Re: [Asterisk-Users] Why echo occurs
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 ISDN lines)? 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zombie SIP channels
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=0002938 I 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 ___ 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] Why echo occurs
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 ISDN lines)? > >> > >> > > > >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 <[EMAIL PROTECTED]> ___ 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] Why echo occurs
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 ISDN lines)? 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 ___ 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] Searchable Mailing Lists & NooB Question
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 > <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> > > > > ___ > > 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 ___ 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] Why echo occurs
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 ISDN lines)? 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 <[EMAIL PROTECTED]> ___ 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] dtmfmode and IAX protocol
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" is moot... > > -Michael > - > No. > > > > > Can the Sipura SPA-3000 do IAX? > > -Michael > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] 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 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 > > > ___ > 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
Re: [Asterisk-Users] Why echo occurs
> 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. ___ 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] dtmfmode and IAX protocol
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 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 ___ 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] dtmfmode and IAX protocol
No. > Can the Sipura SPA-3000 do IAX? > -Michael > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 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
Re: [Asterisk-Users] Asterisk not accepting multiple SIP phone logins
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, [EMAIL PROTECTED] >> >> Any ideas are most welcome. >> > -- > Steven Critchfield <[EMAIL PROTECTED]> > > -- Rgds, Juki. ___ 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] Asterisk not acceptingmultiple SIP phone logins
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, [EMAIL PROTECTED] 101 => 2348,juki, [EMAIL PROTECTED] 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, [EMAIL PROTECTED] >> >> 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. ___ 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] TelIAX troubles
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:[EMAIL PROTECTED] [teliax] type=friend context=tollfree host=voip.teliax.com auth=md5 secret=password Good Luck! ___ 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] Searchable Mailing Lists & NooB Question
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- ___ 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] Bri problem
Good day all I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension "s"?? Is this something to do with the telecoms provider or a asterisk config? Please Help ore advice Thanks Altus ___ 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] dtmfmode and IAX protocol
Can the Sipura SPA-3000 do IAX? -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 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
[Asterisk-Users] [Asterisk-Dev] Asterisk not accepting multiple SIP phone logins
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, [EMAIL PROTECTED] Any ideas are most welcome. -- Rgds, Juki ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] Asterisk not accepting multiple SIP phone logins
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, [EMAIL PROTECTED] Any ideas are most welcome. -- Rgds, Juki ___ 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] Why echo occurs
Hi Eric, Before VoIP and digital cellular systems, audio samples passed down the communication path with little more delay than the pseed of light. A couple of samples delay might be incurred in switching equipment, but most operators demand a 3 sample maximum hardware delay down the line. Now audio samples are bundled (typically in 20-30ms chunks) for packetising, and to go through low bit rate codecs. Then the bundles get queued in IP routers, creating further latency. VoIP uses massive amounts of complexity in a struggle to offset the effects of these delays, and approach the quality older telephony achieved with simple equipment. In the end the simple equipment will always beat it. That said, in another 10 years there will probably be no traditional PSTN. Strange, huh? Regards, Steve Eric Bishop wrote: OK I understand that the $5 handset may indeed have an echo but that it occurs so fast that it is not preceived as an echo. I pose the following questions: 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? 2. Is only a problem in 2-wire technologies (ie analog and BRI ISDN lines)? 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? ___ 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] Searchable Mailing Lists & NooB Question
On February 10, 2005 11:23 pm, Daniel Wright wrote: > >Very rough numbers: iax-gsm consumes about 22kb/s, > I see about 60kb/s Odd. I have realworld numbers of gsm calls doing 3.4kB/sec which is about 27kbps -- that includes the UDP overhead and possibly also the ethernet frame overhead (I don't recall whether the 'rate' program strips that or not) > > g711 about 80kb/s on > I see 155kb/s no it really is about 80kbps. :-) > 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. Since your numbers are about double what everyone else is stating I am suspecting that you are counting the traffic in both directions. gsm is about 22kbps in *and* out, and you're getting about 60 which is about double my 27kbps number. and 155kbps is pretty damn close to 2*80kbps. -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
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 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] asterisk as sip client behind nat
Hi, I am pretty new to all of this but was able to set up an asterisk server and have been able to succesfully connect to asterisk with x-lite as sip client. I have also connected asterisk to FWD (using iax2) and to voipjet (also using iax2). Now I am trying to connect asterisk to Stanaphone. It has to register as a SIP client but I am not being succesful at all. My asterisk server sits behind a Linksys WRT54GS router (latest linksys firmware) acting as DHCP and NAT for my home equipment. I have tried forwarding the SIP and RTP ports to the asterisk machine, and I have also tried putting the asterisk machine on the routers DMZ. No luck. I have tried every configuration recommended for stanaphone that I found on the web, including a couple I found on the asterisk wiki and a few I found on the stanaphone forums. I tried, explicitly defining the private and public ips, qualifying, using the same number as assigned in stanaphone for my local extension (a recommendation I found), etc. Then I found a message in a forum about a person with the same problem that claims it got fixed when asterisk was put with a public IP address. So my question is is it at all possible to connect asterisk as a SIP client when it sits behind a NAT? If yes, can somebody tell me what I should do please. thank you, -guillermo PS1.- When I connected the x-lite to asterisk both where on the same side of the NAT PS2.- The error I continuosuly get is "SIP/2.0 401 Unauthorized". PS3.- I connected x-lite directly to stanaphone with no problems even behind the nat...and I didn't have to set any port forwarding or anything...so I am thinking that whatever x-lite is doing asterisk should do...how do I emulate what its doing? ___ 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] asterisk as sip client behind nat
Hi, I am pretty new to all of this but was able to set up an asterisk server and have been able to succesfully connect to asterisk with x-lite as sip client. I have also connected asterisk to FWD (using iax2) and to voipjet (also using iax2). Now I am trying to connect asterisk to Stanaphone. It has to register as a SIP client but I am not being succesful at all. My asterisk server sits behind a Linksys WRT54GS router (latest linksys firmware) acting as DHCP and NAT for my home equipment. I have tried forwarding the SIP and RTP ports to the asterisk machine, and I have also tried putting the asterisk machine on the routers DMZ. No luck. I have tried every configuration recommended for stanaphone that I found on the web, including a couple I found on the asterisk wiki and a few I found on the stanaphone forums. I tried, explicitly defining the private and public ips, qualifying, using the same number as assigned in stanaphone for my local extension (a recommendation I found), etc. Then I found a message in a forum about a person with the same problem that claims it got fixed when asterisk was put with a public IP address. So my question is is it at all possible to connect asterisk as a SIP client when it sits behind a NAT? If yes, can somebody tell me what I should do please. thank you, -guillermo PS1.- When I connected the x-lite to asterisk both where on the same side of the NAT PS2.- The error I continuosuly get is "SIP/2.0 401 Unauthorized". PS3.- I connected x-lite directly to stanaphone with no problems even behind the nat...and I didn't have to set any port forwarding or anything...so I am thinking that whatever x-lite is doing asterisk should do...how do I emulate what its doing? ___ 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] Why echo occurs
OK I understand that the $5 handset may indeed have an echo but that it occurs so fast that it is not preceived as an echo. I pose the following questions: 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? 2. Is only a problem in 2-wire technologies (ie analog and BRI ISDN lines)? 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? On Thu, 10 Feb 2005 21:10:16 -0500, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Can someone give me a simple rational explanation why a $5 analog > > handset gives me no echo whatsoever on an analog PSTN line, but > > PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require > > software-based echo cancellation. Surely a $5 analog handset does not > > have an "echo canceller". > > > > The echo I mean is when I hear myself while talking to another party. > > When you talk on the PSTN with an analog phone, in fact you have echo, > but it's coming back so fast, that you think that you just ear > yourself while you are talking. > > No mix in the fact that you are talking on a VoIP phone, that takes > the voice, encode it in the proper codec, send it on the network to > your * box, * decode it, plays it on the PSTN line, takes what it > ears, encode it back in VoIP, send it on the network to your phone > that decodes it and play it back to you. Now, this adds a little > delay, that'S why you ear yourself talking just after you actually > said it. > > This delays make it so that you ear it in echo. While when you are > directly on the PSTN, the echo comes back so fast that you ear it > "almost" at the same time that you say it. When you are going only > VoIP to VoIP, you don't have echo at all because there's no analog > link (that's where the echo is) > > I hope I explained it well enough. > > Please correct me if I'm wrong > > > 1. It is not in the Asterisk box because IP to IP calls do not suffer > > this malady > Exactly > > > 2. It is not from the Central Office to my premesis because my $5 > > analogue handset works without echo. Also PRI ISDN works without echo. > Listen more closely, you'll see that there is echo. > > With echo cancellation, you ear the echo only for the first seconds of > the call. Then the echo cancel is trained enough to suppress it > > hope this help > ___ 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] Searchable Mailing Lists & NooB Question
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
Re: [Asterisk-Users] reboot polycom 1.4.1
> > Quota from the polycom admin guide document, > > So it could be a potential DOS attack problem if set to 1. > > > > Correct me if I am wrong - but the potential for a DoS attack would only > be if an untrusted user is able to gain access to the asterisk server > and the CLI to run commands? Or somehow hijack the SIP session coming > from the server? Just trying to understand the security implications of > allowing the phones to be rebooted remotely (which is a big plus imho). What he meant by that is if polycom is using a registered IP address and is accessible from the Internet directly, then it only a matter of time before some evil person will find it and do strange things to it. If the phone is on a firewall/nat'ed internal network, then its not a problem. ___ 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] Asterisk Versioning
--On Wednesday, February 09, 2005 5:50 PM -0500 Leif Madsen <[EMAIL PROTECTED]> wrote: On Wed, 9 Feb 2005 20:54:51 +0200, Walid Azab <[EMAIL PROTECTED]> wrote: Just want to understand the difference between Asterisk Versions and please correct me if I am wrong, I understand they are: Stable CVS CVS Head As I've noticed several posts regarding Asterisk versioning recently, I thought I'd reply with how I understand Asterisk versioning to work. A few questions on this: I checked out two versions: v1-0 and v1-0-5 My understanding is that v1-0 is actually newer, containing v1-0-5 plus all changes that have happened since then. Is this correct. one more question: When did v1-0-5 come out? This would help me to start reading the asterisk-cvs list to see what changed after it came out. 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] Searchable Mailing Lists & NooB Question
> > with 32Kbps upstream you can run into troubles while trying to transmit > > voice to vonage or other operator due to bandwidth. > > > > > > Just to make sure we are not confusing bits and Bytes here... I have > 32 kilo bytes per second... please oh please just be a "b" and "B" > misunderstanding. > > I was actually worried about upstream bandwidth. This is really going > to make swallow some pride if I have to go out and buy Comcast cable. > Looks like they are offering 48 kBps (384 kbps). Critch missed the "B". Your 256k bandwidth should be just fine for one or two simultanous calls, or more it you use a reasonable provider that supports low bandwidth codecs. Very rough numbers: iax-gsm consumes about 22kb/s, g711 about 80kb/s on the wire. Just don't try to run a commercial web server, etc, over the 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] CISCO CP-7902G and chan_skinny.
