[Asterisk-Users] Voicemail With No Messages?
Is there anyway to NOT allow the incoming caller to leave a voicemail message for a certain mailbox? I would like the caller to hear the message and then have the option to press 1(for example) to call the user (make an outgoing call), but not to be able to leave the message. Even if after the unavail message is played the caller gets kicked back to another menu that has the option of calling the user's external number. Thanks, Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail With No Messages?
You can use the unavail.gsm sound file for more than voicemail. Why not have a Background() statement in its own context with the dtmf options you want from there: [leavenomessage] exten = s,1,Background(unavail.gsm) exten = 1,1,Dial(LA/la/land) exten = *,1,Goto(the-main-menu) Man is that a rough-cut!! ..you get the idea. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Michael Stearne |Sent: Friday, May 20, 2005 11:42 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion; Jim Ginn |Subject: [Asterisk-Users] Voicemail With No Messages? | |Is there anyway to NOT allow the incoming caller to leave a voicemail |message for a certain mailbox? I would like the caller to hear the |message and then have the option to press 1(for example) to call the |user (make an outgoing call), but not to be able to leave the message. | |Even if after the unavail message is played the caller gets kicked |back to another menu that has the option of calling the user's |external number. | |Thanks, |Michael |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN-voip/sip echo
I'm still relatively a novice with asterisk and am having issues with echo. The calling party that calls a PSTN number doesnt hear the echo, but the answered side via sip or forwarded to another PSTN number over voip hears excessive echo that makes it difficult to communicate. I've been playing with the zapata.conf settings for echocancel, echotraining, rxgain, txgain, etc and am basically stabbing in the dark (grin) I've read the wiki about it, but it doesn't go into very much detail. Anyone know which parameters fix this issue? Is there an easier way than tweaking settings in zapata.conf, monitoring with ztmonitor, and restarting asterisk over and over? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.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] acd with mysql or ast_data support
Hi, I am using ACD, i.e. application Queue(). Is there a way to use mysql for the configuration file? Thanks, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Voicemail With No Messages?
On 5/21/05, Chris Coulthurst [EMAIL PROTECTED] wrote: You can use the unavail.gsm sound file for more than voicemail. Why not have a Background() statement in its own context with the dtmf options you want from there: [leavenomessage] exten = s,1,Background(unavail.gsm) exten = 1,1,Dial(LA/la/land) exten = *,1,Goto(the-main-menu) Thanks! I was thinking in terms of that. Basically I can just get the user to record their message and then play whatever is the unavail message for that mailbox. Michael |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Michael Stearne |Sent: Friday, May 20, 2005 11:42 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion; Jim Ginn |Subject: [Asterisk-Users] Voicemail With No Messages? | |Is there anyway to NOT allow the incoming caller to leave a voicemail |message for a certain mailbox? I would like the caller to hear the |message and then have the option to press 1(for example) to call the |user (make an outgoing call), but not to be able to leave the message. | |Even if after the unavail message is played the caller gets kicked |back to another menu that has the option of calling the user's |external number. | |Thanks, |Michael |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoipSupply.com
Wilson Pickett wrote: Just a quick note, if you typically ship to a different address than your credit card billing address, you can file that address with your credit card company. Most cards allow you to have mulitple addresses on file so that your Address Verfication goes through correctly. Not universally true, I'm afraid. Amex France won't do it and Wells Fargo (calif) won't do it. OTH, Paypal will ship an order billed to Wisconsin to Nigeria with a blick. Nor is that totally correct. The *seller* determines the degree of where to shipness for Paypal. There are various degrees of paranoia possible on the seller's part. Some will, in fact, only ship to an address that is registered official for the credit card. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Set CallerID in zapata.conf with QuadBri or other solution with parallel call signalling
Hi,
we are using
asterisk with Junghanns QuadBri and some sip phones. 2 channels are configured
in NT mode (ISDN PBX connected, internal ) and 2 channels are connected to the
public ISDN network (bri-cpe). We use Bristuff 0.2.0 RC8C from
Junghanns.
