[Asterisk-Users] Recording channels
Hi: Does anyone know if it's possible to record channels in raw PCM instead of GSM format? Thanks, -Stephen- ___ --Bandwidth and Colocation sponsored by Easynews.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] cisco phones problems
Do you have a computer connected to the cisco phone? Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edwin Lam Sent: Wednesday, September 28, 2005 8:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] cisco phones problems Greg Oliver wrote: use the qualify= syntax in your sip.conf and make sure it exceeds the latency between the phones and asterisk server in ms. i've set qualify=3000, the unreachable message still popping up occationally. does it seems normal? the server all the Cisco phones are on the same subnet, i can ping them from the operating system with average respond time less than 1 ms. the drop call still happening and i've notice other warning messages on the console: Sep 28 17:08:21 WARNING[8182]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing
Steve, I hope you didn't feel bad about my opinions. This is why I asked a couple of weeks ago about to clarify regarding T.38 on the 1.0.9 CVS. I read on the mailing list that someone sent an e-mail about the release of the last CVS with the T.38 passthrough included. And then I started to test, ask put a lot of time trying. I understand the time that this involves. What I don't understand is the reason for spread false news. That get me crazy. We're waiting for any feed, Thanks regards. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Wednesday, September 28, 2005 8:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T.38 Faxing Carlos Alperin wrote: This is not an issue about free. The cost of the implementation of the Hylafax, plus the multiport card, plus the channel bank, plus the additional PRI is not cheap. Someone started to announce a couple of weeks ago that the 1.0.9 release was handling T.38. FALSE. All that I was able to get was T.38 detection by the new version. I asked 3 times after someone confirmed that is not even thinked to be on the 2.0 next release. So, I'm far to look for free stuff. I cannot afford to keep testing with customers that had no more patient with me. That is all. Nothing against Asterisk, or the developers. Only with the people that start to announce false espectations. If you look in the bug tracker you will see discussion has only just begun on *how* T.38 fits into *. I hope to be feeding some of the implementation into CVS soon. Right now there is nothing. Regards, Steve ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing
Thanks Steve about your great work. I am very antious to test it! Thanks again. Rosario - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 28, 2005 8:52 PM Subject: Re: [Asterisk-Users] T.38 Faxing Carlos Alperin wrote: This is not an issue about free. The cost of the implementation of the Hylafax, plus the multiport card, plus the channel bank, plus the additional PRI is not cheap. Someone started to announce a couple of weeks ago that the 1.0.9 release was handling T.38. FALSE. All that I was able to get was T.38 detection by the new version. I asked 3 times after someone confirmed that is not even thinked to be on the 2.0 next release. So, I'm far to look for free stuff. I cannot afford to keep testing with customers that had no more patient with me. That is all. Nothing against Asterisk, or the developers. Only with the people that start to announce false espectations. If you look in the bug tracker you will see discussion has only just begun on *how* T.38 fits into *. I hope to be feeding some of the implementation into CVS soon. Right now there is nothing. Regards, Steve ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] How to change ${VM_DATE} in voicemail.conf
Hi,
no, not in french, in italian but the matter is the sameI found the
only solution is to change ${VM_DATE} is to change the source code...
::))
Giorgio
Nathan Pralle wrote:
What exactly are you trying to do? Get it to say the date in French?
Nathan
gincantalupo wrote:
Hi,
I have * 1.0.7 and I have your same problem.
I dunno what version you have but till 1.0.7 simply you cannot.
We have to wait new * versions
I don't find where you can setup the date (${VM_DATE}) in french for
the mail. Is anybody can help me?
___
--Bandwidth and Colocation sponsored by Easynews.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] Is realtime meetme supported by Asterisk
Hi all, Is realtime meetme conference supported by Asterisk? Regards. ___ --Bandwidth and Colocation sponsored by Easynews.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] chan_capi-cm, Euro ISDN bus: 2 extensions on same BRI port not working
On Wed, 28 Sep 2005 [EMAIL PROTECTED] wrote: Hello, I am using a system with an AVM ISDN PCI card (fcpci) and asterisk with chan_capi-cm-0.6. The hardware is connected to a Siemens Hipath 3550 PBX. As a BRI connection has 2 channels and allows 2 simultaneous calls, numbers/MSNs 6391 and 6392 were for provisioned for each channel. The system is working (partly, read on), the trick is the correct cable wiring and setup the PBX's port as S0 Euro Bus. Calls from asterisk to PBX are working ok, really nice! But calls from PBX to asterisk are partly working. Calls to 6391 are okay, but calls to 6392 are not. They simply did not appear on the BRI port (checked with capi debug). The problem seems to be something related to MSN routing. The PBX manual says it is possible to connect up to 64 devices in one S0/BRI port and even show the wiring diagram for that. However, there is not a clear way to define MSN x goes to channel 1, MSN y goes to channel 2 in the PBX management software. I am starting to think it is impossible to configure this through PBX... The PBX manual and on-line help says (as far as I understood) MSNs are configured at the user station but, this time, I was unable to configure MSN x=channel 1, MSN y=channel 2 in the capi.conf file of chan_capi which acts as the station. Actually, I am super-confused! Is it possible to configure the MSN routing as I explained? Is Siemens right, that is, it is up to the station to advertise it is reachable and its MSN? Will chan_capi + fcpci do that? No, your PBX must be configured to signal a call to these MSNs on that S0 port. In capi.conf you just specify which MSNs shall be signaled to Asterisk. Using incomingmsn=* means all MSNs and the extensions.conf can decide what to do with it. If you have a verbose level of 5 and set 'capi debug', you should see at least a CONNECT_IND message on a call. If you don't see that, than the PBX is not signaling this call on that S0 port. In that case, chan_capi and me cannot help you, it's an PBX issue. Armin ___ --Bandwidth and Colocation sponsored by Easynews.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] Dealt with IAreaNet before?
I want to see if any of my fellow Asterisk-Users list members have dealt with these guys. I'm a admin for a VOIP provider, and have encountered a few PBX customers that want consulting/support for the IAreaNet provided Asterisk pbxs. These guys are selling AAH servers to the public, and are (at least the techs that I talked to trying to get our service working) not knowledgeable at all. What's your experience with them if any? Oh, and if you want examples of why I think they're not knowledgeable, I'll be glad to point them out, publicly or privately. SKM ___ --Bandwidth and Colocation sponsored by Easynews.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] Voice Prompts, what do you think? Good voice. Should we record a new prompt-set?
Hello all, I have someone working for me who has a nice phone voice. I looked at some available prompts for asterisk, and found both the free and commercial ones to be pretty horrible. The asterisk ones are good, but I wish I had more to choose from sometimes. My question is, what do you think, should I bother having her record a full spectrum of prompts for asterisk? If you want to hear her voice, the ivr/bri number is 1-914-693-0821 If it is something the community would want, I can spend the time with her, otherwise I am fine with the regular voices. If you wanted any custom prompts done, we could do it. She works for me anyway so its not a big deal, but for anything extensive I should give her some money to do it. This is not a studio setup, but we have a good microphone setup and the quality even on BRI sounds great. I just need to spend some time adjusting the audio levels on my bri and in the wav files. For me, this worked out well, since my own voice I admit sounds very anal on the phone. Feel free to call and listen, but the timeout goes to fax. There is a good chance I will do it, but want some feedback. What would be especially helpful at this point would be suggestions for new prompts for asterisk. This way if I spend like 2 hours doing it with her, at least I can cover any new prompts that may come up. I like the Allison prompts, but sometimes they sound too sexy. April's voice(my employee) is a bit flirty, but I think it works in a good way. If I did it I would probably give away a basic prompt set, but charge $50 or $100 for the extra stuff like pin codes and things that are not normally used for a personal nature. Regards, Gregory Wiktor [EMAIL PROTECTED] Web: www.adcomcorp.com ___ --Bandwidth and Colocation sponsored by Easynews.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] zttest - 100% ?
Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.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] Is realtime meetme supported by Asterisk
Hi :) Am Donnerstag, 29. September 2005 09:03 schrieb Voice over IP: Hi all, Is realtime meetme conference supported by Asterisk? Yes and no. I wrote a patch for an older CVS-Version and will port it to the latest CVS version. Will take 2 or 3 weeks ;) So current versions do not support MeetMe with Realtime. Regards. Kind regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.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] PRI value
Hi group, anyone can explain me the exact difference between pri value in zapata.conf ? ; PRI Dialplan: Only RARELY used for PRI.;; unknown: Unknown; private: Private ISDN; local: Local ISDN; national: National ISDN; international: International ISDN If I use it, I also must use prilocaldialplan = local ? Thanks Giordano ___ --Bandwidth and Colocation sponsored by Easynews.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] Calling voicemail from external phone.
Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail([EMAIL PROTECTED]) Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.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] Calling voicemail from external phone.
On Thu, 2005-09-29 at 10:43 +0200, Arne Morten Johansen wrote: Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail([EMAIL PROTECTED]) Look at DISA -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Asterisk-Users Digest, Vol 14, Issue 178
hi, are any one working on h324 codec with asterisk for 3g video communication ...does asterisk support this regards kiran ___ --Bandwidth and Colocation sponsored by Easynews.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] Calling voicemail from external phone.
We ended up doing it in the c code made it so user can hit * and it will prompt them for a password. We figured that was the easiest way to go about it. Carlos Alcantar Race Technologies, Inc. 101 Haskins Way South San Francisco, CA 94080 P: 650.246.8900 F: 650.246.8901 E: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Thursday, September 29, 2005 1:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Calling voicemail from external phone. On Thu, 2005-09-29 at 10:43 +0200, Arne Morten Johansen wrote: Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail([EMAIL PROTECTED]) Look at DISA -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
Hey ho! We have a functional t.38 implementation for asterisk, but its far from complete. (meaning it doesnt work for all devices, and i only tested it on 1 fax). I hope to take our t.38 developper with me to Astricon and maybe even demo it there. (Maybe oej could bring a fax or two ? :) I will provide more details when i have them... Cheers, Zoa. --- www.asteriskguru.com Rosario Pingaro wrote: Thanks Steve about your great work. I am very antious to test it! Thanks again. Rosario - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 28, 2005 8:52 PM Subject: Re: [Asterisk-Users] T.38 Faxing Carlos Alperin wrote: This is not an issue about free. The cost of the implementation of the Hylafax, plus the multiport card, plus the channel bank, plus the additional PRI is not cheap. Someone started to announce a couple of weeks ago that the 1.0.9 release was handling T.38. FALSE. All that I was able to get was T.38 detection by the new version. I asked 3 times after someone confirmed that is not even thinked to be on the 2.0 next release. So, I'm far to look for free stuff. I cannot afford to keep testing with customers that had no more patient with me. That is all. Nothing against Asterisk, or the developers. Only with the people that start to announce false espectations. If you look in the bug tracker you will see discussion has only just begun on *how* T.38 fits into *. I hope to be feeding some of the implementation into CVS soon. Right now there is nothing. Regards, Steve ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
for the community, I think it is important to have at least t.38 passthrough first then the other devolpments. In this way t.38 can be easly spreaded and catch up more supporters. Rosario - Original Message - From: Zoa [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, September 29, 2005 5:02 AM Subject: Re: [Asterisk-Users] T.38 Faxing - at astricon ? ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
On Thu, Sep 29, 2005 at 05:12:21AM -0400, Rosario Pingaro wrote: for the community, I think it is important to have at least t.38 passthrough first then the other devolpments. In this way t.38 can be easly spreaded and catch up more supporters. What do you mean more supporters. t.38 is only *reliable* way for transporting fax over ip. Fax over g711 is pure luck... ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Dialtone problems with phpagi and asterisk
So, after some research I can provide you with some more information.
According to our employees on every fourth call the dialtone is choppy.