Hi everybody, I try use my CISCO 7902 with asterisk. When i OnHOOK and OffHOOK i have a message: RECEIVED UNKNOWN MESSAGE TYPE: 26 And i can't do new call and answer call. -- Best regards, Andrew Kochetkoff mailto:[EMAIL PROTECTED] ___ 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] Wireless LANs and Asterisk
I am doing this with an ancient WAP-11 in Access Point Client Mode. I have it connected to a Sipura 2000 via a crossover cable. It's been very reliable and clean to talk on. I have a WET-11 on order. If it doesn't do a quality job, back it goes. --On Thursday, February 10, 2005 4:17 PM -0800 Scott Laird <[EMAIL PROTECTED]> wrote: On Feb 10, 2005, at 2:39 PM, jurgen wrote: This is interesting - it's something that I've been considering doing for the Asterisk rollout at my company. We don't have enough Ethernet ports and I'm not thrilled about the expense of re-wiring the place. Have you tried D-Link's dual-channel gear for even more bandwidth, or do you feel that bandwidth is not really a problem? How resilient is 802.11g against interference from other sources? Microwave ovens, gigarange phones, etc. Thanks for reporting your success here to the list, just proves I'm not alone with my funny ideas. For what it's worth, I tried using a 7940G with a Linksys WET-11 at home for a few months, and it just wasn't reliable enough to work. The WET11 was only ~40 feet from the base station, but it'd still drop off the net from time to time, and there were tons of random packets dropped. I ended up spending a weekend pulling Cat 5, and life's been a lot better since then. My laptop has similar problems with wireless in the same location, but it's a lot easier to put up with a few lost packets when reading mail or browsing the web--lost VoIP packets stand out like a sore thumb. Scott ___ 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] dtmfmode and IAX protocol
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 just cut the call. -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Thursday, February 10, 2005 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] dtmfmode and IAX protocol Joseph wrote: >>IAX and IAX2 do not support a DTMF mode option. They use out of band >>DTMF *ALWAYS*. >> > > So what you are saying I can not press (# 1, 2 etc) when I dial > somewhere and ask to press a number? > Is there a solution for it? No. I'm saying that DTMF digits are always sent out of band when using IAX/IAX2(very similar idea to RFC2833 for SIP). I don't know why your DTMF is not working, but it's not a problem with inband DTMF, of that I am sure. ___ 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] reboot polycom 1.4.1
Hi Richard, Richard wrote: Quota from the polycom admin guide document, So it could be a potential DOS attack problem if set to 1. Correct me if I am wrong - but the potential for a DoS attack would only be if an untrusted user is able to gain access to the asterisk server and the CLI to run commands? Or somehow hijack the SIP session coming from the server? Just trying to understand the security implications of allowing the phones to be rebooted remotely (which is a big plus imho). tia Matt -- Matt Gibson VOIP Administrator NJ Tech Solutions PSTN: 1.877.999.4678 ex. 6400 FWD: 472645 IAXTEL: 1.700.761.1828 ___ 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
FW: [Asterisk-Users] really easy FOP asterisk@home question
That's what I thought it used to be but it isn't working now Here is what I have in my op_buttons.conf [801]; Meetme must be defined by its room number Position=15 Label="MeetMe 801" Extension=801 Context=rooms Icon=5 [802] Position=16 Label="Meetme 802" Extension=802 Context=rooms Icon=5 This is in the meetme.conf [rooms] #include meetme_additional.conf This is in the meetme_additional.conf conf => 801 conf => 802 In the context instead of rooms I have tired Conferences From-internal Meetme I know it's worked in the past, I'd hate to have to install [EMAIL PROTECTED] from scratch just to fix this FOP (conference rooms themselves work fine). Any suggestions? Thanks, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolás Gudiño Sent: Thursday, February 10, 2005 8:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] really easy FOP [EMAIL PROTECTED] question Hello, > > I deleted the config examples in the op_buttons.conf folder for how to set > up the meetme representation [skip] > > > > [Meetme/801]; Meetme must be defined by its room number change the above line to: [801] Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] dtmfmode and IAX protocol
Joseph wrote: IAX and IAX2 do not support a DTMF mode option. They use out of band DTMF *ALWAYS*. So what you are saying I can not press (# 1, 2 etc) when I dial somewhere and ask to press a number? Is there a solution for it? No. I'm saying that DTMF digits are always sent out of band when using IAX/IAX2(very similar idea to RFC2833 for SIP). I don't know why your DTMF is not working, but it's not a problem with inband DTMF, of that I am sure. ___ 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] Searchable Mailing Lists & NooB Question
On Thu, 2005-02-10 at 17:30 -0800, Geoff Scott wrote: > On Thu, 10 Feb 2005 19:11:38 -0600, Steven Critchfield > <[EMAIL PROTECTED]> 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. > > > I'm new to *. Hence the question. If it's a bother, no need to hit > the reply button. Being new doesn't excuse lack of effort. > >From your answer then, am I to assume no one is running * servers on a > standard DSL line? Yeah, "standard" DSL, what a joke. With all the speeds listed around as being DSL, and the throw away gmail account, we couldn't determine if you where speaking about a deployment in the US or better developed countries where we needed to pay attention to the capitalization of your 'b' or if you where in a less developed area where DSL rollouts actually where of similar speed to dialup. BTW, bandwidth is pretty much always referred to in bits as it is a nice high number and make marketing people very happy. Add to that the fact that the encoding determines if it takes 8 or 10 bits to actually transmit a byte down the line. On my Comcast account back when we did VoIP in my home, we could squeeze 2 concurrent gsm encoded calls down the 128k upstream. At the moment the 3rd concurrent call came up, lag would start setting in and eventually all calls where useless. Of course you also couldn't browse the web very much if more than one call was in progress either. Once you get some time to start looking around you will find there has been several people write up nice little graphs showing what protocols using what codecs use what bandwidth. Then you can do the math on how much you think you want to get accomplished. You will probably find out that you still can't do much on the link outbound and have a call in place at the same time. So you probably want to still turn off the webserver and jabber server, they would be better off coloed anyways and there are a lot of cheap colo places for non critical hosting. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] TelIAX troubles
> We are having issues setting up our Asterisks server with Teliax > service. We are able to place calls, but cannot seem to get our > Asterisk box to answer from Teliax service. > > We are using Asterisk with the latest AMP interface. > > Teliaxâs examples are for single SIP phone, not the voice response > systems, nor do they provide any support of Asterisk other than > basic sample scriptsâ Teliax's examples are fine. You will need to learn how to administer Asterisk, however, to make it work with _any_ provider. -Ron ___ 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] Voice Recognition
Has anyone out there had any success with any type of voice recognition? I would LOVE to see something work within Asterisk. I have tried for days and can't seem to get this right. Is it even possible? I have read everything I can find, but it's like the sphinx integration/documentation was dropped because of lack of interest. I tried this with Sphinx-2 because it contained the programs I read about. Sphinx-4 looks like a total rework in Java and was pretty confusing to me. Please share your experiences!! Thanks! Mark Johnson ___ 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] Why echo occurs
Eric, youmight want to search the archives. I wrote a rather lengthy explaination on echo and echo control last year. Eric Bishop wrote: Hi all, Can someone give me a simple rational explanation why a $5 analog handset gives me no echo whatsoever on an analog PSTN line, but PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require software-based echo cancellation. Surely a $5 analog handset does not have an "echo canceller". The echo I mean is when I hear myself while talking to another party. I have heard it said it is because of some slowdown of the signal. But where is this mysterious bottleneck? 1. It is not in the Asterisk box because IP to IP calls do not suffer this malady 2. It is not from the Central Office to my premesis because my $5 analogue handset works without echo. Also PRI ISDN works without echo. Can anyone explain what I am missing? ___ 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] Why echo occurs
On February 10, 2005 08:57 pm, Eric Bishop wrote: > Can someone give me a simple rational explanation why a $5 analog > handset gives me no echo whatsoever on an analog PSTN line, but > PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require > software-based echo cancellation. Surely a $5 analog handset does not > have an "echo canceller". The $5 handset isn't introducing signficant delay into the audio stream. The $5 handset has just as much echo as a TDM400/Sipura or T100P, you just don't hear it as echo because you're hearing it "at the right time" and all you percieve it as is a sidetone. > I have heard it said it is because of some slowdown of the signal. But > where is this mysterious bottleneck? It's in the digitization of the voice frames. It's additionally in the transformation between codecs. It's additionally in the time it takes to perform these steps and move the intermediary data around in memory multiple times. It's additionally in the time it takes to get the data out the network card and across the internet and finally, it's additonally in all these reverse steps to get the data back out to a POTS interface like your friend's TDM400P. :-) > 1. It is not in the Asterisk box because IP to IP calls do not suffer > this malady IP to IP calls don't have a hybrid circuit to introduce reflected voice energy, which is the basic source of the echo problem. The $5 handset has the same problem, but without the delay, all you hear is a comfortable "sidetone" (your own voice in the earpiece). > 2. It is not from the Central Office to my premesis because my $5 > analogue handset works without echo. Also PRI ISDN works without echo. Actually no it doesn't. I have significant echo problems on my ISDN PRI, as do many others. Those without ISDN PRI echo problems have good echo cancellation hardware sitting on the physical T1. This isn't an asterisk-specific problem. Every single piece of VOIP (or cellular, for that matter) hardware that interfaces to the PSTN has to deal with this in one form or another. Some methods are just better than others. Googling will reveal a LOT of research into this problem. It's by no means trivial. -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