When a call comes
in from the public phone for a specific extension (Hotline Number), we initiate
a parallelcall to some SIP phones and also to our PBX through the quadbri and we
also do a signalling on one mobile phone(through the second channelwhich
is connected to the oublic phone network).
Problem:
The sip phones and
the internal phones on the PBX see the number of the calling party correctly
(e.g. 040-987654321). Cause we can´t set a callerid to the public phone network
(to show the calling party number), we want to show an extension of our numbers
on our isdn-bri (asigned by Carrier, e.g. our numbers are 12345-0 to 12345-99).
If we use our current configuration, everythings works good, execpt the isdn
call to the mobile phone. As calling id it shows 12345 w/o an extension. We
would like to set a specific extension, so that for the call to the mobile phone
it is displayed 12345-88 (so we see that is a call forwarded from asterisk for a
specific extension) and the rest (SIP an PBX) should display the ID of the
calling party (e.g. 040-987654321). Does anybody has an idea ? I tried to set a
callerid in the zapata.conf for the channels dialing the mobile phone with
callerid= "" 1234588and also same w/
setcallerid(""1234588). Result is that it´s shown 12345 to the mobile
phone.
thanks for
help
Andreas
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[Asterisk-Users] ISDN data connection through Asterisk
Hi, Is there a simply way to allow dialout from ISDN modem to outside number through Asterisk? I've got an server with an Asterisk and the following cards: 1. TE110 -- to telco 2. TE400P with one FXS to analog phone 3. Two HFC-S based cards in NT mode I'd like to connect ISDN modem to one HFC-S card and allow dialout through TE100 to some external number (being more precisely, to some dumb bank which got only homebanking and no Internet banking). I've searched the list archive but found no clues. Thanks in advance for your help, -- Marcin ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ISDN data connection through Asterisk
Hello, On Sat, 21 May 2005, Marcin wrote: Hi, Is there a simply way to allow dialout from ISDN modem to outside number through Asterisk? I've got an server with an Asterisk and the following cards: 1. TE110 -- to telco 2. TE400P with one FXS to analog phone 3. Two HFC-S based cards in NT mode We've done this several times with Junghanns Cards - nearly no problem, just the normal dialplan entries. [data-isdn] exten = 246,1,Wait exten = 246,2,zapEC(off) exten = 246,3,Dial(Zap/48/5552,20) exten = 246,4,Macro(hangupcall) For outbound calls just use your normal dialout-settings. I'd like to connect ISDN modem to one HFC-S card and allow dialout through TE100 to some external number (being more precisely, to some dumb bank which got only homebanking and no Internet banking). I've searched the list archive but found no clues. The only mentionable thing is, that we had to take away all settings for txgain and rxgain in zapata.conf for the affected channels. If we had something like rxgain=1.0 in zapata.conf for the PRI-card the ISDN-Connection was established through asterisk (briging of e.g. Zap/1-15 and Zap/48) but the LCP-Negotiaton of the PPP failed. Perhaps someone has a clue on this. Tested version was Asterisk 1.0.5 with bristuff patches. This affected inbound an outbound calls Regards Torsten Thanks in advance for your help, -- Media Online Internet Services Marketing GmbH Torsten Krueger [EMAIL PROTECTED] fon: 49-231-5575100fax: 49-231-55751098 Kurze Str. 10 D-44137 Dortmund ___ Asterisk-Users mailing list Asterisk-Users@lists.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 7960s and skinny
Hi Andy, I have been trying to get 7910's work with *. I have tried with both skinny and chan_sccp. Could you please instruct me about the configuration? I have found some detalis about 7920 with sccp in voip-info.org. But I haven't find any document for 7910. Please help me to get them work properly. I have four 7960's and working fine with SIP.But I am in trouble with those (total of 8) 7910 IP phones. On 5/2/05, Andy Hamilton [EMAIL PROTECTED] wrote: Anton: Yes, the whole legally licensing the phones from Cisco is a major drawback. I just compiled and install the chan_sccp-mayday CVS snapshot. It works like a charm (thanks Julien!). In fact, I did some dialing and redialing to my 7960 (SIP) from my 7910 and the 7960 ended up needing to be rebooted. I was quite pleasantly surprised; the 7910/chan_sccp seemed to be having a field day. Here is what I have found for the 7910: - Hold button works (a little weird when more than one call) - Line button brings up the line - Transfer button doesn't seem to work - Msgs button doesn't seem to work - Conf button doesn't seem to to work - Forward button doesn't seem to work - Speed1 doesn't seem to work - Speed2 doesn't seem to work - Redial works Hope this helps. -Andy On 5/1/05, Julien Goodwin [EMAIL PROTECTED] wrote: On Sat, Apr 30, 2005 at 01:00:18PM -0500, Andy Hamilton arranged a set of bits into the following: I'll be able to get back to you Sunday night about specifics; the phone is not where I am right now. Using chan_sccp, (I think November 2004 or so CVS Head) I know I can receive calls, place calls, etc. It is a rather low volume phone, so I don't know off hand about specific keys; I'll check those later. Generally if the phone supports the function, and support is in chan_sccp for that function it will work for all phones. Additionally, I have not yet tried a new copy from CVS. Occasionally, I think the chan_sccp driver blips out in Asterisk (it may be the phone; I've had it apart several times because the on/off hook switch membrane is a little sketchy). I have dealt with this by That's one of the big things that causes problems, both with chan_sccp and the phones themselves, both get a little confused. However several other crash issues have been recently fixed, so running CVS_HEAD is advised. restarting Asterisk. The only other thing I can say right now about the 7910 is that it and my Cisco FastHub don't get along. At all. I have the 7910 plugged into my 7960. That's odd, the only time I've ever had ethernet incompatabilities was with a very cheap switch. Overall, I would say that if you have a non-critical system and would like to use a 7910, chan_sccp should be able to handle it fine. However, if you budget permits, the 7960 and 7940 phones are quite nice (use SIP with those -- it's far more reliable. I must say, though, that my 7960 has frozen/crashed a handful of time when running the SIP image. That was the phone itself, Asterisk was fine.) I have yet to purchase a 7905 or 7912, but I've played around with some 7912's on a CCM system -- they seem quite nice and I think they take SIP. Yep, they do. (Don't know about the 7902, but really can't see why anyone would buy one) The 7920 is also nice because it's wireless. However, I don't think Cisco has anything but a Skinny image for it [yet]. No they don't, and forget the yet, if a phone isn't announced with SIP support it probably never will have it (witness: 7935/6, 7970) I would stick with SIP wherever you can. And I agree Thanks, Julien chan_sccp project lead ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help Understanding ISDN Channels
Hi: The phone company here use Europe telephone system. They offere ISDN lines with multiple channels, starting with two and they add channels on demand. Am I write to understand ISDN of more like E1 or T1 line. If not, what's the difference. __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.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-Hylafax
Hi all, I try to setup Asterisk TDM400P (1fxs/1fxo) and Hylafax as gateway to PSTN: PSTNASTERISK+TDM400P-modem---HYLAFAX | SIP From modem I can dial number to fxs card however when Hylafax server send fax to modem Asterisk failed !? What's wrong ? Regards Harry _ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com extensions.conf Description: 3949034846-extensions.conf zapata.conf Description: 664965922-zapata.conf zaptel.conf Description: 2142627058-zaptel.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.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] LiveVoip setup
We have applied for LiveVoip termination. Although the account is now available via Web, I have not got any information how to set it up. Can anybody guide me to make my paid advanture a success? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Help Understanding ISDN Channels
The two-channel ISDN is called Basic Rate ISDN, and it's a smaller cheaper version of the Primary Rate ISDN (PRI). With a BRI, you have two B-Channel (each 64kbps) and one D-Channel (16kbps). It's very common in Germany, Spain and other european countries, as reliable low-cost low-bandwidth data solution or as a replacement for two POTS lines with much better voice quality and added services. You can even get a DSL over a BRI line... I didn't know you could add more channels to a BRI... maybe what they offered you is just adding more BRI ISDNs, and not adding more channels to your BRI. Or maybe they're offering you a PRI with only two channels active and they activate it on-demand... - Original Message - From: chawki hammoud [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Saturday, May 21, 2005 12:14 PM Subject: [Asterisk-Users] Help Understanding ISDN Channels Hi: The phone company here use Europe telephone system. They offere ISDN lines with multiple channels, starting with two and they add channels on demand. Am I write to understand ISDN of more like E1 or T1 line. If not, what's the difference. __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPSwitchBoard now supports CAPI
