering about > the numbers. > > > Dan > > >same link unless you can set up QoS, etc. > > > >Lots of good info on the wiki ( www.voip-info.org ) for reference. > > > > > >_______________________________________________ > >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---------------End of Original Message----------------- ------------------------------ Message: 2 Date: Thu, 10 Feb 2005 21:44:05 -0800 From: "Scott Bussinger" <scottb@opto-pps.com> Subject: RE: [Asterisk-Users] TelIAX troubles To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <f0502102148410582@local> Content-Type: text/plain; charset="us-ascii" We're just getting our Asterisk server setup with TelIAX and it's working fine. I did have to play with settings a bit. Basically I just used the setting they recommended instead of the generic settings I started with. Here are the significant settings we're using in IAX.CONF: [general] disallow=all allow=gsm register=username:password@voip.teliax.com [teliax] type=friend context=tollfree host=voip.teliax.com auth=md5 secret=password Good Luck! ------------------------------ Message: 3 Date: Fri, 11 Feb 2005 08:44:52 +0300 (EAT) From: "Juki" <juki@one2net.co.ug> Subject: Re: [Asterisk-Users] Asterisk not acceptingmultiple SIP phone logins To: "Roger Gulbranson" <roger@gulbranson.com> Cc: asterisk-users@lists.digium.com Message-ID: <3255.81.199.88.27.1108100692.squirrel@mail.one2net.co.ug> Content-Type: text/plain;charset=iso-8859-1 Sorry, I had omitted the second SIP phone details but they do exist and are at different SIP addresses as shown below; My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201 callerid="101" <2125> nat=yes [102] type=friend host=dynamic username=102 secret=test1 dtmfmode=rfc2833 context=from-sip mailbox=202 callerid="102" <2135> nat=yes My extensions.conf has: exten => 101,1,Dial(SIP/101,20,tr) exten => 101,2,VoiceMail,u101 exten => 101,102,VoiceMail,b101 exten => 102,1,Dial(SIP/102,20,tr) exten => 102,2,VoiceMail,u102 exten => 102,102,VoiceMail,b102 My voicemail.conf has: 101 => 2348,Emma, kidjue@yahoo.co.uk 101 => 2348,juki, juki@one2net.co.ug How do I proceed from here then?> This is not a -dev question. It should only be posted to -users. > > On Thu, 2005-02-10 at 22:15 -0700, Juki wrote: >> Hi all, >> >> I have Asterisk running on FreeBSD 4.x and I have made configurations to >> sip.conf, extensions.conf and voicemail.conf. I have also setup SIP >> phones >> on two different PCs. My problem is that when one of the SIP phones >> logins >> in, the other won't. >> >> My sip.conf has: >> [101] >> type=friend >> host=dynamic >> username=101 >> secret=test >> dtmfmode=rfc2833 >> context=from-sip >> mailbox=201 >> callerid="101" <2125> >> nat=yes >> >> My extensions.conf has: >> exten => 101,1,Dial(SIP/101,20,tr) >> exten => 101,2,VoiceMail,u101 >> exten => 101,102,VoiceMail,b101 >> >> My voicemail.conf has: >> 101 => 2348,Emma, kidjue@yahoo.co.uk >> >> Any ideas are most welcome. > > I see only one address here. If you have multiple phones at the same > address, only the last to be registered will be recognized. > > Put each PC/phone at a different sip address. > > > >-- Rgds, Juki. ------------------------------ Message: 4 Date: Fri, 11 Feb 2005 08:49:12 +0300 (EAT) From: "Juki" <juki@one2net.co.ug> Subject: Re: [Asterisk-Users] Asterisk not accepting multiple SIP phone logins To: "Steven Critchfield" <critch@basesys.com> Cc: asterisk-users@lists.digium.com Message-ID: <3259.81.199.88.27.1108100952.squirrel@mail.one2net.co.ug> Content-Type: text/plain;charset=iso-8859-1 Sorry, I had omitted the second SIP phone details but they do exist and are at different SIP addresses. How do I fix my dial to dial both at the same time? How do I proceed from here then?> This is not a development question. It is a user question. From the > snippit it looks like you are trying to have both phone log in as the > same user. That is why I think you are asking a user question. Make > separate accounts for each phone and fix your dial to dial both at the > same time. > > Then, next time, unless you are quoting C code, you probably don't need > to post it here. > > On Fri, 2005-02-11 at 08:07 +0300, Juki wrote: >> Hi all, >> >> I have Asterisk running on FreeBSD 4.x and I have made configurations to >> sip.conf, extensions.conf and voicemail.conf. I have also setup SIP >> phones >> on two different PCs. My problem is that when one of the SIP phones >> logins >> in, the other won't. >> >> My sip.conf has: >> [101] >> type=friend >> host=dynamic >> username=101 >> secret=test >> dtmfmode=rfc2833 >> context=from-sip >> mailbox=201 >> callerid="101" <2125> >> nat=yes >> >> My extensions.conf has: >> exten => 101,1,Dial(SIP/101,20,tr) >> exten => 101,2,VoiceMail,u101 >> exten => 101,102,VoiceMail,b101 >> >> My voicemail.conf has: >> 101 => 2348,Emma, kidjue@yahoo.co.uk >> >> Any ideas are most welcome. >> > -- > Steven Critchfield <critch@basesys.com> > >-- Rgds, Juki. ------------------------------ Message: 5 Date: Thu, 10 Feb 2005 23:50:43 -0600 From: Rich Adamson <radamson@routers.com> Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <Chameleon.1108101058.adar0@vegas> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 No. ------------------------> Can the Sipura SPA-3000 do IAX? > -Michael > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Joseph > Sent: Thursday, February 10, 2005 10:50 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol > > Actually, I don't know what might be the problem. > I'm using Sipura SPA-3000 unit connected to standard cordless phone > and connecting to FWD over IAX > > 1.) > If I call FedEx or Bank and enter my account number using numeric keys > it works > > 2.) > If I dial UPS 1-800-742-5877 and try to use one of the option provided > it doesn't work. > > Could it be their phone system? > > -- > #Joseph > > On Thu, 2005-02-10 at 21:36 -0600, Michael Giagnocavo wrote: > > Actually, there are some phones that will do inband DTMF over IAX2. Soif> > he's using one of these, he has to make sure his settings are correct. > Yes, > > the PA168 phones. The correct setting is RFC2833 for IAX (inside these > > phones). Otherwise it's inband. The other options they provide just cut > the > > call. > > > > -Michael > > > _______________________________________________ > 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---------------End of Original Message----------------- ------------------------------ Message: 6 Date: Fri, 11 Feb 2005 00:01:25 -0600 From: "Michael Giagnocavo" <mgg-digium@atrevido.net> Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <A6B3F6B9F0204802BBCF10F6854EC.MAI@colo901.fullcontrol.net> Content-Type: text/plain; charset="us-ascii" Exactly. (I was hoping he'd come to his own conclusions.) So... if the Sipura does not do IAX, then it's quite possible that you're not doing IAX on the Sipura. Which means the whole "dtmfmode and IAX protocol" is moot... -Michael -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich Adamson Sent: Thursday, February 10, 2005 11:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol No. ------------------------> Can the Sipura SPA-3000 do IAX? > -Michael > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Joseph > Sent: Thursday, February 10, 2005 10:50 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol > > Actually, I don't know what might be the problem. > I'm using Sipura SPA-3000 unit connected to standard cordless phone > and connecting to FWD over IAX > > 1.) > If I call FedEx or Bank and enter my account number using numeric keys > it works > > 2.) > If I dial UPS 1-800-742-5877 and try to use one of the option provided > it doesn't work. > > Could it be their phone system? > > -- > #Joseph > > On Thu, 2005-02-10 at 21:36 -0600, Michael Giagnocavo wrote: > > Actually, there are some phones that will do inband DTMF over IAX2. Soif> > he's using one of these, he has to make sure his settings are correct. > Yes, > > the PA168 phones. The correct setting is RFC2833 for IAX (inside these > > phones). Otherwise it's inband. The other options they provide just cut > the > > call. > > > > -Michael > > > _______________________________________________ > 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---------------End of Original Message----------------- _______________________________________________ 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 ------------------------------ Message: 7 Date: Fri, 11 Feb 2005 00:00:46 -0600 From: Rich Adamson <radamson@routers.com> Subject: Re: [Asterisk-Users] Why echo occurs To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>, Eric Bishop <asterisk.eric@gmail.com> Message-ID: <Chameleon.1108102351.adar0@vegas> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1> 1. Is the echo (regardless of it's speed) a side effect of long > distance communications or is it there by design for some technical > purpose?Echo is not there by design (with one small exception noted below). Echo frequently is the result of imperfections in