Re: [Asterisk-Users] Why echo occurs
On Feb 10, 2005, at 5:57 PM, Eric Bishop wrote: Hi all, Can someone give me a simple rational explanation why a $5 analog handset gives me no echo whatsoever on an analog PSTN line, but PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require software-based echo cancellation. Surely a $5 analog handset does not have an "echo canceller". Conventional wisdom says that it takes two things to get audible echo: 1. A 2-wire leg to the connection, such as a POTS line. 2. A substantial delay. If you don't have a 2-wire leg, then you won't generally get an echo. If you have a minor echo, but the delay is short, then you don't perceive it as an echo--your voice in the earpiece just sounds a little bit louder then normal. If you have both an echo and a delay, though, then you get a noticable echo, because the sound of your voice echos off the far end of the call and comes back late enough for your ears to hear it. So, you don't hear an echo with a POTS line because it's too fast to hear. You don't hear it with a pure VoIP call because the two ends of the call never get mixed together. Scott ___ 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] No dialtone in a E1
On Thu, 10 Feb 2005, Marco Castillo wrote: > Hi, I'm having a little problem when trying to make a call from asterisk. I > connect a SIP phone to asterisk, and in the asterisk box I have a TE110P > card connected to a E1. When a SIP client makes a call through the E1, I > received no dialtone in the SIP client. > In the same manner, when somebody from the POTS network makes a call to a > SIP client (through * and the E1) he doesn't receive the apropiate tone of > call progress. Does anyone has some ideas about this? Are you talking about an ISDN E1 or another form of E1? On isdn dialtone is an optional feature of the specification and there are many implementations of isdn. I think it is mandatory on EuroISDN. Since asterisk normally generates the dialtone itself there should be little nead for the dialtone from the pstn. We use the dialtone from the network ourselves, but asterisk could provide it as well. In band call progress is also a feature of the net on isdn. If the net does not provide it you will have to do so yourself. Just add the proper options to Dial to generate ringback and if the call fails you generate the matching sound (Busy etc). Peter ___ 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] Why echo occurs
> Can someone give me a simple rational explanation why a $5 analog > handset gives me no echo whatsoever on an analog PSTN line, but > PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require > software-based echo cancellation. Surely a $5 analog handset does not > have an "echo canceller". > > The echo I mean is when I hear myself while talking to another party. When you talk on the PSTN with an analog phone, in fact you have echo, but it's coming back so fast, that you think that you just ear yourself while you are talking. No mix in the fact that you are talking on a VoIP phone, that takes the voice, encode it in the proper codec, send it on the network to your * box, * decode it, plays it on the PSTN line, takes what it ears, encode it back in VoIP, send it on the network to your phone that decodes it and play it back to you. Now, this adds a little delay, that'S why you ear yourself talking just after you actually said it. This delays make it so that you ear it in echo. While when you are directly on the PSTN, the echo comes back so fast that you ear it "almost" at the same time that you say it. When you are going only VoIP to VoIP, you don't have echo at all because there's no analog link (that's where the echo is) I hope I explained it well enough. Please correct me if I'm wrong > 1. It is not in the Asterisk box because IP to IP calls do not suffer > this malady Exactly > 2. It is not from the Central Office to my premesis because my $5 > analogue handset works without echo. Also PRI ISDN works without echo. Listen more closely, you'll see that there is echo. With echo cancellation, you ear the echo only for the first seconds of the call. Then the echo cancel is trained enough to suppress it hope this help ___ 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] Why echo occurs
Hi all, Can someone give me a simple rational explanation why a $5 analog handset gives me no echo whatsoever on an analog PSTN line, but PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require software-based echo cancellation. Surely a $5 analog handset does not have an "echo canceller". The echo I mean is when I hear myself while talking to another party. I have heard it said it is because of some slowdown of the signal. But where is this mysterious bottleneck? 1. It is not in the Asterisk box because IP to IP calls do not suffer this malady 2. It is not from the Central Office to my premesis because my $5 analogue handset works without echo. Also PRI ISDN works without echo. Can anyone explain what I am missing? ___ 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] Searchable Mailing Lists & NooB Question
Geoff, I believe that there are * users running with DSL connections but with 128Kbps or more of upstream. Personally, I've used to have Asterisk behind a 128Kbps upstream connection and always had a problem with ulaw codec, i could hear clearly but couldn't be hear without large cuts on audio (lack of bandwidth) them moved to G729 and started to speak with pretty good quality but only to one max two calls because I had data passing thru the router also. With 32K I don't believe you could do much even because I don't know any VoIP operator for termination on landlines or mobiles that support GSM... - Original Message - From: "Geoff Scott" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, February 11, 2005 1:30 AM Subject: Re: [Asterisk-Users] Searchable Mailing Lists & NooB Question > On Thu, 10 Feb 2005 19:11:38 -0600, Steven Critchfield > <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> > > > > ___ > > 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 > > -- > I have some G-Mail invites. > Let me know if you want one. > ___ > 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] Searchable Mailing Lists & NooB Question
On Thu, 10 Feb 2005 17:38:09 -0800, Max Klein <[EMAIL PROTECTED]> wrote: > 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 > Okay, maybe all of this is my fault. I wrote "32kBps" assuming the capital B indicated Bytes. Being that there are 8 bits in a bite, 32 kBps is equal to 256kbps, using the lower case b to indicate bits. I shall hence forth never refer to bandwidth in terms of Bytes, but rather in bits. Geoff -- I have some G-Mail invites. Let me know if you want one. ___ 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] 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 > <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> > > > > ___ > > 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
Re: [Asterisk-Users] Searchable Mailing Lists & NooB Question
On Thu, 10 Feb 2005 19:11:38 -0600, Steven Critchfield <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> > > ___ > 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 -- I have some G-Mail invites. Let me know if you want one. ___ 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] Searchable Mailing Lists & NooB Question
Geoff Scott wrote: If that's the case, goodbye Verizon (Internet and phone) and hello Comcast (Internet) and Vonage. Ack. Vonage? There are lots better ways of doing VoIP than Vonage. B. ___ 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] really easy FOP asterisk@home question
Hello, > > I deleted the config examples in the op_buttons.conf folder for how to set > up the meetme representation [skip] > > > > [Meetme/801]; Meetme must be defined by its room number change the above line to: [801] Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] Searchable Mailing Lists & NooB Question
On Fri, 11 Feb 2005 00:03:09 -, Helder Rogério [MICROREDE] <[EMAIL PROTECTED]> wrote: > Hi Geoff, > > with 32Kbps upstream you can run into troubles while trying to transmit > voice to vonage or other operator due to bandwidth. > > Just to make sure we are not confusing bits and Bytes here... I have 32 kilo bytes per second... please oh please just be a "b" and "B" misunderstanding. I was actually worried about upstream bandwidth. This is really going to make swallow some pride if I have to go out and buy Comcast cable. Looks like they are offering 48 kBps (384 kbps). If that's the case, goodbye Verizon (Internet and phone) and hello Comcast (Internet) and Vonage. Geoff -- I have some G-Mail invites. Let me know if you want one. ___ 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] Fail to detect DTMF over direct ISDN pri link
Title: Fail to detect DTMF over direct ISDN pri link Hello, I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test, I want to used a direct ISDN link with the platform, without to change anything at the Asterix side, I configure the platform to be the ISDN Network side (DMS100) with a twist cable. The D-Channel can up; I am able to perform call, except Asterisk doesn't detect any DTMF anymore? Why? What is the relation with the ISDN link? I use the "monitor" command to record the call, and I really hear the DTMF tone correctly... I try to put relaxdtmf=yes in the Zapata.conf, but no success Thanks for any help or suggestion to diagnose this problem. Sylvain Gagnon, B.Ing., M.Sc.A. Speech Technology Integrator Intégrateur en technologie de la voix BCE Elix Specialist in contact center solutions Spécialiste en solutions pour centres de contacts 14 Commerce Place, 5th Floor Nuns' Island (Québec) CANADA H3E 1T5 t: 514-768-1000, ext. 2224 f: 514-768-7680 www.bceelix.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
[Asterisk-Users] TelIAX troubles
We are having issues setting up our Asterisks server with Teliax service. We are able to place calls, but cannot seem to get our Asterisk box to answer from Teliax service. We are using Asterisk with the latest AMP interface. Teliax’s examples are for single SIP phone, not the voice response systems, nor do they provide any support of Asterisk other than basic sample scripts… ___ 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] Proper Contexts in extensions.conf