That happens, not like I said first, when we dial trough phpagi AND when
we dial directly with x-pro (but both times through asterisk).
In X-Pro its a bit better, but still a bit choppy.
The other problem is that the whole call is choppy.
What actually could be a bandwith problem... (although it's mysterious
because the first 2-3 weeks that problem hasn't occured (while approx. 5
people worked like today (2005-09-29))
Another problem is that asterisk hangs up the call sometimes after some
seconds without interaction of the user. I don't know why :-/
So I think one part of the solution is a higher bandwidth.
Butthe hangup-problem is another kind of problem I think.
Do you have some ideas what could causes such problems (as described above)
Regards
Michael
Michael Häberle wrote:
Ok :)
the dialplan looks like that (mynumber is a tel-number):
-
[general]
static=yes
writeprotect=no
[telout]
exten = _X.,hint,SIP/41
exten = _X.,1,dial(SIP/${EXTEN})
exten = _X.,2,SetCIDName(anonymous)
exten = _X.,3,dial(SIP/[EMAIL PROTECTED],30,r)
exten = _X.,4,Hangup
-
I dial out of a webapplication, when I press a button, we connect to
asterisk through phpagi.
here are the php-functions:
function startCall($number,$uid) {
$returnValue = false;
$state = getStatus();
if ($state = 0 $state 4) {
$asm = new AGI_AsteriskManager();
if($asm-connect())
{
$call = initCall($asm, $number);
$asm-disconnect();
if (trim($call['Response']) == Error) {
$returnValue = false;
} else {
$returnValue = true;
}
} else {
echo Connect to Asterisk FAILED;
}
} else {
echo Extension already in use;
}
function initCall($asm, $number) {
$call = $asm-send_request('Originate',
array('Channel'=SIP/ . $_COOKIE['extension'],
'Context'='telout',
'Exten'=$number,
'Priority'=1,
'Timeout'=3,
'Async'=false,
'Callerid'='anonymous'));
return $call;
}
for the cookie we have defined a channel in sip.conf.
Later we start to monitor the call (writing *.wav files)
Dont know if that causes the described problems.
If the connection is made an the user on the other side of the line
takes the phone, we phone with x-pro.
Johann wrote:
Without information about your dialplan and what the phpagi script
does there is not much anyone can do. I do not know of any known
issues that may account for the problem you are having.
Update with further information and maybe someone will be able to
provide some insight.
--johann
Michael Häberle wrote:
Does nobody know a solution or an approach to a solution?
Michael
Michael Häberle wrote:
Hi there
In our php-application we use phpagi to communicate with asterisk
(as the voip-client we use x-pro)
Sometimes it occurs that the dialtone is very choppy or not present.
If we dial directly in x-pro this problem has never occured.
I dont know what the problem is, first I thought it is the bandwith
(which is actually a problem), but if that would be the major
problem it wouldnt work in x-pro either, I assume.
Another problem is that sometimes after two or three times ringing
the phone hangs up. No idea what the problem is. (this problem does
not occur with x-pro directly)
We use phpagi 2.14
Suse Linux 8.x
I dont know the asterix version (we downloaded it in july 2005)
Michael
___
--Bandwidth and Colocation sponsored by Easynews.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
--
Immosky AG
Service-Zentrale
Dufourstr. 5
CH-8702 Zollikon-Zürich
Tel+41 (0)43 344 52 52
Fax +41 (0)43 344 52 58
www.immosky.ch
[EMAIL PROTECTED]
___
--Bandwidth and Colocation sponsored by Easynews.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] Don't call
I have set up extension.conf and sip.con with default parameter of UNIVOICE server, but Asterisk show this message when I call a number: Sep 29 11:34:52 WARNING[4179]: chan_sip.c:1899 create_addr: No such host: univoice,Ttr Sep 29 11:34:52 NOTICE[4179]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/100-2331, ) in new stack ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
Kresimir Petrovic schrieb: ... What do you mean more supporters. t.38 is only *reliable* way for transporting fax over ip. Fax over g711 is pure luck... Hi, it is rather a question of IP quality than good luck. I think, 99.9% of all faxes are transported via G.711. Is there any telecom network operator left using ananlog lines? Analog is either the 5 meter way from the fax machine to the BRI/PRI adapter at the wall or maybe even the 1000 meter way from the home/office to the next switch of the network operator. Thus the main difference is the high quality of the managed ISDN network compared to the unmanaged IP network. That's why, imho, analog faxing via VoIP is not pure luck, since one _does_ know the relevant parameter. If I have some means for a very good internet connectivity, faxing will work without problems, if not, you are right, it will become pure luck or just impossible (without T.38). Roger. ___ --Bandwidth and Colocation sponsored by Easynews.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] digits won't play
Hi! I have a strange problem. In an AGI I tell Asterisk to playback a number, for example 31. I then use the AGI SAY NUMBER command and I only hear thirty and then get: -- Playing 'digits/30' (language 'de') Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open (format alaw): No such file or directory Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File digits/1N does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open digits/1N (format alaw): No such file or directory Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open (format alaw): No such file or directory I have looked inside of /var/lib/asterisk/sounds/digits and all files are present... does it have to do anything with the language 'de'? Where do I change that? Thanks, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.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] setting up asterisk as an sms central?
yes On 28. sep. 2005, at 15.54, Tom Hayden wrote: You're going to need to explain a little more. When you say central are you talking about an SMSC? -- Tom On 9/28/05, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi is it possible to use asterisk as an sms central to send SMSes directly to clients on PSTN instead of just communicating with a central? the telco to which we're currently connected doesn't have a central roy ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Recording channels
Try using filename:wav instead of filename:WAV ___ --Bandwidth and Colocation sponsored by Easynews.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] Calling voicemail from external phone.
You want something like this:
exten=_+1NXXNXX,1,SIPDtmfMode(inband)
exten=_+1NXXNXX,2,Wait(4)
exten=_+1NXXNXX,3,Playback(please-enter-your)
exten=_+1NXXNXX,4,Background(ha/mailbox)
exten=_+1NXXNXX,5,DigitTimeout,5
exten=_+1NXXNXX,6,ResponseTimeout,10
exten=_+1NXXNXX,7,WaitExten(10)
exten=_+1NXXNXX,8,Goto(2)
exten=_1NXXNXX,1,VERBOSE(1|${DATETIME} -- Remote Voicemail Access for
${EXTEN} from ${CALLERIDNUM})
exten=_1NXXNXX,2,VoicemailMain(${EXTEN})
exten=_1NXXNXX,3,Hangup
exten=_NXXNXX,1,VERBOSE(1|${DATETIME} -- Remote Voicemail Access for
${EXTEN} from ${CALLERIDNUM})
exten=_NXXNXX,2,SetVar(EXT=1${EXTEN})
exten=_NXXNXX,3,VoicemailMain(${EXT})
exten=_NXXNXX,4,Hangup
exten=i,1,Playback(please-try-again)
exten=i,2,Goto(${NUM},2)
The DTMFMode force is something that our system wants, and the VERBOSE
messages are just for my system's readout so I can watch the output in
realtime. Also, the pattern matching will probably only work if you change
it, as my system has several thousand dids, so I use full numbers instead of
extension style numbers.
Hope this was helpful,
SKM
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Arne Morten Johansen
-Sent: Thursday, September 29, 2005 4:43 AM
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: [Asterisk-Users] Calling voicemail from external phone.
-
-Hey.
-
-How would I set up my dialplan if a user wants to call its
-voicemail from an external phone?
-
-I'm thinking of getting the user to enter its mailbox number.
-
-Something like this:
-
-1. User calls the dedicated voicemail number.
-2. Phone prompts for mailbox number.
-3. Voicemail([EMAIL PROTECTED])
-
- Thanks.
-
-
-
-___
---Bandwidth and Colocation sponsored by Easynews.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
-
___
--Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
for the community, I think it is important to have at least t.38 passthrough first then the other devolpments. In this way t.38 can be easly spreaded and catch up more supporters. What do you mean more supporters. t.38 is only *reliable* way for transporting fax over ip. Fax over g711 is pure luck... apart from t.37, that is, and t.37 is IMHO a far more delicate solution... but then, finding an ATA that supports t.37 is even harder than finding one that supports t.38 roy ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
What do you mean more supporters. t.38 is only *reliable* way for transporting fax over ip. Fax over g711 is pure luck... Hi, it is rather a question of IP quality than good luck. I think, 99.9% of all faxes are transported via G.711. Is there any telecom network operator left using ananlog lines? Analog is either the 5 meter way from the fax machine to the BRI/PRI adapter at the wall or maybe even the 1000 meter way from the home/office to the next switch of the network operator. Thus the main difference is the high quality of the managed ISDN network compared to the unmanaged IP network. That's why, imho, analog faxing via VoIP is not pure luck, since one _does_ know the relevant parameter. If I have some means for a very good internet connectivity, faxing will work without problems, if not, you are right, it will become pure luck or just impossible (without T.38). see http://soft-switch.org/foip.html for a brief explaination of why this generally doesn't work... ___ --Bandwidth and Colocation sponsored by Easynews.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] Variable in call parking
Hi, Can anyone tell me if Asterisk sets some variable when doing a call parking (when someone presses an exten set in features.conf). In can't find this information on a wiki. Cheers ;) Andrew ___ --Bandwidth and Colocation sponsored by Easynews.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] digits won't play
It was indeed the problem with the language 'de' setting, setting the SIP client to US gives me the numbers. On Thursday 29 September 2005 12:00, Christoph Eicke wrote: Hi! I have a strange problem. In an AGI I tell Asterisk to playback a number, for example 31. I then use the AGI SAY NUMBER command and I only hear thirty and then get: -- Playing 'digits/30' (language 'de') Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open (format alaw): No such file or directory Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File digits/1N does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open digits/1N (format alaw): No such file or directory Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open (format alaw): No such file or directory I have looked inside of /var/lib/asterisk/sounds/digits and all files are present... does it have to do anything with the language 'de'? Where do I change that? Thanks, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] chan_cap-cm-0.6 is not working for incomming calls
Hi, I tried to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it will not work: --- snip --- *CLI capi debug CAPI Debugging Enabled -- CONNECT_IND (PLCI=0x101,DID=97,CID=0179903,CIP=0x1,CONTROLLER=0x1) == reventix: Incoming call '0179903' - '97' -- reventix: info element CALLED PARTY NUMBER -- reventix: info element CHANNEL IDENTIFICATION 89 Urgent handler -- reventix: info element CALLED PARTY NUMBER -- reventix: info element CHANNEL IDENTIFICATION 89 Urgent handler == reventix: CAPI Hangingup == reventix: Interface cleanup PLCI=0x101 Urgent handler --- snap --- --- snip: capi.conf --- [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 ;ulaw=yes [reventix] ;ntmode=yes isdnmode=msn incomingmsn=* controller=1 group=1 ;prefix=0 softdtmf=on relaxdtmf=on accountcode= context=reventix-incoming holdtype=local ;immediate=yes ;echosquelch=1 ;echocancel=yes echocancelold=yes ;echotail=64 ;bridge=yes ;callgroup=1 ;deflect=1234567 devices=2 --- snap --- Asterisk is located behind a EURACOM PBX. With chan_capi-cm 0.5.4 it worked good. What's going wrong in my configuration? Regards Bastian ___ --Bandwidth and Colocation sponsored by Easynews.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] Asterisk registering with vonage
Hello everyone. Ive seen postings for connecting asterisk to vonage but Im still having trouble achieving that. I have a vonage softphone and I'm trying to register to vonage using asterisk. I have not had any luck. I am behind a firewall. I've successfully gotten xlite to connect and work from the same network. When I change the port setting in [general] to 5061, I am able to register then, but all my other extensions cannot connect. I thought that specifying the port in the register string should be sufficient. Can someone please tell me what I'm doing wrong? Below is my sip configuration. sip.conf: == [general] port=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 allow=g723 context=from-sip-external ; Send unknown SIP callers to this context callerid=Unknown register=1XX:[EMAIL PROTECTED]:5061 [vonage] username=1XX type=friend secret=PASSWORD port=5061 nat=yes host=sphone.vopr.vonage.net fromuser=1XX fromdomain=sphone.vopr.vonage.net dtmfmode=rfc2833 auth=md5 [vonage_inbound] username=1XX type=friend secret=PASSWORD port=5061 nat=yes insecure=very host=sphone.vopr.vonage.net fromuser=1XX fromdomain=sphone.vopr.vonage.net context=from-pstn canreinvite=no auth=md5 allow=all == CLI sip debug : : : Retransmitting #5 (no NAT): REGISTER sip:sphone.vopr.vonage.net SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK674c6ab6 From: sip:[EMAIL PROTECTED];tag=as5c8219d8 To: sip:[EMAIL PROTECTED] Call-ID: 2ba197c665043c8a08a6119e733083df [!at] 192.168.1.10 (replace the [!at] with a @) CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 to 216.115.25.198:5061 == CLI sip show registry Host Username Refresh State sphone.vopr.vonage.net:5061 1XX 120 Request Sent ___ --Bandwidth and Colocation sponsored by Easynews.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] sip calleridnum
Hello,
I have one simple question. Is it bug that for From: 1234 1234
sip:[EMAIL PROTECTED];user=phone; ${CALLERIDNUM} is 1234 instead of 5678 ?