Version 0.117 - 21. may 2005 * CAPI support has now been added to IPS * Save all you speed dial number in the asterisk server and retrieve them from any other intance of IPS. This way you can easily share all your speed dial numbers between all users of IPS. * All speed dial number can be saved with the family cidname - use this to set Caller ID on incoming call in your dial plan. * Many bug fixes FREE download from: http://ipswitchboard.thorben.dk ___ IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPS is an Windows.NET application which gives you: * Unattended/attended transfers. * Park calls and retrieve/forward them again. * Organize all your SIP, IAX, CAPI and Zap extensions (automatically retrieved from Asterisk). * Hotel/Call shop Billing module * Monitor all extensions, queues, agents and Parked Calls. * Dynamically log extensions in and out of queues. * Integration with CRM software on the web. * Browse Call Records and make Charts. * Record calls and transfer wav files to the PC automatically. * Set Do Not Disturb on Extensions and give a reason. * Speed Dialing. Speed dial number can be shared from the server. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] VoipSupply.com
The *seller* determines the degree of where to shipness for Paypal. There are various degrees of paranoia possible on the seller's part. Call you and raise you one! Yup, in the end, the seller decides, period. The decision is based on whether they think the people doing the transacation (PP, bank, Amex) will pay or object if there is later complaint. I ordered some stuff for about $1500 a while back (not voip or voipsupply, but a similar type company selling musical equipment). When I spoke to sales, they were doubtful, but their accounting dept made enquiries and decided they'd ship to a different address. Anyway, as we've all said, it varies. Paypal will ship an order billed to Wisconsin to Nigeria with a blick. btw, With a blick was supoposed to be without blinking :) ___ Asterisk-Users mailing list Asterisk-Users@lists.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: IPswitch cannot delete lines double lines
You can choose Refresh Extensions from the file menu in IPSwitchBoard, that will delete all extensions and read all from your server again. Thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] I had to cancel Broadvoice, but IPswitch does not like to delete me that line, ... I use instead voipjet, but this one pops up twice, as well as nufone, ... How can I get the name - info into Zap-1 .. Zap-4 (FXS and FXO type)? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set CallerID in zapata.conf with QuadBri or other solution with parallel call signalling
On Sat, 21 May 2005, Companity wrote: The sip phones and the internal phones on the PBX see the number of the calling party correctly (e.g. 040-987654321). Cause we can´t set a callerid to the public phone network (to show the calling party number), we want to show an extension of our numbers on our isdn-bri (asigned by Carrier, e.g. our numbers are 12345-0 to 12345-99). If we use our [snip] Most (or some?) operators will actually allow you to set the outgoing calling party number provided you have signed a separate agreement. ETS 300 092-1 by default requires a strict checking of the calling number (paragraph 9.3). An alternate method available by special ararngement (paragraph 9.4) allows the sending of unscreened calling numbers. In this later case Annex B allows the network to add the network provided calling party number after the unscreen part. Similar rules are in effect in the USA. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip setup
We have applied for LiveVoip termination. Although the account is now available via Web, I have not got any information how to set it up. Can anybody guide me to make my paid advanture a success? You should have received an email with a sample config in it. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Boosting Internet Bandwidth for VOIP
chawki hammoud wrote: There was errors when I tried to start the script recommended by Andrew to boost bandwidth for voip ./rc.tc start RTNETLINK answers: File exists RTNETLINK answers: File exists Looks like you are already running some type of QoS script, you'll need to stop it did befor trying the new script. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.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 doesn't call cellphones