What about... [incoming] Include => internal [sip-extensions] Include => internal Include => long-distance [internal] ... internal extensions ... [long-distance] ... > -Original Message- > From: Max Clark [mailto:[EMAIL PROTECTED] > Sent: Thursday, February 10, 2005 3:31 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Proper Contexts in extensions.conf > > Hi all, > > I am looking for examples of the extensions.conf that puts > all incoming calls into a context where extensions can be > dials, and all phones in a context where extensions and > outside calls can be dialed. > > i.e. I have seen: > > [incoming] > include => sip-extensions > > [sip-extensions] > include => longdistance > > [longdistance] > > > Doesn't this allow any internal callers to make external > calls? How do you properly set this up? > > Thanks, > Max > > -- >Max Clark >max [at] clarksys.com >http://www.clarksys.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 > ___ 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] Searchable Mailing Lists & NooB Question
On Thu, 2005-02-10 at 16:00 -0800, Geoff Scott wrote: > Quick question. > > New to the project, and generally I like to sniff through the mailing > list archives. Are there no searchable archives? I've only found the > monthly archive files for download. Normally I would have searched > the for the answer to the following question: google google google, learn to use google. Learn to look at the advanced search options. Notice that you can set the results to only come from a site. Also, maybe you should notice that the last link of every page sends you to the appropriate mailman page for the mailing list. On that page is a link to the archives. On that page, you can select to view any month by thread, subject, Author, or date. Pretty simple. > I'm getting ready to set up a * server at home. No digium equipment. > Using FWD and maybe Vonage in the not too distant future. After that, > I will consider adding one FXO and FXS port (maybe the digium dev > kit). I'm planning this on a Celleron 400mhz. Should I even attempt > this? The box will also be running a web server and a chat server > (Jabber). > > Any thing I should watch out for? Any surprises I could run into? I > have a DSL line that is 32 kBs up. 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 <[EMAIL PROTECTED]> ___ 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] Wireless LANs and Asterisk
On Fri, 2005-02-11 at 09:39 +1100, jurgen wrote: > Hi Mike, > > This is interesting - it's something that I've been considering doing > for the Asterisk rollout at my company. We don't have enough Ethernet > ports and I'm not thrilled about the expense of re-wiring the place. > > Have you tried D-Link's dual-channel gear for even more bandwidth, or > do you feel that bandwidth is not really a problem? How resilient is > 802.11g against interference from other sources? Microwave ovens, > gigarange phones, etc. > > Thanks for reporting your success here to the list, just proves I'm > not alone with my funny ideas. You also need to think about cost. Most cheap wireless bridges are $50 or more. I bet you could wire 2 phones for the cost of each bridge. Then you don't have the trouble of what happens if you get interference. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] Wireless LANs and Asterisk
We are deploying an * soultion at the WISP that I freelance for. We are using a tranport that I designed called MATE: Multplexed Audio Transmited over Ethernet MATE is designed to be a better TDMoE. It uses uLAW and huffman compression. We also use custom Customer Premis Equipment that garenties the dilivery of the MATE streams. So far MATE supports 64 channels per stream. Streams are MAC -> MAC. Each MATE client note can support 16 streams. Each MATE server node can suport unlimmited streams. The system is still in prototype stage, but will be in full use within about three months. -- Christopher Dobbs ___ 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] Wireless LANs and Asterisk
On Feb 10, 2005, at 2:39 PM, jurgen wrote: This is interesting - it's something that I've been considering doing for the Asterisk rollout at my company. We don't have enough Ethernet ports and I'm not thrilled about the expense of re-wiring the place. Have you tried D-Link's dual-channel gear for even more bandwidth, or do you feel that bandwidth is not really a problem? How resilient is 802.11g against interference from other sources? Microwave ovens, gigarange phones, etc. Thanks for reporting your success here to the list, just proves I'm not alone with my funny ideas. For what it's worth, I tried using a 7940G with a Linksys WET-11 at home for a few months, and it just wasn't reliable enough to work. The WET11 was only ~40 feet from the base station, but it'd still drop off the net from time to time, and there were tons of random packets dropped. I ended up spending a weekend pulling Cat 5, and life's been a lot better since then. My laptop has similar problems with wireless in the same location, but it's a lot easier to put up with a few lost packets when reading mail or browsing the web--lost VoIP packets stand out like a sore thumb. Scott ___ 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] Searchable Mailing Lists & NooB Question
Hi Geoff, with 32Kbps upstream you can run into troubles while trying to transmit voice to vonage or other operator due to bandwidth. - Original Message - From: "Geoff Scott" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, February 11, 2005 12:00 AM Subject: [Asterisk-Users] Searchable Mailing Lists & NooB Question > Quick question. > > New to the project, and generally I like to sniff through the mailing > list archives. Are there no searchable archives? I've only found the > monthly archive files for download. Normally I would have searched > the for the answer to the following question: > > I'm getting ready to set up a * server at home. No digium equipment. > Using FWD and maybe Vonage in the not too distant future. After that, > I will consider adding one FXO and FXS port (maybe the digium dev > kit). I'm planning this on a Celleron 400mhz. Should I even attempt > this? The box will also be running a web server and a chat server > (Jabber). > > Any thing I should watch out for? Any surprises I could run into? I > have a DSL line that is 32 kBs up. > > Sorry if this has been covered ad-naseum before on the list. > > Geoff > > -- > I have some G-Mail invites. > Let me know if you want one. > ___ > 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] Round Robin Strategy doesn't seem to work
Let me consolidate a little to address the multiple responses that I've gotten to my question - Thanks to all who've replied so far. Essentially, I am setting up a small company's auto attendant, and they have people who cover each other's job functions as needed. So, there is a 'Sales' guy who should be first in line for each and every sales call. But, if he's unavailable (say on another call) then the sales call should go on to the Dedicated Support guy (and on and on). So, I have another queue, dedi-support, where the Dedicated Support agent should be first in line for every call, with the sales person's agent on down the list. I know that this is a fairly strange scenario, but if I set this all up as queues and agents now (versus contexts in extensions.conf, which I could easily configure to do what I'm looking for), it becomes much easier for this company to maintain their auto attendant as they grow. What I'm looking for is exactly a linear hunt group (so that they can collect queue statistics and change to a different strategy when the have more employees). I've been going off the descriptions of strategies at: http://www.voip-info.org/wiki-Asterisk+call+queues I think I now understand the slight difference between roundrobin and rrmemory - rrmemory will place the next call first to the queue member after the one that answered the current call, while roundrobin will place the next call first to the member after the one that got the first crack at the current call. So is there a 'linear' method available? I've tried using penalties, but then I can't get the calls to ring past the first agent (the one with no penalty) if they are logged into the queue but don't answer. For example, when I have: [sales] music = default strategy = roundrobin timeout = 20 retry = 5 reportholdtime = yes announce = queue-sales member => Agent/9021 member => Agent/5901,1 member => Agent/9020,2 member => Agent/1114,3 And agent 9021 doesn't answer within the timeout, it will retry starting with agent 9021 again - we have the potential to get stuck in a loop here. I could use the autologout feature, but then a caller who reaches voicemail (the next step after Queue(sales|t|||120) in my auto attendant context) has effectively logged all agents out of the queue. So I'm a little stuck on how to handle this. Any suggestions or further insight (I'm probably overlooking a simple little solution). Thanks, Todd ___ 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] Round Robin Strategy doesn't seem to work
On Thu, 2005-02-10 at 13:31 -0800, Todd Gunsolley wrote: > Hi, > > I have configured a call queue as follows: > [sales] > music = default > strategy = roundrobin > timeout = 20 > retry = 5 > reportholdtime = yes > announce = queue-sales > member => Agent/9021 > member => Agent/5901 > member => Agent/9020 > member => Agent/1114 > > Now, I would expect all calls to this queue to be delivered first to 9021, > then to 5901, then to 9020, then to 1114 (all agents are logged in and > available). However, what actually happens is that the first call to the > queue is delivered first to 9021 (and on through the queue order from > there), but then the next call is delivered first to 5901 (and on through > the queue order from there). The third call to the queue is delivered first > to 9020, the fourth call to 1114. > > What I want is for all calls to the queue to be delivered to available > agents in the order that they are listed in the member configuration each > and every time. Am I missing the point here? Has anyone else run into this > problem? IMHO, this sounds like a bug. The events you are describing are acting as though you had used "strategy = rrmemory" which is round robin + memory. Either your config file is wrong, or you haven't reload'ed the config file, or there be a bug. Please check the output of "show queues" and then post a bug to mantis if required. Also you haven't told us the version of asterisk that you are using this on, so make sure you include that in your bug report. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] SIP jitter?