Asterisk 1.0.9
--
Michal Olejnik
___
--Bandwidth and Colocation sponsored by Easynews.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] chan_cap-cm-0.6 is not working for incomming calls
On Thu, 29 Sep 2005, Bastian Schern wrote: Hi, I tried to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it will not work: --- snip --- *CLI capi debug CAPI Debugging Enabled -- CONNECT_IND (PLCI=0x101,DID=97,CID=0179903,CIP=0x1,CONTROLLER=0x1) == reventix: Incoming call '0179903' - '97' -- reventix: info element CALLED PARTY NUMBER -- reventix: info element CHANNEL IDENTIFICATION 89 Urgent handler -- reventix: info element CALLED PARTY NUMBER -- reventix: info element CHANNEL IDENTIFICATION 89 Urgent handler == reventix: CAPI Hangingup == reventix: Interface cleanup PLCI=0x101 Urgent handler --- snap --- --- snip: capi.conf --- [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 ;ulaw=yes [reventix] ;ntmode=yes isdnmode=msn incomingmsn=* controller=1 group=1 ;prefix=0 softdtmf=on relaxdtmf=on accountcode= context=reventix-incoming holdtype=local ; immediate=yes ; echosquelch=1 ; echocancel=yes echocancelold=yes ; echotail=64 ; bridge=yes ; callgroup=1 ; deflect=1234567 devices=2 --- snap --- Asterisk is located behind a EURACOM PBX. With chan_capi-cm 0.5.4 it worked good. What's going wrong in my configuration? It seems your configuration is correct and your provider is not sending SETUP/SENDING-COMPLETE on incoming call (chan_capi-cm now waits for these elements). Since more people do have this problem, I have added the option of capi.conf 'immediate=yes' to support the old behaviour. This is in CVS of chan_capi-cm now and will be part of version 0.6.1 soon. Armin ___ --Bandwidth and Colocation sponsored by Easynews.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] Music on Hold Quality
On Wed, September 28, 2005 5:41 pm, Matt said: I have heard this issue when on hold with Cisco and Vonage... Idon't think it's an asterisk problem I htink it's a G711 problem... orgsm problem. Basically they are made for voice, and I think the music goes outside their encoding ranges... sound logical?Rolling off the high-end of the audio range above 4 KHz helps. Try madplay instead of mpg123 and also have the playback gain reduced~12 db with this musiconhold.conf line:default =custom:/usr/local/lib/asterisk/mohmp3/,/usr/local/bin/madplay--mono --sample-rate=8000 --attenuate=-12 --output=raw:- -kim ___ --Bandwidth and Colocation sponsored by Easynews.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] Calling voicemail from external phone.
Arne, been posted many times do a search on the voip-info site on Disa. Does exactly what you are after. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Arne Morten Johansen Sent: Thursday, 29 September 2005 4:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Calling voicemail from external phone. Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail([EMAIL PROTECTED]) Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Dealt with IAreaNet before?
I bought some USB soundcard/handsets from them with no issues. I did not deal with them on any PBX or config issues though. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past.--- - --- - - - - - - - -- - - - --- - -- - - --- - - -- - - - -- - - - "Sherwood McGowan" [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... I want to see if any of my fellow Asterisk-Users list members have dealt with these guys. I'm a admin for a VOIP provider, and have encountered a few PBX customers that want consulting/support for the IAreaNet provided Asterisk pbxs. These guys are selling AAH servers to the public, and are (at least the techs that I talked to trying to get our service working) not knowledgeable at all. What's your experience with them if any? Oh, and if you want examples of why I think they're not knowledgeable, I'll be glad to point them out, publicly or privately. SKM ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] Major bug solved in IPSwitchBoard
I have been working on solving a major issue with IPSwitchBoard. It was reported that IPS would use all available memory and get the PC to grind to a halt. I could not understand this as I had it running on many different PCs in Denmark. I now found the bug: IPS would crash on any PC that had . configured as decimal point (in Denmark we use ,) this meant that IPS would consume all memory as it tried to make fonts 100 times larger than meant to be. I have now installed a PC with US windows on it just for testing. I hope that I have not caused you too much trouble. Download the software for FREE: http://ipsoftware.thorben.dk IPSwitchBoard is totally customizable and will give you, among other things: Unattended/attended transfers. Park calls and retrieve/forward them again. Organize all your SIP, IAX extensions (automatically retrieved from Asterisk). Monitor all extensions, queues and Parked Calls. Dynamically log extensions in and out of queues. Set Do Not Disturb on Extensions and give a reason Set Call forwarding for extensions Set Dual call for extensions Monitor multiple calls on an Extension/Queue Monitor Extension online status Totally customizable Make you own skins with logos etc. Make your own buttons DND, Online, Queue Status, Call Forward, Dual Call, MWI etc. IPDesigner is a unique tool for customizing and setting up IPSwitchBoard. With IPDesigner you can design your own IPSwitchBoard with Company logo and all the buttons you need for the Operator. IPSwitchBoard works with .IPS files. These files contain everything needed for IPSwitchBoard such as bitmaps, server and extension configuration. You can build .IPS files with IPDesigner. Download the manual to read more: http://www.ipdanmark.dk/IPSwitchBoard/IPswitchBoard%20Manual.pdf Thorben ___ --Bandwidth and Colocation sponsored by Easynews.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] soft phones for Zaurus PDA
Can anyone recommend a soft phone for my Zaurus PDA that will play well with Asterisk? TIA Jason ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Dealt with IAreaNet before?
Well, I had an issue with them charging funds on PayPal for stuff they never sent out, and they justsat on their hands for 3 months till I contacted them to get a refund back (took me some time to check my paypal), and then it took them 3 weeks to refund me. Nir S From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of StevenSent: Thursday, September 29, 2005 2:07 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: Dealt with IAreaNet before? I bought some USB soundcard/handsets from them with no issues. I did not deal with them on any PBX or config issues though. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past.--- - --- - - - - - - - -- - - - --- - -- - - --- - - -- - - - -- - - - "Sherwood McGowan" [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... I want to see if any of my fellow Asterisk-Users list members have dealt with these guys. I'm a admin for a VOIP provider, and have encountered a few PBX customers that want consulting/support for the IAreaNet provided Asterisk pbxs. These guys are selling AAH servers to the public, and are (at least the techs that I talked to trying to get our service working) not knowledgeable at all. What's your experience with them if any? Oh, and if you want examples of why I think they're not knowledgeable, I'll be glad to point them out, publicly or privately. SKM ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Calling voicemail from external phone.
I just copied the *98 extension to the extension of one of our DID numbers. So now if I dial 5686 from inside or 1-xxx-xxx-5686 from outside, I get the same prompts as dialing *98. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Arne Morten Johansen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail([EMAIL PROTECTED]) Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] setting up asterisk as an sms central?
Well, it depends what country you're in and what kind of protocols you are using. Here in the US, I prefer to *not* use asterisk and use the perl module Net::SMPP to handle my SMS traffic between my gateway/aggregator and the carriers SMSC. It's somewhat easier to configure with special services, and database application. You could also take a look at a peice of software called Kannel, which handles SMS aggregation. However, if you are in Europe you should probably be able to get a fixed-line SMS service, and use the Asterisk cmd SMS: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Sms Cheers, -- Tom, On 9/29/05, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: yes On 28. sep. 2005, at 15.54, Tom Hayden wrote: You're going to need to explain a little more. When you say central are you talking about an SMSC? -- Tom On 9/28/05, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi is it possible to use asterisk as an sms central to send SMSes directly to clients on PSTN instead of just communicating with a central? the telco to which we're currently connected doesn't have a central roy ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.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 1.0.9 + spandsp 0.0.2pre20 = crash on boot
I have asterisk 1.0.9 installed with spandsp 0.0.2pre20. Asterisk crashes on boot while loading app_txfax.so app_rxfax.so. If I move the files out of /usr/lib/asterisk/ modules asterisk boots fine. Running on FC3, Linux asterisk.crocker.com 2.6.11-1.27_FC3smp #1 SMP Tue May 17 20:43:11 EDT 2005 i686 i686 i386 GNU/Linux Edit /etc/ld.so.conf file, add the /usr/local/lib directory then do ldconfig Asterisk doesnt start because app_tx/rx miss a library. Must be something else. I added /usr/local/lib during the spandsp install. [EMAIL PROTECTED] ~]# cat /etc/ld.so.conf include ld.so.conf.d/*.conf /usr/X11R6/lib /usr/lib/mysql /usr/local/lib [EMAIL PROTECTED] ~]# ls -al /usr/local/lib/ total 888 drwxr-xr-x 2 root root 4096 Sep 28 14:13 . drwxr-xr-x 14 root root 4096 Sep 28 13:37 .. -rw-r--r-- 1 root root 440142 Sep 28 14:12 libspandsp.a -rwxr-xr-x 1 root root822 Sep 28 14:12 libspandsp.la lrwxrwxrwx 1 root root 19 Sep 28 14:13 libspandsp.so.0 - libspandsp.so.0.0.2 -rwxr-xr-x 1 root root 419802 Sep 28 12:49 libspandsp.so.0.0.2 [EMAIL PROTECTED] ~]# Any other ideas? -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --Bandwidth and Colocation sponsored by Easynews.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] Tiny Echo on PSTN via Zaptel
I have had problems between the sip/FXO lies and was able to kill the echo by trying different combinations of the echocancel line to 64 (I think it has settings in 32 bit increments) Just kept trying different ones till it went away. Here is my config: group=1 context=line1 signalling=fxs_ks usecallerid=yes callerid=asreceived echocancel=64 echocancelwhenbridged=yes callgroup=1 rxgain=1.2 channel = 1 context=line2 signalling=fxs_ks usecallerid=yes callerid=asreceived echocancel=64 echocancelwhenbridged=yes callgroup=1 rxgain=1.2 channel = 2 musiconhold=default context=line3 signalling=fxs_ks usecallerid=yes callerid=asreceived echocancel=64 echocancelwhenbridged=yes callgroup=1 rxgain=1.2 channel = 3 group=2 context=line4 signalling=fxs_ks usecallerid=yes callerid=asreceived echocancel=96 echocancelwhenbridged=yes callgroup=2 channel = 4 Hope this helps! t o n yOn 9/28/05, Shaw Terwilliger [EMAIL PROTECTED] wrote: I'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhinochannel banks (one 12FXO/12FXS, the other 24 FXS).So it's an analogphone on the inside connected to one of the FXS ports, and PSTN lineconnected to one of the FXO ports. My problem is that as soon as I hear the _first_ ring when I dial outthrough the PSTN line, I hear a tiny echo on the phone (I estimatebetween 20ms and 40ms), which never goes away for this call.It's just loud enough to bug the heck out of me when I'm talking (I could estimatethe gain relationship with ztmonitor if it would help).The soundon the recipient end of the connection is perfect.If I make a call from the phone to the another internal extension (another phone on an FXS port), there is no echo.If I call into Comedian mail,there is no echo.I've checked all my gains.The internal gains were a bit loud to startwith because of the powered phones, but now they all fall comfortably within ztmonitor's dynamic range display.The PSTN line is pretty good at tx 0 andrx 0, so I left it.I've tried turning them down, but that didn't killthe echo.My zapata.conf includes these lines at the bottom: echocancel=yesechocancelwhenbridged=noechotraining=yescontext=companyA-pstntxgain=0.0rxgain=0.0signalling=fxs_ksgroup=1channel=1-7context=companyB-pstntxgain=0.0 rxgain=0.0signalling=fxs_ksgroup=2channel=11-12context=internaltxgain=-12.0rxgain=-8.0signalling=fxo_lscallerid=asreceivedgroup=3channel=13-48When the calls are connected, I can use zap show channel 11 and verify that the echo cancellation is ON.But I can still hear one.I've also tried echocancelwhenbridged=yes, but it didn't make any difference.--Shaw Terwilliger [EMAIL PROTECTED]SourceGear LLC-BEGIN PGP SIGNATURE-Version: GnuPG v1.2.5 (GNU/Linux)iD8DBQFDOuxyPEbgvbl6u4ERAh/gAKCAu+gAr9TYsMG5TYqozV3ebIvezwCdGIHkkftrFLFK4purux/sVIPRhKk= =9WG3-END PGP SIGNATURE-___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A.G. (Tony) NicholsI.S. Manager ___ --Bandwidth and Colocation sponsored by Easynews.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] cisco phones problems
Edwin, They are on the same VLan and on the same Subnet? If that's the case check you switch log for details, if you havent changed anything on the * Server. Looks like a serious package lost, even with a high segment this shouldn't occur. At least for the info you send, these are the POF. *Server, NIC or drivers, be sure everything its right Switch, Check your vlan settings, vlan encap (none,dot1q/isl/??), spanning-tree loops Phone, are u using vlan encapsulation? Check the settings. Are u using autoprovission? Post the file, phones SW version, etc. IMHO, seems like a vlan missconfiguration between your phones and switch. LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edwin Lam Sent: Wednesday, September 28, 2005 9:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] cisco phones problems Carlos Alperin wrote: How do you power your cisco phones? Are you using any 35xx XL switch? If that is the case, you need to redo your switch settings enabling QOS. By default CISCO didn't enable it on their switches made for POE the phones. they are all powered with external power supply. we're not using Cisco switches but the server all the phone units are on the same vlan and i've set the QOS priority to highest on our switch for that vlan. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Calling voicemail from external phone.