the two-wire to four- wire hybrid. All analog phones are two-wire devices, but most electronic central offices and long distance facilities are essentially four-wire. The Internet, ISDN, SIP and IAX protocols are essentially four-wire. Each point where that two-wire to four-wire conversion takes place, some amount of echo is likely to result. Commercial echo cancelling hardware usually does a good job of removing the echo. So, you might originate a call using a SIP phone, transport that call via ISDN, but if the called end is an analog phone then a hybrid exists at that point and could create some echo. You'll find more detail on the wiki. All phones (analog and digital) feed some of your transmit audio back into the earpiece, and that is called 'sidetone'. That is sort of an intended echo. ------------------------------ Message: 8 Date: Fri, 11 Feb 2005 00:17:49 -0600 From: Rich Adamson <radamson@routers.com> Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <Chameleon.1108102986.adar0@vegas> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 Joseph has been working at bringing up an asterisk box as kind of a newbie, and I think he's using a Sipura as his fxs interface into asterisk. He's having a problem with asterisk passing dtmf to FWD, but didn't say how he's accessing the bank or fedex. So, without a fair amount more detail from him, there's no way to answer his questions or guess at the problem. ------------------------> Exactly. (I was hoping he'd come to his own conclusions.) So... if the > Sipura does not do IAX, then it's quite possible that you're not doing IAX > on the Sipura. Which means the whole "dtmfmode and IAX protocol" ismoot...> > -Michael >-------------------------> No. > > ------------------------ > > > Can the Sipura SPA-3000 do IAX? > > -Michael > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Joseph > > Sent: Thursday, February 10, 2005 10:50 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol > > > > Actually, I don't know what might be the problem. > > I'm using Sipura SPA-3000 unit connected to standard cordless phone > > and connecting to FWD over IAX > > > > 1.) > > If I call FedEx or Bank and enter my account number using numeric keys > > it works > > > > 2.) > > If I dial UPS 1-800-742-5877 and try to use one of the option provided > > it doesn't work. > > > > Could it be their phone system? > > > > -- > > #Joseph > > > > On Thu, 2005-02-10 at 21:36 -0600, Michael Giagnocavo wrote: > > > Actually, there are some phones that will do inband DTMF over IAX2. So > if > > > he's using one of these, he has to make sure his settings are correct. > > Yes, > > > the PA168 phones. The correct setting is RFC2833 for IAX (inside these > > > phones). Otherwise it's inband. The other options they provide justcut> > the > > > call. > > > > > > -Michael > > > > > > _______________________________________________ > > 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 > > ---------------End of Original Message----------------- > > > _______________________________________________ > 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---------------End of Original Message----------------- ------------------------------ Message: 9 Date: Fri, 11 Feb 2005 01:25:33 -0600 From: Steven Critchfield <critch@basesys.com> Subject: Re: [Asterisk-Users] Why echo occurs To: Eric Bishop <asterisk.eric@gmail.com>, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <1108106734.11927.68.camel@critch> Content-Type: text/plain On Fri, 2005-02-11 at 15:35 +1100, Eric Bishop wrote:> 2. Is only a problem in 2-wire technologies (ie analog and BRI ISDNlines)? It is just an analog problem. That is why a BRI can actually transmit direct digital data instead of audio data.> 3. Where exactly is the slowdown occuring? For example take my Supira > 3000 as a case in point. It takes no longer for the PSTN signal to > reach the Sipura's FXO port than it does my $5 handset. Going from the > other end it takes no longer for the SIP signal to reach to the > Sipura's ethernet port than it does any other IP phone. So logically > the slowdown is happening as Sipura converts the PSTN signal to SIP > and so forth. Is it just that the Sipura/TDM400 etc. have a too slow > conversion CPU. Would a faster digital to analogue audio converter > "fix" the the problem?Steve Underwood pointed out that most Telco equipment has a max delay of 3 samples. On your Sipura, you will have a initial packetization delay of 160 samples. And that is if there isn't any compression