To be totally honest: I wrote the thing. I don't think it's ready to go into HEAD, until the core people can at least agree on the overall structure of the implementation and integration.. There's at least one major fork that one could take with it's architecture (basically, whether it should be applied separately to each VoIP technology, or in common channel handling code), and there are pros and cons to be weighed there. If we choose to continue down with this fork (presenly, it's set up to apply to each technology separately, with a sample integration point for IAX2), we could make things configurable, so you can switch between the old and new jitterbuffers (perhaps with some limitations, like only at start-time, or for new calls, etc), get it applied, and work on perfecting it. It _is_ being used in iaxclient right now, so most of the up-to-date IAX softphones have it, BTW. But, in that case, the integration is much simpler, and I have more control over the project.. -SteveK ___ 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] Searchable Mailing Lists & NooB Question
Quick question. New to the project, and generally I like to sniff through the mailing list archives. Are there no searchable archives? I've only found the monthly archive files for download. Normally I would have searched the for the answer to the following question: I'm getting ready to set up a * server at home. No digium equipment. Using FWD and maybe Vonage in the not too distant future. After that, I will consider adding one FXO and FXS port (maybe the digium dev kit). I'm planning this on a Celleron 400mhz. Should I even attempt this? The box will also be running a web server and a chat server (Jabber). Any thing I should watch out for? Any surprises I could run into? I have a DSL line that is 32 kBs up. Sorry if this has been covered ad-naseum before on the list. Geoff -- I have some G-Mail invites. Let me know if you want one. ___ 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] Wireless LANs and Asterisk
The ulaw codec is the heaviest codec there is. Have you tried lighter codecs such as the gsm codec? Erik On Thu, 10 Feb 2005 22:01:01 +0100, Michiel van Baak <[EMAIL PROTECTED]> wrote: > Hi, > > My experience: > A handfull of concurrent calls, all works fine on 54mbit. > Dont try to go beyond that. Specially when your link is not > totally 100%. We tried to do 10 calls on a dedicated > Conceptronic AP and all fell down. even with the ulaw codec > it was not doable for normal conversations. Even disabling > web was not the answer, so we took the good old wires again. > > And another concern: privacy. > As you know, WEP is not that strong. And as long as there is > no solid encrypted RTP stream everyone with a laptop is able > to monitor/record your calls. > > just my 2 cents > -- > Michiel van Baak > http://lunteren.vanbaak.info > [EMAIL PROTECTED] > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D > > "Two of the most famous products of Berkeley are LSD and BSD. I don't think > that this is a coincidence." > > ___ > 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] Proper Contexts in extensions.conf
Hi all, I am looking for examples of the extensions.conf that puts all incoming calls into a context where extensions can be dials, and all phones in a context where extensions and outside calls can be dialed. i.e. I have seen: [incoming] include => sip-extensions [sip-extensions] include => longdistance [longdistance] Doesn't this allow any internal callers to make external calls? How do you properly set this up? Thanks, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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
Re: [Asterisk-Users] Asterisk - SER Configuration
In my opinion this would be overkill. Just use Asterisk to forward calls to other Asterisk boxes. $0.02 Alberto Zuin wrote: Hello all! I'm new in this ML and I write you for a suggestion about integrate Asterisk and SER. My idea is to use Asterisk as a local PBX server where users can authenticate and make local calls, but when a user dial a non local number, an asterisk extension call SER Server who redirct to right remote asterisk. Originally I make this only with asterisk where in everyone I setted iax.conf to connect to every remote server. The size of my net in increasing, and then I want to modify it in a "star center" network and I want use ser in center to be sure to avoid rtp traffic. Now, you can point me to a working configuration example for asterisk and ser? Thanks, Alberto Zuin ___ 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] Codec passthrough patch for IAX
Hi there, I had a problem, basically, I have 4 different types of end users (gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider. My provider supports all 4 codecs. The issue is then: When an incoming call comes in, a codec is negotiated (usually ULAW), later on, when the extension is dialed, we'll see we're doing GSM, and thus transcode. Here's an example dialplan: [incoming] exten => 123,1,Dial(IAX2/gsmUser) exten => 456,2,Dial(IAX2/ilbcUser) exten => 789,3,Dial(IAX2/g729User) You're pretty much forced to accept ULAW, and then transcode. Not fun if your provider does it for you (that's what you pay them for, right?). So, with this patch, just add a new config file. codec_passthrough.conf: [iax_my-did-provider] 123=gsm 456=ilbc 789=g729 Now, when an incoming call comes in, the user/extension will be found, and your preferred codec changed. No more transcoding. http://bugs.digium.com/bug_view_page.php?bug_id=0003553 My main question is: Can this be done without this patch? I've heard it's impossible, and it sure seems that way. Any suggestions? -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] No dialtone in a E1
Hi, I'm having a little problem when trying to make a call from asterisk. I connect a SIP phone to asterisk, and in the asterisk box I have a TE110P card connected to a E1. When a SIP client makes a call through the E1, I received no dialtone in the SIP client. In the same manner, when somebody from the POTS network makes a call to a SIP client (through * and the E1) he doesn't receive the apropiate tone of call progress. Does anyone has some ideas about this? Ing. Marco Antonio Castillo Chief Design Engineer Van Der Kaaden IT Consulting Guatemala, Guatemala C.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
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
Hi! > >>Oh right.. I remember seeing that.. yeah that looked a whole lot more > >>elegant than *8. Why isn't it in HEAD? > > > >I'm not sure. Once it started getting some testing BKW closed it. If > >someone is interested in testing the patch I'm sure the bug could be > >reopened. > > > I'll test it.. it'll be a few weeks before I can.. but I'm very interested.. Might be easier to look at the PickUp, PickDown etc applications that come with bristuff. Cheers, Philipp ___ 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: Terrible inbound call quality vs. outbound
To Bryan, Brian, Daryl, David and others who have reported the same VoicePulse inbound problem I am experiencing: In my extreme efforts to debug this inbound call quality situation, it seems that if I disable the IAX2 registry and instead register a SIP channel against VoicePulse (gw5.voicepulse.com) for my inbound context, the call quality issue is completely fixed. Do any of you guys have a moment to try that out on your configuration? It may help us all in our attempts to pinpoint this issue... even if VP is not willing to put any efforts into diagnosing this. Thanks, /rg On Feb 2, 2005, at 10:44 AM, bryan tholen wrote: Just to add some weight here, I am having the exact same issue. My VoicePulse 512 DID is very unstable but out bound calls are fine. Also my Toll-Free DID through NuFone is fine in both directions. I spent a lot of time troubleshooting my end (QOS,Asterisk server capabilities,Hardware timing) none of which resolved the incoming call quality issues. I finally got the NuFone DID and could confirm the problem is not on my end. I have not contacted VoicePulse regarding this issue but I will be doing that soon. Any further input is much appreciated. David McNett wrote: On 01-Feb-2005, Robert Goodyear wrote: Sadly, VP seems to have a fairly high comparative "rating" against other VOIP service while they seem to maintain horrible customer support and crappy line quality. Sigh. I wonder why the TX side of the conversation is clear though? Seems like the packets would be treated identically since it's a full-duplex conversation. I have this exact same problem with one of my voicepulse connect DIDs, but not the other one. This, I think, pretty clearly rules out any local asterisk-side configuration issues. My area code 512 DID is effectively unusable. 20-30% of all inbound calls afflicted. The other party can hear me fine, but I can't make any sense out of what they're saying. My area code 510 DID has been flawless, however. Clearly the problem is specific to the provider that voicepulse is using to supply their 512 DID service. I opened a ticket with voicepulse about three weeks ago, and I've called two or three times to complain about the issue, but I've seen no progress or improvement. ___ 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] WAS: Strategy for a stable IAXy NOW: IAXy vs old P-3
> >And the only IAX2 box made is the Digium one, with it's current > shortcomings ? hmm why not try one of the pa-168 http://voip-info.org/tiki-index.php?page=PA168 based iax2 phones ? http://www.ariavoice.com/items/108 http://ipphone.eezeephone.com/index_files/Page466.htm http://www.iaxtalk.com/ etc ___ 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] sip_notify.conf
Kelly Griffin wrote: I went to the Mantis site yesterday and attempted to do a search on "sip" and "sip_notify" with no luck. I did not, however, uncheck the "closed" box. Too bad :-) However, if you had mentioned that in your message, I would have responded in a far different fashion. Some people are not looking for the easy way out. Some people just need a "helping hand". If reading newbie questions stresses you out, maybe you shouldn't read them. I'm not stressed at all... be thankful it wasn't Steven Critchfield that responded to you :-) We were all newbies at one time. There is a big difference between a newbie that shows that they have actually tried to solve their own problem first vs. one that does not. You may have already spent hours perusing the available sources of information, but if you didn't let us know that you did that, we have to assume that you didn't. ___ 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] how to pop up called number details using phpscripts in agi scripts