Yup that's what I was going to suggest you do.. we've been using that and it works great. On 9/29/05, Steven [EMAIL PROTECTED] wrote: I just copied the *98 extension to the extension of one of our DID numbers. So now if I dial 5686 from inside or 1-xxx-xxx-5686 from outside, I get the same prompts as dialing *98. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Arne Morten Johansen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail([EMAIL PROTECTED]) Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Audio Files, Filtering, and Formats for Asterisk
I listened to all the demos you showed. My ear discerns a little muffling and minor slushiness in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to de-ess the recording. Or, I could just be rather tired. Either way, your best bet is to have the system use WAV files but also make available GSM and ulaw versions of the same files. This is so that the system can pick (and it does this automatically) the best format that requires the least amount of CPU power for transcoding. If I remember correctly, FXO/FXS cards can use straight PCM files. What I do to prepare a file is this: I record in 48Khz 32 bit mode (32 bit is just higher resolution so processing has more to work with, 16 bit is fine since it's the end result), then I compress with de-essing and pop removal (maximizes volume, removes slushy or whistling s's and popping p's), then run through an FFT (Fast Fourier Transform) to bring the frequency response within the natural range of a telephone (if I remember correctly it's within the 300 - 4K range, I can check if need be). Then I reduce the file to a 8Khz, 16 bit mono file. Then I check the audio again, and normalize (like compression but only raises the whole file to where the highest peak of audio reaches the level requested, instead of raising or lowering the level on a dynamic basis by using readahead of a couple milliseconds). I usually normalize to around -3db. The end result is a WAV file that sounds good over the phone. I then put the file on the asterisk server (or another server with sox installed) and convert to gsm, ulaw, and alaw (using the original WAV, not using converted gsm or whatever). Hope this was helpful, and I wish you luck. If nothing else, for like $50 or so (depending on how many files there are) I would be willing to take a series of WAV files from you and perform the filtering and whatnot for you, and supply gsm, wav, ulaw, and alaw. I'm also going to just try and get a series of recordings together through my partner's studio in Phoenix AZ, make the files available to the community, and take up donations for the studio and processing time. Talk to you soon. I'm going to cc the asterisk-users list for this, so that the community can benefit from the information. SKM --Original Message- -From: Stephen Bosch [mailto:[EMAIL PROTECTED] -Sent: Thursday, September 29, 2005 12:18 AM -To: Sherwood McGowan - -Hi, Sherwood: - -If you'll forgive me, I'd like to e-mail you directly with a -few comments and questions. - -Sherwood McGowan wrote: - I have to barge in here... - - Guys, the reason the audio sounds like hell is most probably for - mulitple reasons. First, what codec is the prompt encoded? -Next, what - codec is the client using? If you're using, for example SIP clients - with G711u, asterisk has to re-encode (on the fly I might add) the - prompts to ulaw format to be usable to that client. Only on -GSM based - clients would the audio sound mildly like the actual file. Also, - what's the current load, memory, how many calls are running on the - server, how many calls on hold listening to that music? If -you're not - encoding only one format for those calls, guess what, -you're incurring load to re-encode to each format in use on the fly. - -It's obvious I have much to learn about file formats, but -bear with me here. - -In this particular case, the card is the Digium TDM-400 and -the phone is an analog phone (an old ITT touch-tone phone -- -indestructible and superb sound quality). I don't know if -there's any transcoding happening on-the-fly. - -The thing is, though -- the prompts don't sound that great -when I play them with play on my workstation -- and it's -using the libgsm library, so there shouldn't be any -transcoding happening there. - -All this time I've just been talking about the prompts -provided with Asterisk. Asterisk records voicemail (for -example) in the same GSM format that these prompts are in (if -the recordings sound as poor as these prompts do, I don't -know whether I'll even be able to use Asterisk for this application). - -While the delivery is professional and I'm sure the original -sources sound great, these GSM files don't sound so hot. -They're muffled and there is a slight bit of static. I've -attached the demo-congrats.gsm file for example. Try it yourself. - - I am a music producer, have been for several years. One of -the things - I do on the side from my day job as a VOIP Admin/Engineer is make - prompts and music for customer's PBXs. I typically make a WAV file, - and then do my filtering, compressing, and finally -normalizing, then - save it. Finally, I put the files in the server in question and use - sox to re-encode multiple - formats: gsm, ulaw, alaw, etc You'll find the results are much - better than just
[Asterisk-Users] Asterisk Echo problems, Urgent, please help,
Hi all, I hope someone can help, as I have an urgent problem. I've got a production Asterisk server thats been deployed, but we are seeing a strange voice echo problem. There is about a 250ms echo for the users in the office, and they are hearing their own voice back at them. I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of memory. The client SIP phones are Polycom Soundpoint IP600's, WiFi ZyXel 2000w handsets, and X-Lite (free) PC clients. All see the same problem. There is a bridge into the POTS (BT's SystemX) using a Voicetronix OpenSwitch12 card and the vpbhp driver. The echo occurs on both SIP-POTS calls, and SIP-SIP calls. I've tried a number of volume adjustments to correct the echo but it is always the same. If anyone has any ideas I'd really appriciate some help, as this is a major urgency, Many many thanks, Ian Bonham _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: * mod core dump help
You could look up at '/tmp' if you are runing * in safe mode ...guess that help you G. ___ --Bandwidth and Colocation sponsored by Easynews.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 Echo problems, Urgent, please help,
What kind of POTS trunks/cards are you using? -- Tom On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote: Hi all, I hope someone can help, as I have an urgent problem. I've got a production Asterisk server thats been deployed, but we are seeing a strange voice echo problem. There is about a 250ms echo for the users in the office, and they are hearing their own voice back at them. I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of memory. The client SIP phones are Polycom Soundpoint IP600's, WiFi ZyXel 2000w handsets, and X-Lite (free) PC clients. All see the same problem. There is a bridge into the POTS (BT's SystemX) using a Voicetronix OpenSwitch12 card and the vpbhp driver. The echo occurs on both SIP-POTS calls, and SIP-SIP calls. I've tried a number of volume adjustments to correct the echo but it is always the same. If anyone has any ideas I'd really appriciate some help, as this is a major urgency, Many many thanks, Ian Bonham _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.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] chan_cap-cm-0.6 deflect support
Hi, I've recently reinstalled a Diva in my asterisk server (alongside a QuadBRI :-) to test the nice features Armin has been adding in chan_capi. The capi.conf format has changed, so my question is how do I define a deflect= statement for different incoming MSN's? I've tried to define a section for each (group of) MSN with a different deflect. Is that correct? [DIVA1] isdnmode=msn incomingmsn=146472130 controller=1 group=5 accountcode=diva context=default deflect=0612110618 devices=2 [DIVA2] isdnmode=msn controller=1 group=5 incomingmsn=146472131,146472132,146472133,146472134,146472136 accountcode=diva context=default deflect=0187174173 devices=2 [DIVA3] isdnmode=msn controller=1 group=5 incomingmsn=146472135 accountcode=diva context=default deflect=0780350669 devices=2 Also, is there a way to detect that a SIP phone has an active forward number and capi-deflect any incoming calls to that number? ___ --Bandwidth and Colocation sponsored by Easynews.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] maximum retries exceeded on call
Hi, I phone with phpagi and/or x-pro. Sometimes I get this warning in the asterisk-console: maximum retries exceeded on call. I noticed when this message shows up, asterisk hangs up the call (even when i'am in the middle of a call, according to our employess) When they restart x-pro it seems to work properly again (at least some time). Asterisk and the clients are in the same LAN. I read the FAQ at voip-info.org but it didn't help. Here is my sip.conf -- [general] context=telin port=5060 bindaddr=0.0.0.0 srvlookup=yes toos=lowdelay allow=g726 allow=ulaw rtptimeout=60 rtpholdtimeout=300 useragent=EASYCOM nat=yes - after that comes the whole register-thing here comes a sample user (all are the same) - [user] context=telout type=friend secret=XXX dtmfmode=rfc2833 host=dynamic allow=all canreinvite=no - in x-pro everything is standard (nothing changend but the network-settings and sip-proxy) Since Iam neither a linux nor a asterisk-crack, I don't really have a clue what's going on. Hope you can help me :) Regards Michael -- Immosky AG Service-Zentrale Dufourstr. 5 CH-8702 Zollikon-Zürich Tel+41 (0)43 344 52 52 Fax +41 (0)43 344 52 58 www.immosky.ch [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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] Change language to spanish
Hi there: Is there any way to change the language in asterisk to Spanish...I mean I want to change all the dialogs to Spanish in my * box can u help me pls. Hector ___ --Bandwidth and Colocation sponsored by Easynews.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] Caller ID, Attended Transfers, Polycom
We have contracted with an outside call center to provide sales for a certain product. We want to be able to transfer people over to those dedicated sales agents using an attended transfer (so we can prepare them with as much information as we have), to a regular extension. So far, so good. All of this is working just great. We want the caller's information presented as the CallerID so that the outside staff can use the information for tracking the calls. When the call leg is created that starts the transfer sequence, the CallerID is set to our outgoing CallerID by the Polycom phone. Are there any good tricks to determine how to set it, such that it will match the caller ID of the number that called? ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
Zoa wrote: Hey ho! We have a functional t.38 implementation for asterisk, but its far from complete. (meaning it doesnt work for all devices, and i only tested it on 1 fax). I hope to take our t.38 developper with me to Astricon and maybe even demo it there. (Maybe oej could bring a fax or two ? :) I will provide more details when i have them... What have you implemented so far? Terminal mode? Gateway? UDPTL? TPKT? RTP? IAX2? SIP? etc.? Regards, Steve ___ --Bandwidth and Colocation sponsored by Easynews.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] Getting asterisk to send e-mail to mailbox-users
Ok. I've been searching the wiki and google for a long time now. HOW do I enable asterisk to send mail when users get new messeages in there mailbox? Do i need to change mailcmd in voicemail.conf? Regards, Arne morten ___ --Bandwidth and Colocation sponsored by Easynews.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] Getting asterisk to send e-mail to mailbox-users
Solo probando -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Arne Morten Johansen Enviado el: Jueves, 29 de Septiembre de 2005 07:50 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Getting asterisk to send e-mail to mailbox-users Ok. I've been searching the wiki and google for a long time now. HOW do I enable asterisk to send mail when users get new messeages in there mailbox? Do i need to change mailcmd in voicemail.conf? Regards, Arne morten ___ --Bandwidth and Colocation sponsored by Easynews.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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.8/114 - Release Date: 28/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.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] Remotely dialing calls from a polycom phone
I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is fine, except it can be confusing for the receptionist. Why do I have to answer the phone so I can page someone? What I'm looking for is a little more direct integration between the touchpad and the polycom. So, the question is, does anyone know of a way to initiate a call on a polycom via the remote access port. I was thinking of something similar to the way you can reboot one using a NOTIFY message containing check-sync. I've tried monitoring the sip messages between the phone and asterisk, but so far have only succeeded in making a phone call itself. Any suggestions would be appreciated. Eric. ___ --Bandwidth and Colocation sponsored by Easynews.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 Echo problems, Urgent, please help,