Hi all, I am using a Sangoma with two PRIs. As far as land phones, the calls are fine but it refuses all cellphone calls: My configuration in Zaptel is span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-61 and on Zapata.conf: [channels] language=it context=default switchtype=national signalling=pri_cpe echocancel=yes group=1 callgroup=1 pickupgroup=1 when I call a cellphone I get the following error from Pri Span Debug: login as: root Authenticating with public key rsa-key-20050520 Passphrase for key rsa-key-20050520: Wrong passphrase Authenticating with public key rsa-key-20050520 Passphrase for key rsa-key-20050520: Wrong passphrase Authenticating with public key rsa-key-20050520 Passphrase for key rsa-key-20050520: Last login: Sat May 21 13:16:10 2005 from 212.102.34.109 Linux 2.4.29. [EMAIL PROTECTED]:~# cd /etc/asterisk [EMAIL PROTECTED]:/etc/asterisk# joe zapata.conf Processing '/etc/joe/joerc'...done Processing '/etc/joe/joerc'...done I zapata.conf (Modified) Row 15 Col 1 1:56 Ctrl-K H for help language=it context=default switchtype=national signalling=pri_cpe echocancel=yes group=1 callgroup=1 pickupgroup=1 group = 1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-61 [channels] language=it context=default switchtype=national signalling=pri_cpe echocancel=yes group=1 callgroup=1 Verbosity is at least 7 -- Remote UNIX connection -- Executing Dial(SIP/200-4f96, Zap/g1/3337885836|100|T) in new stack -- Making new call for cr 32778 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 10/0xA) (Originator) Message type: SETUP (5) [04 03 80 90 a3] 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: A-Law (35) [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 06 b1 55 73 65 72 31] Display (len= 6) Charset: 31 [ User1 ] [6c 05 21 81 32 30 30] Calling Number (len= 7) [ 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) '200' ] [70 0b a1 33 33 33 37 38 38 35 38 33 36] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3337885836' ] -- Called g1/3337885836 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 10/0xA) (Terminator) Message type: STATUS (125) [08 03 82 e3 28] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (6) ] Cause data 1: 28 (40, Display IE) [14 01 01] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 10/0xA) (Terminator) Message type: CALL PROCEEDING (2) [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 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 10/0xA) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) -- Channel 0/1, span 1 got hangup request NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 10/0xA) (Originator) Message
RE: [Asterisk-Users] Boosting Internet Bandwidth for VOIP
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chawki hammoud Sent: 21 May 2005 05:32 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Boosting Internet Bandwidth for VOIP There was errors when I tried to start the script recommended by Andrew to boost bandwidth for voip snip iptables v1.2.9: Couldn't load match `p2p':/lib/iptables/libipt_p2p.so: cannot open shared object file: No such file or directory Try `iptables -h' or 'iptables --help' for more information. iptables v1.2.9: Couldn't load match `ipp2p':/lib/iptables/libipt_ipp2p.so: cannot open shared object file: No such file or directory Try `iptables -h' or 'iptables --help' for more information. iptables: No chain/target/match by that name Any suggestions? Doesn't look like iptables is installed properly. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX losing registration
My * box keeps losing its registration to all the servers it is registering to, the only way to fix it is to restart asterisk and then it works fine for another 2 hours or so. I'm on a static IP, but this happens like clockwork every time. I have seen other people that have this problem but never an answer. Please can any guru out there help me. This is the only problem with this system that is keeping me from going live with it. Here is my IAX2 show registry. Host UsernamePerceived Refresh State 216.94.102.***:4569 ** Unregistered 60 Request Sent 139.142.184.***:4569 ** Unregistered 60 Request Sent * them out just for security but these fields show up fine. Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] paging thru sipura-841
Hey steve I remember a tip somewhere where they used a conference room and added all the users into that conference muted, then kicked them out at the end of the call. Sorry I can't remember at all where this was but it looked like it could work. How did you get the autoanswer to work, I have tried different patches and non work? joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Clark Sent: Friday, May 20, 2005 9:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] paging thru sipura-841 Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( extextext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX losing registration