Hi, Covide looks interesting. Is this a killer combination of groupware and Asterisk I was looking for ? Is it open source ? Do you have any more english info ? Thanks in advance, regards, Rob. - Original Message - From: "Michiel van Baak" <[EMAIL PROTECTED]> To: Sent: Thursday, February 10, 2005 6:32 PM Subject: Re: [Asterisk-Users] how to pop up called number details using phpscripts in agi scripts > On 03:14, Wed 09 Feb 05, Matt Gibson wrote: > > Michiel van Baak wrote: > > >On 05:14, Tue 08 Feb 05, Mazhar Hussain wrote: > > > > > >If this sounds usefull to you, reply so on the list and I > > >will try to setup a clear txt doc where and how to find the > > >sourcecode. > > > > > i > > I would like to see the information you can provide on this. > > > ok. > I have put a small tarball online with the files that do the > trick for me. I slightly modified them. To get the original > files you can check out the cvs tree of the Covide project: > http://sourceforge.net/projects/covide > The small tarball is at: http://michiel.vanbaak.info/Files/voip.tgz/file_view > The agi script searches in the CRM database for records that > match the phone nr. > The tel1.php makes a XML file from the info in the table > that is filled with the agi script. > tell.php is a php script that uses XMLHttpRequest to reload > this XML file all the time without blocking/reloading your > web application. showtel.php is the popup window. All is > very basic, but it seems to work perfectly here at our > company. > Feel free to burn it down or to make adjustments. > > -- > Michiel van Baak > http://lunteren.vanbaak.info > [EMAIL PROTECTED] > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D > > "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence." > > ___ > 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] Round Robin Strategy doesn't seem to work
- Original Message - From: "Todd Gunsolley" <[EMAIL PROTECTED]> To: Sent: Thursday, February 10, 2005 9:31 PM Subject: [Asterisk-Users] Round Robin Strategy doesn't seem to work Hi, I have configured a call queue as follows: [sales] music = default strategy = roundrobin timeout = 20 retry = 5 reportholdtime = yes announce = queue-sales member => Agent/9021 member => Agent/5901 member => Agent/9020 member => Agent/1114 Now, I would expect all calls to this queue to be delivered first to 9021, then to 5901, then to 9020, then to 1114 (all agents are logged in and available). However, what actually happens is that the first call to the queue is delivered first to 9021 (and on through the queue order from there), but then the next call is delivered first to 5901 (and on through the queue order from there). The third call to the queue is delivered first to 9020, the fourth call to 1114. What I want is for all calls to the queue to be delivered to available agents in the order that they are listed in the member configuration each and every time. Am I missing the point here? Has anyone else run into this problem? Thanks for any insight. Regards, Todd Excuse my Asterisk ignorance but in telephony terms your queue (or hunt group) is doing what would be expected from a round-robin group, to do what you want would be referred to as a linear group, maybe some * expert can confirm. 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
Re: [Asterisk-Users] Round Robin Strategy doesn't seem to work
Todd Gunsolley wrote: What I want is for all calls to the queue to be delivered to available agents in the order that they are listed in the member configuration each and every time. Am I missing the point here? Has anyone else run into this problem? Not sure it is a "problem". More of a feature. You have set up a ACD or UCD, which is what most people want, so that calls get distributed more or less equally across a group . Especially in a sales environment, commissioned sales people would raise the roof on what you want. They want equal shots at incoming calls. Can one person be taken out of the list on demand, in case they go to lunch or have to take a leak? What you really want is a circular hunt group ( of stations ) , but in a circular hunt, if all stations are busy, the caller will still hear ring until one ( station ) is free. A terminal hunt will ring up the list until all are busy, then you either need to return busy or perhaps go to a VM box? Sorry I don't have an answer for you. Are there different types of "queues" or hunt groups that you can define? John Novack ___ 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] Is there a Caller ID issue in the latest CVSStable
> Paul, 1.0.5 stable suffers from caller id issues as well, at least for > SIP channels. What fixed things for me was swapping in app_dial.c from > 1.0.2 stable (didn't try others). You could also just diff app_dial.c > between versions to find the problem but I took the lazy way out the > first time around. Drumkilla reverted the callerid changes on the latest stable (thanks Russell!). You will be fine if you checkout stable from CVS now. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] asterisk GUI's that supports zap fxs extensi ons
by "GUI" do you mean a configuration utility or a User Interface? MATT--- -Original Message- From: Jon Gabrielson [mailto:[EMAIL PROTECTED] Sent: Thursday, February 10, 2005 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] asterisk GUI's that supports zap fxs extensions Are there any gui's that support zap fxs extensions? AMP seems to be one of the more popular gui's but it doesn't support zap fxs devices. Thanks, Jon. ___ 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] Dial SIP peers
On Thu, 10 Feb 2005 16:33:46 -0500, Gene Willingham <[EMAIL PROTECTED]> wrote: >exten => s,1,Dial(SIP/192.168.1.8:,20); Connect to 192.168.1.8 on > port , with a 20 sec timeout. >exten => s,1,Dial(SIP/[EMAIL PROTECTED]:9876,20,r) ; Connect to sip.com > port 9876, requesting extension 8500. > > > > I defined a sip peer called sip-gateway. If I dial by ip address and port > the call goes through on requested port. If I try to dial by peer and port > the call will try to go through on the default port. You should define the non-standard port in a port= statement in the peer entry in sip.conf. Peter ___ 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] Context fails so falling back to extension " s" ?
>Extension 's'? I thought 's' meant Start, not an actual extension. If >there's something I'm not reading or need to read again, don't >hesitate to hit me with a clue stick. Sort of. 's' is used when there is no matching extension in the context. It's the fallback extension if there's no match. http://www.voip-info.org/wiki-Asterisk+s+extension You don't list your extensions.conf, but taking a stab at it, you would put in something like: [from-pstn] exten => s,1,Dial(YourInternalExtension,15) 'Dial whatever your internal extension is for 15 seconds exten => s,2,Hangup() 'Hang up the line if nobody answers. You could put in a goto to fire the call to the [from-internal] context in extensions_additional.conf so it can have voicemail logic. I found that the best part of AMP is they have a really really good extensions.conf you can use as a template to make a customized dialplan. Starting from the base AMP extensions.conf and extensions_additional.conf, I have modified my dialplan *way* beyond what AMP can do, but it's AMP's template that got me started. I shudder to think of the hours I would have wasted creating all of the dialplan logic over again from scratch without AMP giving me a leg-up. Now, I don't even use AMP anymore except for FOP and the call detail logs. YMMV. hth ___ 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] Asterisk and Sipura SPA-841 SIP phones
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 10 February 2005 07:07 am, Giovanni Powell wrote: > Nothing to do with your question, but by any chance, when you plugged > the phone into the wall did you hear a dialtone or is this something > generated by asterisk Nope, no dialtone.can't get that without registering with the Asterisk server for the SIP channels. I did get the registration to work about 3:00 am this morning. Had to change the host= in the sip.conf file to dynamic and added a defaultip= parameter pointing to the phone's IP and it registered immediately without a problem. Does not seem right since I am not using DHCP - the phone IP addresses are all statically assigned - but it is working nevertheless now. Chris -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCC94oG4PxJjbMvv0RAr6pAJ9UtlAMdCGYkD48TN/DFcsRiPy/XACdEECy rR2lrTJegyI8HK0JLLYWhzU= =d67F -END PGP SIGNATURE- ___ 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] Round Robin Strategy doesn't seem to work
You could use penalties to enforce the order like so: member => Agent/9021 member => Agent/5901,1 ; the number after the comma is a penalty member => Agent/9020,2 ; those with the highest penalty should member => Agent/1114,3 ; receive calls last Hope this helps, ~Senyo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Gunsolley Sent: Thursday, February 10, 2005 1:32 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Round Robin Strategy doesn't seem to work Hi, I have configured a call queue as follows: [sales] music = default strategy = roundrobin timeout = 20 retry = 5 reportholdtime = yes announce = queue-sales member => Agent/9021 member => Agent/5901 member => Agent/9020 member => Agent/1114 Now, I would expect all calls to this queue to be delivered first to 9021, then to 5901, then to 9020, then to 1114 (all agents are logged in and available). However, what actually happens is that the first call to the queue is delivered first to 9021 (and on through the queue order from there), but then the next call is delivered first to 5901 (and on through the queue order from there). The third call to the queue is delivered first to 9020, the fourth call to 1114. What I want is for all calls to the queue to be delivered to available agents in the order that they are listed in the member configuration each and every time. Am I missing the point here? Has anyone else run into this problem? Thanks for any insight. Regards, Todd Todd Gunsolley Spry Hosting [EMAIL PROTECTED] ___ 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] Play Voicemails in reverse order?