I'm using a Voicetronix OpenSwitch12 analouge card. It's on the latest driver for Asterisk (2.4.9) and has echo cancellation turned on. This works fairly well on on SIP-POTS calls after it trains up, but there is still a small echo. The SIP-SIP calls are really echoy though. Cheers, Ian (Sorry, only hit 'reply' last time, not 'reply all'!) From: Tom Hayden [EMAIL PROTECTED] Reply-To: Tom Hayden [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk Echo problems, Urgent, please help, Date: Thu, 29 Sep 2005 09:02:05 -0400 What kind of POTS trunks/cards are you using? -- Tom On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote: Hi all, I hope someone can help, as I have an urgent problem. I've got a production Asterisk server thats been deployed, but we are seeing a strange voice echo problem. There is about a 250ms echo for the users in the office, and they are hearing their own voice back at them. I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of memory. The client SIP phones are Polycom Soundpoint IP600's, WiFi ZyXel 2000w handsets, and X-Lite (free) PC clients. All see the same problem. There is a bridge into the POTS (BT's SystemX) using a Voicetronix OpenSwitch12 card and the vpbhp driver. The echo occurs on both SIP-POTS calls, and SIP-SIP calls. I've tried a number of volume adjustments to correct the echo but it is always the same. If anyone has any ideas I'd really appriciate some help, as this is a major urgency, Many many thanks, Ian Bonham _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
On Thu, Sep 29, 2005 at 12:38:40PM +0200, Roy Sigurd Karlsbakk wrote: for the community, I think it is important to have at least t.38 passthrough first then the other devolpments. In this way t.38 can be easly spreaded and catch up more supporters. What do you mean more supporters. t.38 is only *reliable* way for transporting fax over ip. Fax over g711 is pure luck... apart from t.37, that is, and t.37 is IMHO a far more delicate solution... but then, finding an ATA that supports t.37 is even harder than finding one that supports t.38 ok, I ment to said real time fax over ip ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
I didnt implement anything myself and am not very familiar with t.38, but i think its udptl, sip, iax2, and soon gateway too. I will try to get a little more info from the developper when he gets back.. Maybe its even based on your work, i should check. Sorry for the incomplete reply, i just don't know... Cheers, Zoa. Steve Underwood wrote: Zoa wrote: Hey ho! We have a functional t.38 implementation for asterisk, but its far from complete. (meaning it doesnt work for all devices, and i only tested it on 1 fax). I hope to take our t.38 developper with me to Astricon and maybe even demo it there. (Maybe oej could bring a fax or two ? :) I will provide more details when i have them... What have you implemented so far? Terminal mode? Gateway? UDPTL? TPKT? RTP? IAX2? SIP? etc.? Regards, Steve ___ --Bandwidth and Colocation sponsored by Easynews.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 signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: [Asterisk-biz] Problem with sending fax froma SIP extension
Why is what he is doing different than having the fax machine on a Sipura ATA? Just because both those ports are on the pci card that doesn't make them not Voice in betweenif I'm wrongeh...oh well -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, September 28, 2005 9:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: [Asterisk-biz] Problem with sending fax froma SIP extension Before responding please read my message again, in my message I worte: 'faxing over VoIP' what you are doing is NOT over VoIP. On 9/28/05, u [EMAIL PROTECTED] wrote: On 9/27/05, C F [EMAIL PROTECTED] wrote: 1. Search the archives 2. Search again 3. Now search the internet 4. The fact is that faxing over VoIP without T.38 doesnt really work. Because it works 60% of the time it doesn't mean that it works, until it works 99% of the time. Since Asterisk does NOT support T.38 it doens't support faxing over VoIP other solutions are available, search the wiki. On 9/26/05, Mark Armstrong [EMAIL PROTECTED] wrote: I am having exactly the same problem. Any information would be appreciated. Faxing is a commercial proposition that my team is working on, any limitations of Asterisk or E1 cards needs to be highlighted. Regards Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, 27 September 2005 9:42 AM To: [EMAIL PROTECTED]; Commercial and Business-Oriented Asterisk Discussion Subject: Re: [Asterisk-biz] Problem with sending fax from a SIP extension why don't you post this to asterisk-users? On 9/22/05, Andy Kuo [EMAIL PROTECTED] wrote: Hi All, I'm having problem sending fax from SIP extensions (Linksys PAP2) through Asterisk Zap channels (ISDN PRI). The SIP extensions can receive fax without problems, but sending fax fails most of the time. Does anyone have this problem? Please advice. Thank you. AK I have a fax connected to a zaptel pci card (FXS). Fax are sent right into an FXO card connected to my PSTN. I don't experience problems but then again I am a light user. Just out of curiosity, is the problem related to sending FAX via IAX or SIP to another asterisk server? if asterisk will support T.38 passthru to say Hylax Fax, does that mean I need a Hylax fax daemon in every asterisk server? ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Prueba
Solo probando ___ --Bandwidth and Colocation sponsored by Easynews.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] Change language to spanish
hi Hector. Just use the * command SetLanguage(), passing as argument es http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SetLanguage for more info check the link, and other related links at the bottom of that page. best regardsOn 9/29/05, Hector Elias Menjivar [EMAIL PROTECTED] wrote: Hi there:Is there any way to change the language in asterisk to Spanish...Imean I want to change all the dialogs to Spanish in my * box can u help mepls.Hector___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation sponsored by Easynews.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] Getting asterisk to send e-mail to mailbox-users
Hi Arne, In /etc/asterisk/voicemail.conf, under the [default] section, you need to declare the users like this : box# = passnumber for box, Name of User,email address e.g. 221 = 1234,Ian Bonham,[EMAIL PROTECTED] Do that for each mailbox you require. Then in the sources directory, under 'contrib/scripts' there is a script 'addmailbox'. Run that and add the mailboxes you have declared in voicemail.conf. Just enter over the first option (which will set it to default) then type in the mailbox number. This creates the mailboxes. Then, if you are on Linux it should use sendmail to transmit the voicemails as email whenever a user gets a voicemail. HTH, Ian From: Hector Elias Menjivar [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Getting asterisk to send e-mail to mailbox-users Date: Thu, 29 Sep 2005 07:49:49 -0600 Solo probando -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Arne Morten Johansen Enviado el: Jueves, 29 de Septiembre de 2005 07:50 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Getting asterisk to send e-mail to mailbox-users Ok. I've been searching the wiki and google for a long time now. HOW do I enable asterisk to send mail when users get new messeages in there mailbox? Do i need to change mailcmd in voicemail.conf? Regards, Arne morten ___ --Bandwidth and Colocation sponsored by Easynews.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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.8/114 - Release Date: 28/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.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 _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
Prueba -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Zoa Enviado el: Jueves, 29 de Septiembre de 2005 07:57 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] T.38 Faxing - at astricon ? I didnt implement anything myself and am not very familiar with t.38, but i think its udptl, sip, iax2, and soon gateway too. I will try to get a little more info from the developper when he gets back.. Maybe its even based on your work, i should check. Sorry for the incomplete reply, i just don't know... Cheers, Zoa. Steve Underwood wrote: Zoa wrote: Hey ho! We have a functional t.38 implementation for asterisk, but its far from complete. (meaning it doesnt work for all devices, and i only tested it on 1 fax). I hope to take our t.38 developper with me to Astricon and maybe even demo it there. (Maybe oej could bring a fax or two ? :) I will provide more details when i have them... What have you implemented so far? Terminal mode? Gateway? UDPTL? TPKT? RTP? IAX2? SIP? etc.? Regards, Steve ___ --Bandwidth and Colocation sponsored by Easynews.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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.8/114 - Release Date: 28/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: [Asterisk-biz] Problem with sending fax froma SIP extension
Well, I think what he means is that it's not VoIP, because you are using TDM on both ends. It looks like this: fax machine - TDM - * - TDM - PSTN If you had a SIP ATA attached to a fax machine, you would be using VoIP. That would look like this: fax machine - SIP/VoIP - * - TDM - PSTN I have found using faxes over any VoIP to be *extremely* unreliable. SpanDSP works pretty good, but you won't have much luck using a fax SIP extension. Cheers, -- Tom On 9/29/05, Jonathan k. Creasy [EMAIL PROTECTED] wrote: Why is what he is doing different than having the fax machine on a Sipura ATA? Just because both those ports are on the pci card that doesn't make them not Voice in betweenif I'm wrongeh...oh well -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, September 28, 2005 9:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: [Asterisk-biz] Problem with sending fax froma SIP extension Before responding please read my message again, in my message I worte: 'faxing over VoIP' what you are doing is NOT over VoIP. On 9/28/05, u [EMAIL PROTECTED] wrote: On 9/27/05, C F [EMAIL PROTECTED] wrote: 1. Search the archives 2. Search again 3. Now search the internet 4. The fact is that faxing over VoIP without T.38 doesnt really work. Because it works 60% of the time it doesn't mean that it works, until it works 99% of the time. Since Asterisk does NOT support T.38 it doens't support faxing over VoIP other solutions are available, search the wiki. On 9/26/05, Mark Armstrong [EMAIL PROTECTED] wrote: I am having exactly the same problem. Any information would be appreciated. Faxing is a commercial proposition that my team is working on, any limitations of Asterisk or E1 cards needs to be highlighted. Regards Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, 27 September 2005 9:42 AM To: [EMAIL PROTECTED]; Commercial and Business-Oriented Asterisk Discussion Subject: Re: [Asterisk-biz] Problem with sending fax from a SIP extension why don't you post this to asterisk-users? On 9/22/05, Andy Kuo [EMAIL PROTECTED] wrote: Hi All, I'm having problem sending fax from SIP extensions (Linksys PAP2) through Asterisk Zap channels (ISDN PRI). The SIP extensions can receive fax without problems, but sending fax fails most of the time. Does anyone have this problem? Please advice. Thank you. AK I have a fax connected to a zaptel pci card (FXS). Fax are sent right into an FXO card connected to my PSTN. I don't experience problems but then again I am a light user. Just out of curiosity, is the problem related to sending FAX via IAX or SIP to another asterisk server? if asterisk will support T.38 passthru to say Hylax Fax, does that mean I need a Hylax fax daemon in every asterisk server? ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.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] Getting asterisk to send e-mail to mailbox-users
Arne Morten Johansen wrote: Ok. I've been searching the wiki and google for a long time now. HOW do I enable asterisk to send mail when users get new messeages in there mailbox? Do i need to change mailcmd in voicemail.conf? Make sure sendmail is installed. Doug ___ --Bandwidth and Colocation sponsored by Easynews.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] PRI value
PRI dialplan, in a nutshell, sets dialled digits from your Asterisk box to a pattern that your telco expects. For example, if your telco expects numbers in XXX- format ALWAYS, then you would set it to Local so the MSD of whatever your user dials is stripped off by Asterisk, leaving only a pattern that the PRI expects. 99% of the time, you want to set this to "unknown" and this will allow your Asterisk box to send the digits unaltered to the PRI which is what you want. I see you are in Europe so I can't comment on the numbering pattern your telco expects there, but I suspect that "unknown" will work fine for you here. However, it should be explicitly set and not ignored, if only to "unknown" hth -Original Message-From: Giordano Grandis [mailto:[EMAIL PROTECTED]Sent: Thursday, September 29, 2005 2:11 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] PRI value Hi group, anyone can explain me the exact difference between pri value in zapata.conf ? ; PRI Dialplan: Only RARELY used for PRI.;; unknown: Unknown; private: Private ISDN; local: Local ISDN; national: National ISDN; international: International ISDN If I use it, I also must use prilocaldialplan = local ? Thanks Giordano ___ --Bandwidth and Colocation sponsored by Easynews.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] Remotely dialing calls from a polycom phone
This looks like the info you want: http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config BTW, is your touchpad app publicly available? On 9/29/05, Eric Lawman [EMAIL PROTECTED] wrote: I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is fine, except it can be confusing for the receptionist. Why do I have to answer the phone so I can page someone? What I'm looking for is a little more direct integration between the touchpad and the polycom. So, the question is, does anyone know of a way to initiate a call on a polycom via the remote access port. I was thinking of something similar to the way you can reboot one using a NOTIFY message containing check-sync. I've tried monitoring the sip messages between the phone and asterisk, but so far have only succeeded in making a phone call itself. Any suggestions would be appreciated. Eric. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] Change language to spanish