work time. You can bet that no matter what it never takes longer than half the time to receive the samples to compress them as it is likely a symmetrical codec and the other half of the time is decoding the incoming stream too. Next, if the hop from the Sipura to your PBX takes half a millisecond, you still add the equivalent of 4 samples of delay. If you have asterisk doing any work on the link, you add more delay dependent on speed of system and amount of work done. If you route very far, you add more delay. For example, on a point to point T1 link where it only traveled 20 miles or so and was not congested, I still saw another 3-4ms of delay for a ping. So figure 1.5-2 ms for each side of the hop. So figure another 12-16 samples of delay. Of course out my cable modem and up to my office asterisk machine is showing 46.9ms average round trips. So easily I am looking at a full voice packet in transit while the next one is being created. And I'm only 14 hops away all on the AT&T network. So if normal toll quality calls have no more than 3 samples delay, you are looking at a minimum of 164 samples delay on VoIP and possibly more than 330-340 samples. 110 times slower than what the telcos would use. -- Steven Critchfield <critch@basesys.com> ------------------------------ Message: 10 Date: Fri, 11 Feb 2005 02:02:45 -0500 From: "Ed Guy" <edguy@pulver.com> Subject: RE: [Asterisk-Users] Searchable Mailing Lists & NooB Question To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com>, "Geoff Scott" <geoff.scott.mail@gmail.com> Message-ID: <GOECLHPNBNEOAEDCDJKNAEKHFDAA.edguy@pulver.com> Content-Type: text/plain; charset="iso-8859-1" The notation is confusing, but 32kBs (KBytes/s) is the same as 256kbs (kbit/sec)! Of course, I'm assuming this is what Geoff meant. I've had to rate limit some file transfers to prevent interference with the voice channels. Your experience will depend on how much the web and jabber servers are used. /ed -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Max Klein Sent: Thursday, February 10, 2005 8:38 PM To: Geoff Scott; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Searchable Mailing Lists & NooB Question I would guess that your DSL upstream is more like 256k or 320k, not 32k. Could you confirm this with your provider? I have 256k up right now and can have about 3 calls max with ulaw, or quite a few more with GSM (these are both CODECs used to COmpress and DECompress the audio). --Max On Thu, 2005-02-10 at 17:30, Geoff Scott wrote:> On Thu, 10 Feb 2005 19:11:38 -0600, Steven Critchfield > <critch@basesys.com> wrote: > > > > You are joking right? You think you are going to do any voice over a > > link that is half of the bandwidth of a phone call and you think you > > will have a webserver and jabber server on it. > > > > Even using GSM codec, you will probably only get 1 call to work when > > nothing else is working. Last I checked FWD only accepted G729 and ulaw. > > You will never get ulaw across that link and you will have to purchase a > > G729 license. > > -- > > Steven Critchfield <critch@basesys.com> > > > > _______________________________________________ > > 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 > > > > I'm new to *. Hence the question. If it's a bother, no need to hit > the reply button. > > >From your answer then, am I to assume no one is running * servers on a > standard DSL line? > > gs_______________________________________________ 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 ------------------------------ Message: 11 Date: Fri, 11 Feb 2005 15:05:50 +0800 From: Steve Underwood <steveu@coppice.org> Subject: Re: [Asterisk-Users] Why echo occurs To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <420C594E.6010205@coppice.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Steven Critchfield wrote:>On Fri, 2005-02-11 at 15:35 +1100, Eric Bishop wrote: > > >>2. Is only a problem in 2-wire technologies (ie analog and BRI ISDNlines)?>> >> > >It is just an analog problem. That is why a BRI can actually transmit >direct digital data instead of audio data. > >There is enough spill from the earpiece to the mike on most phones, that EC is required even on a digital phone. Regards, Steve ------------------------------ Message: 12 Date: Fri, 11 Feb 2005 02:02:05 -0600 From: Steven Critchfield <critch@basesys.com> Subject: Re: [Asterisk-Users] Why echo occurs To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <1108108925.11927.84.camel@critch> Content-Type: text/plain On Fri, 2005-02-11 at 15:05 +0800, Steve Underwood wrote:> Steven Critchfield wrote: > > >On Fri, 2005-02-11 at 15:35 +1100, Eric Bishop wrote: > > > > > >>2. Is only a problem in 2-wire technologies (ie analog and BRI ISDNlines)?