Are there any settings in the VoicemailMain handler that would cause it to play the voicemails in reverse order (i.e most recent voicemails first, then second most recent, etc.)? The reason is that we generally use email to get our voicemails, but occasionally it would be nice to get them over the phone instead. But since we don't normally call in for them there's always a great many new messages waiting. We really only care about the last ones in this case. An easy compromise between the manual and email systems would seem to be getting them newest first. Did I miss a setting for this though? 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] Wireless LANs and Asterisk
Hi Mike, This is interesting - it's something that I've been considering doing for the Asterisk rollout at my company. We don't have enough Ethernet ports and I'm not thrilled about the expense of re-wiring the place. Have you tried D-Link's dual-channel gear for even more bandwidth, or do you feel that bandwidth is not really a problem? How resilient is 802.11g against interference from other sources? Microwave ovens, gigarange phones, etc. Thanks for reporting your success here to the list, just proves I'm not alone with my funny ideas. ..jurgen On Thu, 10 Feb 2005 12:39:00 -0600, Mike Meyer <[EMAIL PROTECTED]> wrote: > Has anyone had any experience with wireless LANs and Asterisk? > > We have and here are my impressions. > > We configured an Asterisk in the office as a precaution to see how it > would work for our own retail customers. Our office is open space, about > 800 sq ft. (20x40 area). We use Snom200 and Grandstream SIP phones. > > Using the latest Linksys wireless access point (WAP54g) and 3 wireless > bridges (WET54g), I have found that it works most of the time with WPA > encryption on, but will occasionally drop voice (loosing packets). With > no encryption on the WLAN it seems to work without a hitch! Using a less > CPU intense encryption such as 64bit WEP, things also work fine. There > must be too much delay with higher rate encryption. > > Also we had one bridge that seemed to be a week puppy in the litter. It > could only muster 60-70% signal strength. It seemed to have problems > under all configurations. Finally we positioned it such that it too > works well running WEP 64b. I wonder if having 3 wireless bridges in > close proximity would have anything to do with the signal strength? I > would doubt it though. > > Anyone else with other experiences to share regarding wireless LANs and > encryption? I'd me interested to hear them. > > Thanks, > Mike Meyer > GenDesign Corporation > > ___ > 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 > -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ 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] 3G Video Mobile Phone
There is a thread in asterisk-dev, e.g. http://lists.digium.com/pipermail/asterisk-dev/2005-January/008761.html regards, kalus Kuniyoshi Murata wrote: > Hi, > Is there any future possibility that Asterisk will be compatible with > connection to 3G video mobile phone such as Nokia 7600, Nokia 6630 and > many ohters in Japan, Europe and HongKong? > If this become possible, H.323 video clients and 3G mobile phone will be > able to share video conversation, which will be huge in those countries. > In Japan, more than 3 million 3G video mobile phones are already sold. > > 3G phone's format and codec are below. > http://www.nmscommunications.com/Solutions/3G-324M_VIdeo.html > http://www.nmscommunications.com/Solutions/3g-324m.html > http://www.dilithiumnetworks.com/news/Media%20Coverage/IEEE_Multimedia_04July.pdf > > > ___ 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] sip_notify.conf
http://bugs.digium.com/bug_view_page.php?bug_id=0003231 http://bugs.digium.com/bug_view_page.php?bug_id=0003243 On Thu, 10 Feb 2005 19:17:58 +0100, Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 08:14, Wed 09 Feb 05, Kevin P. Fleming wrote: > > Altus Snyman wrote: > > >Good day all > > > > > >What is the file sip_notify.conf for > > > > Read the Mantis bugnotes about it when it was added. It's very useful. > > what's the bug id ? > -- > Michiel van Baak > http://lunteren.vanbaak.info > [EMAIL PROTECTED] > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D > > "Two of the most famous products of Berkeley are LSD and BSD. I don't think > that this is a coincidence." > > ___ > 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] need asterisk consultant, will pay hourly..
I wouldn't hire that guy. He obviously can't follow simple directions. =P -Matthew - Original Message - From: "BestWay CAN" <[EMAIL PROTECTED]> To: "Apu Islam" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, February 10, 2005 2:43 PM Subject: Re: [Asterisk-Users] need asterisk consultant, will pay hourly.. > I'm located at Toronto of Canada as well, I've > implemented a couple of * based projects. My rate is > USD40/hrs. > > Thx, > Howard Song > > --- Apu Islam <[EMAIL PROTECTED]> wrote: > > > I need an asterisk consultant to setup a private > > network. You can be > > anywhere in the world, but I prefer Minneapolis, > > Minnesota since I am > > located here. > > please send me your rate (I am paying this > > personally, so be > > reasonable), things you have done with asterisk, > > level of expertise > > and hourly rate. > > I prefer paypal for the payment, but will consider > > other options as > > well. I will pay in progression as the work > > progresses. > > Please respond by personal mail. > > > > Apu > > [EMAIL PROTECTED] > > ___ > > 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 > > > > > > > __ > Do you Yahoo!? > Yahoo! Mail - 250MB free storage. Do more. Manage less. > http://info.mail.yahoo.com/mail_250 > ___ > 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] Wireless LANs and Asterisk
what aspect exactly are you talking about? VoIP capacity over WLANs, codecs, delay, what? mohammed smadi Colin Anderson wrote: Has anyone had any experience with wireless LANs and Asterisk? I have played with the LocustWorld distro but not at length. Basically, it works. Some sort of QoS tagging for SIP, the docs on it are scanty. It has it's own internal encryption. Never tried it in full force, mostly because of extremely poor WiFi device support. The Intersil Prism2 chipset with the rev. that Linux likes is becoming scarce these days. Great concept, needs more work on device support. www.locustworld.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 ___ 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] asterisk GUI's that supports zap fxs extensions
Are there any gui's that support zap fxs extensions? AMP seems to be one of the more popular gui's but it doesn't support zap fxs devices. Thanks, Jon. ___ 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] Dial SIP peers
Does anyone know why this is not working? exten => s,1,Dial(SIP/192.168.1.8:,20); Connect to 192.168.1.8 on port , with a 20 sec timeout. exten => s,1,Dial(SIP/[EMAIL PROTECTED]:9876,20,r) ; Connect to sip.com port 9876, requesting extension 8500. I defined a sip peer called sip-gateway. If I dial by ip address and port the call goes through on requested port. If I try to dial by peer and port the call will try to go through on the default port. Gene ___ 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] Asterisk - SER Configuration
Hello all! I'm new in this ML and I write you for a suggestion about integrate Asterisk and SER. My idea is to use Asterisk as a local PBX server where users can authenticate and make local calls, but when a user dial a non local number, an asterisk extension call SER Server who redirct to right remote asterisk. Originally I make this only with asterisk where in everyone I setted iax.conf to connect to every remote server. The size of my net in increasing, and then I want to modify it in a "star center" network and I want use ser in center to be sure to avoid rtp traffic. Now, you can point me to a working configuration example for asterisk and ser? Thanks, Alberto Zuin ___ 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] SIP jitter?
So? That's what CVS-HEAD is there for. Adding in experimental patches willy-nilly, especially ones that have the potential to cause huge problems, confounds attempts to isolate bugs and test functionality. Mark does a pretty good job of keeping the HEAD version solid enough to use in production, as most of us running it on a daily basis can attest. huh? HEAD isn't for production. That's part of the game. HEAD is where new stuff goes, then comes the Feature freeze, and then comes the stabilization period. If not, HEAD should be split to not-so-HEAD and HEAD, IMHO. What stops you from applying the patches to your own copy, and then playing with it to your heart's content--like the rest of us? It would work just like it had really been put into CVS-HEAD. I thought it was agreed that the current jitter handling sucked, and a new one is needed. All people running anything apart from IAX, and also those who do, suffers from jitter now. This should go into HEAD. And if it's important to you, you can then *pretend* that it was in the original download. If this is put into HEAD, lots of people test and use it and it stabilizes a lot faster than me sitting in my office hacking it. That should be obvious. roy ___ 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] Logging agents in and out via manager API or other utility. Is it possible?