Hi thanks. Where can i find this variable Hector -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Moises Silva Enviado el: Jueves, 29 de Septiembre de 2005 08:03 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Change language to spanish hi Hector. Just use the * command SetLanguage(), passing as argument es http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SetLanguage for more info check the link, and other related links at the bottom of that page. best regards On 9/29/05, Hector Elias Menjivar [EMAIL PROTECTED] wrote: Hi there: Is there any way to change the language in asterisk to Spanish...I mean I want to change all the dialogs to Spanish in my * box can u help me pls. Hector ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.8/114 - Release Date: 28/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.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] OOH323C
hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
Roy Sigurd Karlsbakk schrieb: ... see http://soft-switch.org/foip.html for a brief explaination of why this generally doesn't work... Hi, maybe one should update this link. I think, you agree, that VoIP is somewhat similar to ISDN, as it transports analog audio data in a digitally coded way. Noone doubts, that ISDN is suitable to transport analog fax. Finally the PSTN is 99,9% digital (ISDN/SS7), even if some subscriber lines are still analog. (Ok, ISDN is a managed network, and thus very high quality.) Since there are more and more regions in the world, where internet connectivity quality approaches to ISDN quality, analog faxing over VoIP becomes reliable and hassle free. You should have 128kbit in both directions, better 256kbit, maybe some QoS build in your router (e.g. Linux's iproute2), and pingtimes below 20ms to the VoIP-provider (PSTN-gateway). DSL with fastpath or internet by TV cable does provide this standard imho and become more and more available. Thus we shouldn't discourage people generally of faxing, even if there are a lot of trouble reports. Who can count the success stories with (analog) fax over IP, which are not posted? As far as I see, there are more users faxing without observing quality differences to ISDN than users with problems with fax over VoIP. This is, what various partners of ours do report after having replaced BRI connections by VoIP in some small and middle sized companies. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.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] Cannot figure out why calls from my Asterisk appear to be from country code +34?
Hello When I dial out from my Asterisk (using Digium analog TDM04B card over pstn line), calls appear to be from +34rest of number I am in UK which is +44 so cannot work out why seeing +34. In my zapata.conf I have: loadzone = uk defaultzone = uk I can't find any country specific stuff in any other conf files. Any ideas how I can correctly set so that calls from my asterisk do not have +34? Angus ___ --Bandwidth and Colocation sponsored by Easynews.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 Echo problems, Urgent, please help,
Are you certain that the echo on sip--sip calls is not being caused by either a spakerphone or extremely loud handset? On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote: I'm using a Voicetronix OpenSwitch12 analouge card. It's on the latest driver for Asterisk (2.4.9) and has echo cancellation turned on. This works fairly well on on SIP-POTS calls after it trains up, but there is still a small echo. The SIP-SIP calls are really echoy though. Cheers, Ian (Sorry, only hit 'reply' last time, not 'reply all'!) From: Tom Hayden [EMAIL PROTECTED] Reply-To: Tom Hayden [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk Echo problems, Urgent, please help, Date: Thu, 29 Sep 2005 09:02:05 -0400 What kind of POTS trunks/cards are you using? -- Tom On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote: Hi all, I hope someone can help, as I have an urgent problem. I've got a production Asterisk server thats been deployed, but we are seeing a strange voice echo problem. There is about a 250ms echo for the users in the office, and they are hearing their own voice back at them. I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of memory. The client SIP phones are Polycom Soundpoint IP600's, WiFi ZyXel 2000w handsets, and X-Lite (free) PC clients. All see the same problem. There is a bridge into the POTS (BT's SystemX) using a Voicetronix OpenSwitch12 card and the vpbhp driver. The echo occurs on both SIP-POTS calls, and SIP-SIP calls. I've tried a number of volume adjustments to correct the echo but it is always the same. If anyone has any ideas I'd really appriciate some help, as this is a major urgency, Many many thanks, Ian Bonham _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
R: [Asterisk-Users] PRI value
Perfect, thanks very much hth. I just set it to unknown, but it doesnt work. Have I to use also prilocaldialplan ? Thanks again Giordano Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Colin Anderson Inviato: giovedì 29 settembre 2005 16.22 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users] PRI value PRI dialplan, in a nutshell, sets dialled digits from your Asterisk box to a pattern that your telco expects. For example, if your telco expects numbers in XXX- format ALWAYS, then you would set it to Local so the MSD of whatever your user dials is stripped off by Asterisk, leaving only a pattern that the PRI expects. 99% of the time, you want to set this to unknown and this will allow your Asterisk box to send the digits unaltered to the PRI which is what you want. I see you are in Europe so I can't comment on the numbering pattern your telco expects there, but I suspect that unknown will work fine for you here. However, it should be explicitly set and not ignored, if only to unknown hth -Original Message- From: Giordano Grandis [mailto:[EMAIL PROTECTED] Sent: Thursday, September 29, 2005 2:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] PRI value Hi group, anyone can explain me the exact difference between pri value in zapata.conf ? ; PRI Dialplan: Only RARELY used for PRI.;; unknown: Unknown; private: Private ISDN; local: Local ISDN; national: National ISDN; international: International ISDN If I use it, I also must use prilocaldialplan = local ? Thanks Giordano ___ --Bandwidth and Colocation sponsored by Easynews.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] H323 and Asterisk
hi guys I was working on asterisk and h323 for the past 2 weeks i have the following feedback please let me know if i am wrong h323 implementation I managed to install this it works, but the problem is it accecpts all calls from all ips. there is no way i can let it accecpt calls only from the IPs i give and bill depending on IP oh323 implementation managed to install, same as h323 implementation i can't add a list of ips and restrict access, the 729 - 723.1 codec convertion does not work well, get a robort voice ooh323c installeed but do not know how to configure :( woomera let me know if there's any one who has tried this. what i want to do it accecpt h323 calls and bill depending on the ip address and send the calls via h323 depdning on the gateway IP i add Regards Kansihka ___ --Bandwidth and Colocation sponsored by Easynews.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] Cannot figure out why calls from my Asteriskappear to be fr
Not sure about the Digium, but I can tell you +34 is Spain, if that helps you track anything down? I assume you've tested the line with a normal phone to make sure it's not a telco fault? Ian From: Angus Comber [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cannot figure out why calls from my Asteriskappear to be from country code +34? Date: Thu, 29 Sep 2005 15:32:39 +0100 Hello When I dial out from my Asterisk (using Digium analog TDM04B card over pstn line), calls appear to be from +34rest of number I am in UK which is +44 so cannot work out why seeing +34. In my zapata.conf I have: loadzone = uk defaultzone = uk I can't find any country specific stuff in any other conf files. Any ideas how I can correctly set so that calls from my asterisk do not have +34? Angus ___ --Bandwidth and Colocation sponsored by Easynews.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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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 Echo problems, Urgent, please help,
Hi Matt, I've tried using both speaker phone and handset and had the volume levels really low and it still occurs. The transmit volumes on the Polycom IP600's I have are a fixed transmit volume however, which is set to TIA/EIA-810-A standard. I have changed the gain settings in the VPB driver, but wonder if there is a gain setting in * it's self I may have missed? Cheers for your help, Ian From: Matt [EMAIL PROTECTED] Reply-To: Matt [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk Echo problems, Urgent, please help, Date: Thu, 29 Sep 2005 10:34:19 -0400 Are you certain that the echo on sip--sip calls is not being caused by either a spakerphone or extremely loud handset? On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote: I'm using a Voicetronix OpenSwitch12 analouge card. It's on the latest driver for Asterisk (2.4.9) and has echo cancellation turned on. This works fairly well on on SIP-POTS calls after it trains up, but there is still a small echo. The SIP-SIP calls are really echoy though. Cheers, Ian (Sorry, only hit 'reply' last time, not 'reply all'!) From: Tom Hayden [EMAIL PROTECTED] Reply-To: Tom Hayden [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk Echo problems, Urgent, please help, Date: Thu, 29 Sep 2005 09:02:05 -0400 What kind of POTS trunks/cards are you using? -- Tom On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote: Hi all, I hope someone can help, as I have an urgent problem. I've got a production Asterisk server thats been deployed, but we are seeing a strange voice echo problem. There is about a 250ms echo for the users in the office, and they are hearing their own voice back at them. I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of memory. The client SIP phones are Polycom Soundpoint IP600's, WiFi ZyXel 2000w handsets, and X-Lite (free) PC clients. All see the same problem. There is a bridge into the POTS (BT's SystemX) using a Voicetronix OpenSwitch12 card and the vpbhp driver. The echo occurs on both SIP-POTS calls, and SIP-SIP calls. I've tried a number of volume adjustments to correct the echo but it is always the same. If anyone has any ideas I'd really appriciate some help, as this is a major urgency, Many many thanks, Ian Bonham _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
Roger Schreiter wrote: Roy Sigurd Karlsbakk schrieb: ... see http://soft-switch.org/foip.html for a brief explaination of why this generally doesn't work... Hi, maybe one should update this link. Update it in what way? I think, you agree, that VoIP is somewhat similar to ISDN, as it transports analog audio data in a digitally coded way. VoIP is not every remotely in the same ballpark as ISDN, which makes the rest of what you say below meaningless. ISDN has rock solid timing, and (essentially) guaranteed delivery. It is no functionally no different from an analogue line. In fact, every analogue line feeds into an ISDN type connection between exchanges (ISDN, SS7, etc. are functionally equivalent). Noone doubts, that ISDN is suitable to transport analog fax. Finally the PSTN is 99,9% digital (ISDN/SS7), even if some subscriber lines are still analog. (Ok, ISDN is a managed network, and thus very high quality.) Since there are more and more regions in the world, where internet connectivity quality approaches to ISDN quality, analog faxing over VoIP becomes reliable and hassle free. You should have 128kbit in both directions, better 256kbit, maybe some QoS build in your router (e.g. Linux's iproute2), and pingtimes below 20ms to the VoIP-provider (PSTN-gateway). DSL with fastpath or internet by TV cable does provide this standard imho and become more and more available. Thus we shouldn't discourage people generally of faxing, even if there are a lot of trouble reports. Who can count the success stories with (analog) fax over IP, which are not posted? As far as I see, there are more users faxing without observing quality differences to ISDN than users with problems with fax over VoIP. This is, what various partners of ours do report after having replaced BRI connections by VoIP in some small and middle sized companies. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Audio Files, Filtering, and Formats for Asterisk