Maybe you connections pass through a stateful firewall , and these states die after some inactivity time... Check it. - Original Message - From: Joel Duffield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 21, 2005 3:45 PM Subject: [Asterisk-Users] IAX losing registration My * box keeps losing its registration to all the servers it is registering to, the only way to fix it is to restart asterisk and then it works fine for another 2 hours or so. I'm on a static IP, but this happens like clockwork every time. I have seen other people that have this problem but never an answer. Please can any guru out there help me. This is the only problem with this system that is keeping me from going live with it. Here is my IAX2 show registry. Host UsernamePerceived Refresh State 216.94.102.***:4569 ** Unregistered 60 Request Sent 139.142.184.***:4569 ** Unregistered 60 Request Sent * them out just for security but these fields show up fine. Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX-IAX Trunking not works
Hello , I want some tips guidance i am sure this topic discuss alot in list,i try my best to solve it by myself try googling looking wiki everywhere but no luck question is iax-iax trunking not working setting,trying each n every option server2 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [saim] username=saim secret=saim type=friend host=dynamic context=from-sip disallow=all allow=gsm [noman] username=saim secret=noman type=friend host=dynamic context=from-sip disallow=all allow=gsm [asteriskser1] type=friend ;auth=md5 ;secret=qwerty context=local ;host=dynamic defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no server1 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [user1] username=user1 secret=user1 type=friend host=dynamic context=from-sip disallow=all allow=gsm [user2] username=user2 secret=user2 type=friend host=dynamic context=from-sip disallow=all allow=gsm [test2] type=friend context=local defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no I am using Kiax soft phone on both servers using codec GSM asterisk latest stable version OS SLES9 ,any help is highly appreciated i had look almost every place in wiki regarding iax trunking but all in vein. Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ChanIsAvail and SIP
All, I was reading over the chanisavail command in the wiki and was wondering a couple things. First and foremost, what does this command do to determine if SIP is available? All I could tell from a debug is that it simply checks to see if the peer's port is open and doesn't run any callflows. Is this true? Second, I understand that running Cut on SIP may be a little difficult. Because the final destination becomes SIP/peer- .. = random characters, because they can be letters and numbers applying a range in Cut wouldn't be possible. Any suggestions on how to get by this? Is there any other var manipulation command? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Confirmation Of Extension Before Transfer?
Is there any way to have the user confirm the extension they are looking to go to before transfering? i.e. You pressed 5 4 3 3 2. Is this correct? 1 - GoTo extensionPressed 2 - Enter extension again Thanks! Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spanish Voice Messages
Hi, It is possible to change the operator voice announcements to spanish? What files do i need to replace o record new ? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.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] I call an USA MOBILE phone and it is registered as ENUM = failed
I was tracking down an error in my dialplan, .. but at the end it
showed, that I called a mobile phone in USA, which had a successfull
ENUM lookup, ...
Could that be?
[trunkUSA]
;
; USA Canada long distance through trunk
;for ENUM drop 9
;exten = _91Z.,1,NoOp(trunkUSA)
exten = _91Z.,1,SetCIDNum(${CALLERIDNUM}|a)
exten = _91Z.,2,EnumLookup(${EXTEN:1})
;exten = _91Z.,3,BackGround(enum-lookup-successful)
exten = _91Z.,3,Dial(${ENUM},30)
exten = _91Z.,4,Hangup
exten = _91Z.,5,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}} ;
VoipJet.com NANPA
exten = _91Z.,6,hangup
exten = _91Z.,7,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
*CLI
-- Executing SetCIDNum(SIP/615-80b2, 615|a) in new stack
-- Executing EnumLookup(SIP/615-80b2, 1860614) in new stack
-- Executing Dial(SIP/615-80b2, |30) in new stack
May 21 23:36:45 WARNING[9733]: app_dial.c:690 dial_exec_full: Dial
argument takes format
(technology1/number1technology2/number2...|optional timeout)
Any ideas?