I have searched and searched, but been unable to find a method to log agents in and out other than through the phone and AgentCallbackLogin or AgentLogin. Anyone know if there is another way? I have looked through the manager API but was unable to find an alternate way to do this. Thanks, ~Senyo ___ 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] Zombie SIP channels
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? - Pedro On Thu, 10 Feb 2005 08:50:35 -0500, Pedro <[EMAIL PROTECTED]> wrote: > What is odd is no meetme is being used. But may be related - thanks! > > Pedro > > > On Thu, 10 Feb 2005 14:37:31 +0100, Florian Overkamp > <[EMAIL PROTECTED]> wrote: > > Hi, > > > > > -Original Message- > > > This is the first time I have seen this so it does not appear to > > > happen too often. Obviously would rather not upgrade if possible has > > > everything seems running fine. But good to know that if it becomes a > > > problem, I can try upgrading to 1.0.3 or later. > > > > If my memory serves me correctly, this is the issue: > > > > http://bugs.digium.com/bug_view_page.php?bug_id=0002938 > > > > It's a two line fix, so if you want you can easily verify and apply manually > > so you don't have to introduce any other new code. > > > > Florian > > > > > ___ 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] Round Robin Strategy doesn't seem to work
Hi, I have configured a call queue as follows: [sales] music = default strategy = roundrobin timeout = 20 retry = 5 reportholdtime = yes announce = queue-sales member => Agent/9021 member => Agent/5901 member => Agent/9020 member => Agent/1114 Now, I would expect all calls to this queue to be delivered first to 9021, then to 5901, then to 9020, then to 1114 (all agents are logged in and available). However, what actually happens is that the first call to the queue is delivered first to 9021 (and on through the queue order from there), but then the next call is delivered first to 5901 (and on through the queue order from there). The third call to the queue is delivered first to 9020, the fourth call to 1114. What I want is for all calls to the queue to be delivered to available agents in the order that they are listed in the member configuration each and every time. Am I missing the point here? Has anyone else run into this problem? Thanks for any insight. Regards, Todd Todd Gunsolley Spry Hosting [EMAIL PROTECTED] ___ 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] WAS: Strategy for a stable IAXy NOW: IAXy vs old P-3
John Novack wrote: >And the only IAX2 box made is the Digium one, with it's current shortcomings ? >From reading through the archives, it seems there is currently no way to reset to factory default, no >written MAC address on an individual box, and some other instabilities requiring frequent resets. Yeah, there's the rub. Dunno if it's worth it, I'm willing to give it a try with a power reset thingy if that's the only thing I have to deal with. That's why I'm asking for opinions and strategies. I only have 1 shot to make this thing work. If the user is dissapointed, they will have lost all confidence in Asterisk. My other option is to drop an old P-3 in the location and IAX it to our production server, then stick in a TDM400. I'd rather to do an IAXy since there's (supposed to be) less meatspace overhead. Eric Wieling wrote: >The IAXy does not support low bandwidth protocols. Hrm, you're right. Looking at the datasheet, closest I can come is ADPCM. Maybe this IAXy thing I should sit on. OK, new question: IAXy vs old P-3 & TDM-400 - Opinions on hassle, reliability, hackability etc? What would you guys do? ___ 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] Context fails so falling back to extension "s" ?
I realize it's bad form, but I'd really appreciate some hand holding here. AMP is making me pull my hair out and the mountain of configuration data in extensions.conf is starting to get to me... I have the first channel configured in zapata.conf to "take" incoming contexts to "from-pstn". I'm using AMP neat "Incoming Calls" configuration page and extensions-additional.conf's include statement is uncommented. The damn thing still doesn't work... I get this error when calling in: Feb 10 15:55:17 VERBOSE[3477]: -- Starting simple switch on 'Zap/2-1' Feb 10 15:55:22 ERROR[3477]: fsk_serie made mylen < 0 (-46) Feb 10 15:55:22 WARNING[3477]: CallerID feed failed: Success Feb 10 15:55:22 WARNING[3477]: CallerID returned with error on channel 'Zap/2-1' Feb 10 15:55:22 VERBOSE[3477]: == Starting Zap/2-1 at from-pstn,s,1 failed so falling back to exten 's' Feb 10 15:55:22 VERBOSE[3477]: == Starting Zap/2-1 at from-pstn,s,1 still failed so falling back to context 'default' Feb 10 15:55:22 WARNING[3477]: Channel 'Zap/2-1' sent into invalid extension 's' in context 'default', but no invalid handler Extension 's'? I thought 's' meant Start, not an actual extension. If there's something I'm not reading or need to read again, don't hesitate to hit me with a clue stick. Regards, aaron.glenn ___ 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] Wireless LANs and Asterisk
Hi, My experience: A handfull of concurrent calls, all works fine on 54mbit. Dont try to go beyond that. Specially when your link is not totally 100%. We tried to do 10 calls on a dedicated Conceptronic AP and all fell down. even with the ulaw codec it was not doable for normal conversations. Even disabling web was not the answer, so we took the good old wires again. And another concern: privacy. As you know, WEP is not that strong. And as long as there is no solid encrypted RTP stream everyone with a laptop is able to monitor/record your calls. just my 2 cents -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence." ___ 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] need asterisk consultant, will pay hourly..
I'm located at Toronto of Canada as well, I've implemented a couple of * based projects. My rate is USD40/hrs. Thx, Howard Song --- Apu Islam <[EMAIL PROTECTED]> wrote: > I need an asterisk consultant to setup a private > network. You can be > anywhere in the world, but I prefer Minneapolis, > Minnesota since I am > located here. > please send me your rate (I am paying this > personally, so be > reasonable), things you have done with asterisk, > level of expertise > and hourly rate. > I prefer paypal for the payment, but will consider > other options as > well. I will pay in progression as the work > progresses. > Please respond by personal mail. > > Apu > [EMAIL PROTECTED] > ___ > 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 > __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250 ___ 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] Startup Question
Hi Daniel. I am a bit confused about sip and IAX, seems IAX is the nativep rotocol for * right? Are hard phones IAX compatible or just SIP? Do they have the same features or diff. ones depending on the protocol and how about bells and whisles, for example Message Waiting notices and such? Hardphone connected directly to ethernet right? So FXS ports are only useful if you have analog conventional phones right? Thx! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Wright Sent: Miércoles, 09 de Febrero de 2005 09:46 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Startup Question Anton Krall wrote: >Guys, Im new to asterisk and voip but Im have a couple of questions >regarding the initial setup. > >1. Im going to install an asterisk server at home, where I have 2 phone >lines, what kind of card do I need to get? I was thinking about 2 X100P >Cards, so 1 can have 2 FXO ports and regarding phones, what else do I need? >Ive seen the Grandstream HandyTone HT-286, I guess that servers as and >FXS devide for attaching analog phones right? Any phone? Is it better >to use those or use IP Phones? > > Look into the TDM400P card with two FXO modules. It is made by digium. You can have up to 4 modules of any kind,FXO or FXS, so you could even put two FXS modules so you have to inside extensions for your regular telephones. You could also just use 2 FXO modules to connect to the POT lines, and go with a sip hard or softphones for the inside extentions. I have not had any experience with any software or hardware sip phones yet. I am currently researching the hardphones so I can get my hands dirty with sip. >The software solution for voip like sokol's one seems a bit unstable >due to winxp limitations :) so Im thniking about going the HW way.. >What do you think? Are those IAX compatible? > > Not sure >2. I checked stuff about IAXNet and FWD, but Im still unclear as to >what really FWD is? And how does it help you on? For example, Im in >Mexico City, can I setup my asterisk box and for example, how can I use >it so that I can make calls to the US by using a US dialtone? Of what does FWD really does? >Is it a network of asterisk boxes connected to each other so FWD users >can talk to each other? Or is it a network with POTS line that can help >you get international dialouts? > >Thank you for your patience as I can imagine this has been asked a lot >of times :( > > > I believe that FWD is a database of users who have accounts so asterisk know how to route calls to other FWD users. Not sure if it will provide a US dialtone. For that you can look at http://www.voip-info.org/wiki-VOIP+Service+Providers for a provider of that. Good luck in your endevors, and welcome to the world of asterisk. Dan >__ >Anton Krall > > >___ >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] asterisk@home scary log
> > > I had the system setup to allow http and ssh. > > > > > > The hack came in through ssh. > > > > For those that aren't heavily involved with security topics, there > > has been many different approachs from many different IP's attempting > > to: > > a) exploit known ssh holes, and, > > b) ssh password guessing > > > > We tend to watch these attempts rather closely through intrusion detection > > tools like snort. As consultants, we are also under retainers to > > assist other companies with securing their facilities and watching > > for exploits. The exploit attempts happen every single day. > > > > There are multiple password guessing tools commonly available on > > the Internet. I eval'ed one of the tools and it took five seconds > > to guess a password that was five characters in length. It took an > > hour to guess a password that was eight characters, and around > > twenty-four hours to guess a password that was eight characters made > > up of uppercase, lowercase and non-alpha characters (eg, complex). > > Regardless, the guessing process is simply how much time does one > > want to devote to doing it (eg, what's the return value for spending > > the time exploiting a system). > > > > It doesn't make much difference whether one exposes telnet or ssh. > > Both can be exploited. But, the more complex you make the password, > > the more time-consuming and difficult it is to guess it. > > > > So, if you must expose either telnet or ssh, make your passwords very > > long and complex. If your O/S has the capability to lockout the account > > after 'xx' failed passwords, then do that. Automatically resetting the > > process after 'y' minutes disrupts the guessing process without the > > hacker knowing it, but still allows you access after that auto reset. > > Using something like seven failed attempts with a five minute reset > > is more then adequate in most cases. > > > > I know that there are opinions in opposed to it, but what about port > knocking in addition to everything we've discusses. Scanners would > simply move along after seeing no open ports. I realize this is a > form of security through obscurity, but it seems in some instances it > would be a good *addition* to *other* security measures (never to be > used as the sole security measure). I could certainly agree with that. ___ 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] dtmfmode and IAX protocol
Joseph wrote: On Thu, 2005-02-10 at 12:15 -0600, Eric Wieling wrote: Joseph wrote: What dtmfmode should I set for IAX protocol? When I dial FWD over IAX it doesn't recognize the numbers I press. IAX and IAX2 do not support a DTMF mode option. They use out of band DTMF *ALWAYS*. According to IAX information page IAX always sends DTMF inline. http://www.voip-info.org/wiki-IAX So why isn't it working? The Wiki page is wrong. ___ 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