Sherwood McGowan wrote: I listened to all the demos you showed. My ear discerns a little muffling and minor slushiness in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to de-ess the recording. Or, I could just be rather tired. I don't think so. Everybody else I've played them for has made similar observations. Either way, your best bet is to have the system use WAV files but also make available GSM and ulaw versions of the same files. This is so that the system can pick (and it does this automatically) the best format that requires the least amount of CPU power for transcoding. If I remember correctly, FXO/FXS cards can use straight PCM files. What I do to prepare a file is this: I record in 48Khz 32 bit mode (32 bit is just higher resolution so processing has more to work with, 16 bit is fine since it's the end result), then I compress with de-essing and pop removal (maximizes volume, removes slushy or whistling s's and popping p's), then run through an FFT (Fast Fourier Transform) to bring the frequency response within the natural range of a telephone (if I remember correctly it's within the 300 - 4K range, I can check if need be). How do you de-ess and remove pops? What filters do you use? Then I reduce the file to a 8Khz, 16 bit mono file. Then I check the audio again, and normalize (like compression but only raises the whole file to where the highest peak of audio reaches the level requested, instead of raising or lowering the level on a dynamic basis by using readahead of a couple milliseconds). I usually normalize to around -3db. The end result is a WAV file that sounds good over the phone. I then put the file on the asterisk server (or another server with sox installed) and convert to gsm, ulaw, and alaw (using the original WAV, not using converted gsm or whatever). Hope this was helpful, and I wish you luck. If nothing else, for like $50 or so (depending on how many files there are) I would be willing to take a series of WAV files from you and perform the filtering and whatnot for you, and supply gsm, wav, ulaw, and alaw. I'm also going to just try and get a series of recordings together through my partner's studio in Phoenix AZ, make the files available to the community, and take up donations for the studio and processing time. Talk to you soon. I'm going to cc the asterisk-users list for this, so that the community can benefit from the information. Thank you, Sherwood - you've been extremely helpful! I am going to try those steps you suggested (once I learn how to 'de-ess' and remove pops) but I may just take you up on the filtering and processing offer. -Stephen- ___ --Bandwidth and Colocation sponsored by Easynews.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] soft phones for Zaurus PDA
Ziaxphone might fit your needs. http://www.kauss.org/Stephan/ziaxphone/ Haven't used it recently since someone broke the screen on my Zaurus =( -- William ___ --Bandwidth and Colocation sponsored by Easynews.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] Tiny Echo on PSTN via Zaptel
On Thu, Sep 29, 2005 at 08:33:03AM -0400, Tony Nichols wrote: I have had problems between the sip/FXO lies and was able to kill the echo by trying different combinations of the echocancel line to 64 (I think it has settings in 32 bit increments) Just kept trying different ones till it went away. Here is my config: I think I finally found the problem (but I'm not sure). The rxgain on my phone ports was set to -8. When I turned them back up to 0.0 (to boost the outgoing signal on the POTS lines when a call was going on), the echo canceller seems to work pretty well. I've been using ztmonitor to verify this: previously the echo was 20-40% of the original signal. Now it's hardly ever there. I thought I had my gains set correctly, but it seems like louder is better (except for distortingly loud) for the echo canceller. -- Shaw Terwilliger [EMAIL PROTECTED] SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.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] OOH323C
Am Donnerstag 29 September 2005 16:28, Kanishka Somaratne schrieb: hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk Hello, for me it seems that the OOH323 Development is not finished yet. I use the OH323 that you can reach at http://www.inaccessnetworks.com/projects/asterisk-oh323 Yours Hans-Peter Straub -- ---* I-NetPartner GmbH Hans-Peter Straub Seewiesenstrasse 12 D-73054 Eislingen -- Phone: +49 7161 9849955 Fax: +49 7161 9849956 -- eMail: [EMAIL PROTECTED] Web: http://www.I-NetPartner.de ---* ** Informieren Sie Sich über ** -- GigaLan -- ** das Funknetz im Filstal ** http://www.GigaLan.de ---* -- PGP-ID: 24557EED PGP-Key: http://www.i-netpartner.de/hps.asc PGP-Fingerprint: 51F2 31E4 4361 1B7F 8648 60D9 FC1A 68D2 2455 7EED -- ___ --Bandwidth and Colocation sponsored by Easynews.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: R: [Asterisk-Users] PRI value
Have I to use also prilocaldialplan ? Can be left unknown. Explains what you expect as the incoming number to look like Thanks again Giordano ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown:Unknown don't expect anything ; private:Private ISDN special isdn numbering not following E164 Standard ; local: Local ISDN your local telephone number should be part of the number ; national: National ISDN your area code should also be part of the number ; international: International ISDN GIVE US ALL YOU HAVE: Expect an additional international prefix for numbers If I use it, I also must use prilocaldialplan = local ? this is what you expect the other option tells your telco what numbering scheme you send them, so it knows the semantics of the number Jens ___ --Bandwidth and Colocation sponsored by Easynews.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] call center software and asterisk
Hi guys, Need some advise. Is there some kind of call center software which can interconnect with asterisk? So, for example, agents can see on their pc's all info about calling client (based on clid) before they pick up the phone. And that outbound calls are also automated. Commercial solutions more then welcome. Thx, Bartosz Jozwiak ___ --Bandwidth and Colocation sponsored by Easynews.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] PRI value
The values are mutually exclusive so you can only set it once. What you want to do is from the Asterisk console type in PRI DEBUG SPAN 1 (if you only have 1 PRI) and place a call. PRI DEBUG will throw up everything on the screen concerning call setup and teardown at the PRI network layer. A good debug looks like this: -- Making new call for cr 32773 Protocol Discriminator: Q.931 (8) len=54 Call Ref: len= 2 (reference 5/0x5) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 0f b1 43 6f 6c 69 6e 20 41 6e 64 65 72 73 6f 6e] Display (len=15) Charset: 31 [ Colin Anderson ] --important part below-- [6c 06 21 81 31 30 30 31] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '1001' ] [70 08 80 37 30 32 33 30 31 37] Called Number (len=10) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '7023017' ] -- Called g0/7023017 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32773/0x8005) (Terminator) Message type: CALL PROCEEDING (2) --import part above --- The sections in the "important part" are going to give you clues as to what's going on under the hood. If you can't get to this part above with a PRI debug, for example, you have a misonconfiguration in your * setup. Post your debug and let's look at it. -Original Message-From: Giordano Grandis [mailto:[EMAIL PROTECTED]Sent: Thursday, September 29, 2005 8:45 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: R: [Asterisk-Users] PRI value Perfect, thanks very much hth. I just set it to unknown, but it doesn't work. Have I to use also prilocaldialplan ? Thanks again Giordano Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Colin AndersonInviato: giovedì 29 settembre 2005 16.22A: 'Asterisk Users Mailing List - Non-Commercial Discussion'Oggetto: RE: [Asterisk-Users] PRI value PRI dialplan, in a nutshell, sets dialled digits from your Asterisk box to a pattern that your telco expects. For example, if your telco expects numbers in XXX- format ALWAYS, then you would set it to Local so the MSD of whatever your user dials is stripped off by Asterisk, leaving only a pattern that the PRI expects. 99% of the time, you want to set this to "unknown" and this will allow your Asterisk box to send the digits unaltered to the PRI which is what you want. I see you are in Europe so I can't comment on the numbering pattern your telco expects there, but I suspect that "unknown" will work fine for you here. However, it should be explicitly set and not ignored, if only to "unknown" hth -Original Message-From: Giordano Grandis [mailto:[EMAIL PROTECTED]Sent: Thursday, September 29, 2005 2:11 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] PRI value Hi group, anyone can explain me the exact difference between pri value in zapata.conf ? ; PRI Dialplan: Only RARELY used for PRI.;; unknown: Unknown; private: Private ISDN; local: Local ISDN; national: National ISDN; international: International ISDN If I use it, I also must use prilocaldialplan = local ? Thanks Giordano ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
Roger Schreiter wrote: I think, you agree, that VoIP is somewhat similar to ISDN, as it transports analog audio data in a digitally coded way. Noone doubts, that ISDN is suitable to transport analog fax. Finally the PSTN is 99,9% digital (ISDN/SS7), even if some subscriber lines are still analog. (Ok, ISDN is a managed network, and thus very high quality.) The fact that ISDN and VoIP both use digitial audio is irrelevant. The reason why your average VoIP connection cannot be used for faxing is not because of the digitized nature of the audio, but rather the medium over which it is being transported. The PSTN virtually guarantees timely and orderly delivery of every single audio sample. An IP network cannot do this, and so as the amount of jitter on the IP network varies its usability for fax will also vary. Modulated data cannot tolerate receiver-end misordering of audio samples, synthesizing of missing audio samples, or dropping audio samples. If any of this happens then data corruption will certainly occur. And, depending on the frequency and timing of that data corruption, and the resiliancy of the fax endpoints, that data corruption can (and is well-known to) cause fax failures. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.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] Correction: Asterisk sound files, audio bandwidth, and sound quality
On Wed, 28 Sep 2005, Stephen Bosch wrote: [EMAIL PROTECTED] wrote: When I listen to the GSM compressed prompts, I can hear subtle noise when the person is speaking -- this is irrespective of whether I listen to the prompts through the TDM-400 on an analogue phone or whether I do so directly on a workstation. It has to be possible to do better than that. It doesn't have to be CD quality sound, but it should be clear. How can the sound quality be improved within the limitations of an 8 kHz sample rate? I presume you know the answer: augment the .gsm files with 8k sample rate .wave files? If I knew the answer, I wouldn't be asking. There are other ways to improve sound quality, like changing the resolution. I'm sorry, I wasn't trying to be funny. The recorded prompts supplied with Asterisk are encoded with the .gsm codec. That makes them sound like audio sounds on your GSM cellphone. Which is noticably worse than true PCM audio. Now in the telephone world best quality still isn't very good - its ulaw or alaw encoded 8kHz audio. That's frequency response up to 3.5kHz and about 12 or 13 bits of dynamic range. But the fuzzyness you hear on the standard Asterisk prompts is due, I'm sure, to the use of gsm compression. Its not necessary that they are stored in gsm - Asterisk has got an elaborate logic for finding the best version of the prompt, so you can store the same prompt in lots of formats and Asterisk will choose the best. You can see the formats with show file formats So you need them stored in a non-compressed format - the simplest is probably wav format, another possibility is to use sox to put them into ulaw directly Now Digium hasn't made the standard prompts available in a format other than gsm. I don't know why. For us we recorded the prompts in South African voice and so we have those. You need to either extract the original non-compressed prompts from Digium (if they have them), or take it as an opportunity and record your own set in Canadian accent. Hope that helps. Steve ___ --Bandwidth and Colocation sponsored by Easynews.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] call center software and asterisk
AstGUIClient and VICIDIAL seem to be a good tool for the task. I cannot verify, as I have not used them before. http://astguiclient.sourceforge.net/ Nathan Bartosz Jozwiak wrote: Hi guys, Need some advise. Is there some kind of call center software which can interconnect with asterisk? So, for example, agents can see on their pc's all info about calling client (based on clid) before they pick up the phone. And that outbound calls are also automated. Commercial solutions more then welcome. Thx, Bartosz Jozwiak _ -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.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] Variable in call parking
Hi Andrew, Not sure if I understand your question, but this may help - * has the following settings in features.conf that are related to parking: parkext = ;the extension that users xfer calls to in order to park them parkpos = - ;the extension range that * will use to park calls. It will tell the user which extension it has parked each call on when they are parked. Dialling this extension retrieves the parkpos. context = parkedcalls ;context that calls are parked in parkingtime = xxx ;number of seconds a call will be parked for before being sent back to the extension that parked it. Hope this helps, Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Andrew Nowrot wrote: Hi, Can anyone tell me if Asterisk sets some variable when doing a call parking (when someone presses an exten set in features.conf). In can't find this information on a wiki. Cheers ;) Andrew ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] call center software and asterisk
www.inconcertCC.com has a solution based on Asterisk. regards, srsergio -Mensaje original- De: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Enviado el: jueves, 29 de septiembre de 2005 17:17 Para: Asterisk Users Mailing List - Non-Commercial Discussion CC: Commercial and Business-Oriented Asterisk Discussion Asunto: [Asterisk-Users] call center software and asterisk Hi guys, Need some advise. Is there some kind of call center software which can interconnect with asterisk? So, for example, agents can see on their pc's all info about calling client (based on clid) before they pick up the phone. And that outbound calls are also automated. Commercial solutions more then welcome. Thx, Bartosz Jozwiak ___ --Bandwidth and Colocation sponsored by Easynews.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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.8/114 - Release Date: 28/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.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] T.38 Faxing - at astricon ?