> >> > >> > > > >It is just an analog problem. That is why a BRI can actually transmit > >direct digital data instead of audio data. > > > > > There is enough spill from the earpiece to the mike on most phones, that > EC is required even on a digital phone.Fine, but to get an earpiece, you make an analog portion of the link unless someone has made some digital ears with direct data jacks on the side of human heads. So if you say that a SIP handset is like a 4 wire set, and BRI is like a 4 wire set, and asterisk doesn't mix the ins and outs, you effectively are 4 wire through the portion you can control. The remote side is up to whoever you call. -- Steven Critchfield <critch@basesys.com> ------------------------------ Message: 13 Date: Fri, 11 Feb 2005 08:32:43 +0100 From: "Florian Overkamp" <florian@obsimref.com> Subject: RE: [Asterisk-Users] Zombie SIP channels To: "'Pedro'" <traci.asterisk@gmail.com>, "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <E1CzVIj-00007Q-00@clio> Content-Type: text/plain; charset="US-ASCII" Hi,> -----Original Message----- > Ok this is odd - caught it again twice today. The more I thought > about what has changed on the server I realized that I was not using a > timing device before, but am now using ztdummy. I if that could be > causing the zombies?> > > http://bugs.digium.com/bug_view_page.php?bug_id=0002938I don't think so, but who knows. The patch resolves a locking issue that may or may not be timing-source dependant. I've seen the issue occur after call transfers in scenario's where I used a few chan_local's. Do yourself a favour: - If you can, unload the ztdummy and test for a while. However, this may put the issue to sleep - but it won't solve it! - After that, load ztdummy again and apply the two lines in channel.c. Test again. Good chance the issue will be gone. Report results here :) Florian ------------------------------ Message: 14 Date: Fri, 11 Feb 2005 15:32:39 +0800 From: Steve Underwood <steveu@coppice.org> Subject: Re: [Asterisk-Users] Why echo occurs To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <420C5F97.3070203@coppice.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Steven Critchfield wrote:>On Fri, 2005-02-11 at 15:05 +0800, Steve Underwood wrote: > > >>Steven Critchfield wrote: >> >> >> >>>On Fri, 2005-02-11 at 15:35 +1100, Eric Bishop wrote: >>> >>> >>> >>> >>>>2. Is only a problem in 2-wire technologies (ie analog and BRI ISDNlines)?>>>> >>>> >>>> >>>> >>>It is just an analog problem. That is why a BRI can actually transmit >>>direct digital data instead of audio data. >>> >>> >>> >>> >>There is enough spill from the earpiece to the mike on most phones, that >>EC is required even on a digital phone. >> >> > >Fine, but to get an earpiece, you make an analog portion of the link >unless someone has made some digital ears with direct data jacks on the >side of human heads. > >So if you say that a SIP handset is like a 4 wire set, and BRI is like a >4 wire set, and asterisk doesn't mix the ins and outs, you effectively >are 4 wire through the portion you can control. The remote side is up to >whoever you call. > >What you said was not actually wrong. However, 9 out of 10 people reading it will see "echo is something that affects only analogue phones". People keep saying this. Its even in comments in the * source code. Its wrong. Regards, Steve ------------------------------ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 7, Issue 168 ********************************************** --- [This E-mail scanned for viruses by Declude Virus] --- [This E-mail scanned for viruses by Declude Virus]
On Feb 11, 2005, at 16:28, <jnovak@logisticshealth.com> wrote:> Extreme N00b, I am getting the error message "a target does not exist" > when > running the make install inside the zap directory, probably pretty > common, > possibly a package I didn't install, just need some insight on it. The > same > occurs with the libpri and asterisk.I think everyone would appreciate if... - you wrote a new mail instead of highjacking an existing thread by answering it and replacing the subject line - you would not keep 5 miles of completely unrelated stuff in your email message - you could provide a better problem description that includes specific error messages and message stacks. Thanks! jens