The problem as I see it is that if people start expecting it to work then rather than being pleasantly surprised when it does, they will be bitterly disappointed when it doesn't. IMHO analog fax over IP is too flaky to encourage the general public to utilise, and any suggestion to the contrary is misleading. Having said that, I have an analog fax connected to an ATA that works 100% of the time, however I have my ATA and Asterisk on the same ethernet switch. I wouldn't expect to have it work reliably over a WAN or other broadband internet connection. Craig - Original Message - From: Roger Schreiter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, September 29, 2005 10:31 PM Subject: Re: [Asterisk-Users] T.38 Faxing - at astricon ? Roy Sigurd Karlsbakk schrieb: ... see http://soft-switch.org/foip.html for a brief explaination of why this generally doesn't work... Hi, maybe one should update this link. I think, you agree, that VoIP is somewhat similar to ISDN, as it transports analog audio data in a digitally coded way. Noone doubts, that ISDN is suitable to transport analog fax. Finally the PSTN is 99,9% digital (ISDN/SS7), even if some subscriber lines are still analog. (Ok, ISDN is a managed network, and thus very high quality.) Since there are more and more regions in the world, where internet connectivity quality approaches to ISDN quality, analog faxing over VoIP becomes reliable and hassle free. You should have 128kbit in both directions, better 256kbit, maybe some QoS build in your router (e.g. Linux's iproute2), and pingtimes below 20ms to the VoIP-provider (PSTN-gateway). DSL with fastpath or internet by TV cable does provide this standard imho and become more and more available. Thus we shouldn't discourage people generally of faxing, even if there are a lot of trouble reports. Who can count the success stories with (analog) fax over IP, which are not posted? As far as I see, there are more users faxing without observing quality differences to ISDN than users with problems with fax over VoIP. This is, what various partners of ours do report after having replaced BRI connections by VoIP in some small and middle sized companies. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] zttest - 100% ?
Have you checked that the TDM400P isn't sharing an IRQ with anything else? Don't trust /proc/interrupts - run lspci -v to confirm this. We have * running on an x206 and found that the only way to stop the TDP400P sharing an IRQ with other devices was to juggle cards between slots. Hope this helps! Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Marco Supino wrote: Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Remotely dialing calls from a polycom phone
Message: 7 Date: Thu, 29 Sep 2005 09:53:27 -0400 From: Eric Lawman [EMAIL PROTECTED] Subject: [Asterisk-Users] Remotely dialing calls from a polycom phone To: Asterisk-Users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is fine, except it can be confusing for the receptionist. Why do I have to answer the phone so I can page someone? What I'm looking for is a little more direct integration between the touchpad and the polycom. So, the question is, does anyone know of a way to initiate a call on a polycom via the remote access port. I was thinking of something similar to the way you can reboot one using a NOTIFY message containing check-sync. I've tried monitoring the sip messages between the phone and asterisk, but so far have only succeeded in making a phone call itself. Any suggestions would be appreciated. Eric. First thing that comes to mind here is using auto-answer. While I don't have an IP600 to play with (donations accepted), the way I would envision this is to configure an auto-answer extension on hte Polycom, and have the touchpad action call the auto-answer extension using a .call file to Asterisk. Touch the pad, it drops the .call in, which goes to a meetme with all the necessary extensions in the paging group added. Touch again, drop the call (via the manager API would be my approach). All untested- but fairly straightfoward. If you want to email me directly, I'd be happy to help further. -Paul Davidson PlanCommunications, LLC ___ --Bandwidth and Colocation sponsored by Easynews.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] minor(? ) Grandstream phone issue
I hate to bother the list with this potentially minor issue but I just wonder if it's a symtom of some other problem. Every time I make a call the BT-102, with the latest firmware, she just keeps the LED display lit and the timer counting after hangup. I check the CLI and the hangup is being executed, I certainly was concerned it might be keeping the line open but that doesn't seem to be the case. I thought that she should go back to the date/time display but that doesn't happen. I don't have silence suppresion on and I've gone over the other configs compared with what's on voip-info many time. One other thing, there isn't a dial tone when I pick it up. It seems to 'work'; both inbound and outbound. It's connected to * 1.09 which is registered @broadvoice. Thanks for any insights bestowed on this noob :) Bob ___ --Bandwidth and Colocation sponsored by Easynews.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] soft phones for Zaurus PDA
On 9/29/05, William Suffill [EMAIL PROTECTED] wrote: Ziaxphone might fit your needs. http://www.kauss.org/Stephan/ziaxphone/ Haven't used it recently since someone broke the screen on my Zaurus =( I can vouch for the software. I haven't used it in some time, but it DID work when I tried it on my 5500. -- Leif Madsen - http://www.leifmadsen.com Astricon 2005, Anaheim, CA, October 12-14 http://www.astricon.net http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.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] zttest - 100% ?
Hi, My TDM is on its own IRQ, and the x306 has only one full-size PCI slot.. so no playing with it, what results do you get from zttest ? what IRQ is the card on ? Marco. Damian Funnell wrote: Have you checked that the TDM400P isn't sharing an IRQ with anything else? Don't trust /proc/interrupts - run lspci -v to confirm this. We have * running on an x206 and found that the only way to stop the TDP400P sharing an IRQ with other devices was to juggle cards between slots. Hope this helps! Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Marco Supino wrote: Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] TE205P in loopback?
All the config match. Just to make sure, how did you make your loopback cable? Which pins are conected were? Thanks in advance, -f From: "Steve Totaro" [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo: "Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] TE205P in loopback?Date: Wed, 28 Sep 2005 22:28:02 -0700 It flashes red until the properly made cables are connected and then should turn to solid green. Again, I suspect your cableis the issue. Are all the drivers loaded? ztcfg? Asterisk running? One span set to pri_cpe and the other to pri_net? - Original Message - From: Franciraldo Cavalcante Junior To: asterisk-users@lists.digium.com Sent: Wednesday, September 28, 2005 6:47 PM Subject: Re: [Asterisk-Users] TE205P in loopback? The board flashes red without cables and continues flashing red after cables are connected. Should it flash green before? I don't see anything weird in dmegs results. From: "Steve Totaro" [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo: "Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] TE205P in loopback?Date: Wed, 28 Sep 2005 21:12:08 -0700 If the cables are connected and the lights are still flashing red then I suspect that the cable is the issue. - Original Message - From: Franciraldo Cavalcante Junior To: asterisk-users@lists.digium.com Sent: Wednesday, September 28, 2005 5:44 PM Subject: Re: [Asterisk-Users] TE205P in loopback? I did the cable and configurationas recomended, but I may have another problem. After I run: modprobe zaptel modprobe wct2xxp I don't see the boards flashing green it's flashing red and in sync, as I implicitly understand from your message. Should it flash green and when correctly cabled and configured, keep green? How can I debug this? any help is appreciated. -f From: "Steve Totaro" [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo: "Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] TE205P in loopback?Date: Wed, 28 Sep 2005 20:12:40 -0700 I just did this on a te410p. make a t1 crossover, cross pins one with four, and two with five. Setone span for pri_cpe and the other sidefor pri_net.boom, you are done and can start testing. the lights should turnsolid green and you shouldfirst see theD channels come up and then all theB channels, if this happens, you have done it correctly. thanks, Steve - Original Message - From: Franciraldo Cavalcante Junior To: asterisk-users@lists.digium.com Sent: Wednesday, September 28, 2005 4:46 PM Subject: [Asterisk-Users] TE205P in loopback? I'm a new * user and had just intalled A TE205P line in my Linux machine (running linux kernel 2.6.10). I would like to test the T1 interface but I don't have an ISDN PRI interface available right now. Is there a way to make some sort of loopback test? How to configure it? Do I need the cables to have tx/rx exchanged? Any help would be appreciated. - f ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.11.8/113 - Release Date: 9/27/05 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.11.8/113 - Release Date: 9/27/05 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing
Re: [Asterisk-Users] zttest - 100% ?
This might seem a silly question but, what is the true meaning of the numbers zttest spits out?On 9/29/05, Marco Supino [EMAIL PROTECTED] wrote:Hi,My TDM is on its own IRQ, and the x306 has only one full-size PCI slot.. so no playing with it,what results do you get from zttest ? what IRQ is the card on ?Marco.Damian Funnell wrote: Have you checked that the TDM400P isn't sharing an IRQ with anything else?Don't trust /proc/interrupts - run lspci -v to confirm this. We have * running on an x206 and found that the only way to stop the TDP400P sharing an IRQ with other devices was to juggle cards between slots. Hope this helps! Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Marco Supino wrote: Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- We hold [...] that all men are created equal; that they areendowed [...] with certain inalienable rights; that amongthese are life, liberty, and the pursuit of happiness -- Thomas Jefferson ___ --Bandwidth and Colocation sponsored by Easynews.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