bye
Ronald
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Re: [Asterisk-Users] Spanish Voice Messages
Hi Guillermo. Currently im using several sounds that you can download in: http://voip-info.org/wiki-Asterisk+sound+files+international there are other good links there to explain how to make it work. But actually is very easy. If you have troubles making it work, let me know. Best Regards. - moy On 5/21/05, Guillermo Salas M [EMAIL PROTECTED] wrote: Hi, It is possible to change the operator voice announcements to spanish? What files do i need to replace o record new ? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.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; ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Spanish Voice Messages
other tip. you can use a text to speech software like naturalvoices from ATT to develop your own sounds. Asterisk will try to look for them in /var/lib/asterisk/sounds/ if the default language is set to english, for spanish you can use the folder /var/lib/asterisk/sounds/es/ More info is available in the voip-info international page. On 5/21/05, Moises Silva [EMAIL PROTECTED] wrote: Hi Guillermo. Currently im using several sounds that you can download in: http://voip-info.org/wiki-Asterisk+sound+files+international there are other good links there to explain how to make it work. But actually is very easy. If you have troubles making it work, let me know. Best Regards. - moy On 5/21/05, Guillermo Salas M [EMAIL PROTECTED] wrote: Hi, It is possible to change the operator voice announcements to spanish? What files do i need to replace o record new ? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.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; -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Re: [Asterisk-Users] NVFaxDetect on Gentoo
On Fri, 2005-05-20 at 12:06 -0300, Juan Luis Moyano wrote: Hi, I've merged asterisk-0.9.0 on a gentoo (kernel 2.4) system using portage ebuilds. I've just got NVFaxDetect .c files from Justin Newman and I'm about to install them. I want to know which is the best way to accomplish this. Thanks in advance. I would like to know that too. I've initiated a discussion through Gentoo Bugzilla see: http://bugs.gentoo.org/show_bug.cgi?id=92747 If only Mr. Newman gave us the link to the source code we could write a ebuild to install/compile it as an addition. -- #Joseph Joseph, We are setting up a location for all the GPL modules. If you have any specific requirements, please let us know. Justin Newman Newman Telecom, Inc. Justin, thank you for the offer. Please give me the URL of app_nv_backgrounddetect.c and app_nv_faxdetect.c I think we need these two to make NVFaxDetect and NVBackgroundDetect to work on Gentoo. I'll pass it over to our group so we can make an ebuild. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Affecting overhead with Runlevel?
This is a _very_ green question, but I am just beginning to explore and learn Linux. Have to admit I avoided it for years due to other obligations but discovering Asterisk has forced my hand. So: knowing that the X11 window GUI is a resource hog, is it appropriate to use the GUI to install and configure various components, then set RUNLEVEL to 3 once all is nicely set up and running cleanly? Would this give the same effect as doing a minimal install or is the mere presence of the installed (yet not inited?) packages too heavy? Corollary: if Asterisk is running as ROOT, is there any benefit to booting at RUNLEVEL 1 to prune the overhead down even further? Or is that really only for debugging or administrative issues? Thanks, /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Boosting Internet Bandwidth for VOIP
You need to instal the module 'libipt_ipp2p.so' --- Tom Fanning [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chawki hammoud Sent: 21 May 2005 05:32 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Boosting Internet Bandwidth for VOIP There was errors when I tried to start the script recommended by Andrew to boost bandwidth for voip snip iptables v1.2.9: Couldn't load match `p2p':/lib/iptables/libipt_p2p.so: cannot open shared object file: No such file or directory Try `iptables -h' or 'iptables --help' for more information. iptables v1.2.9: Couldn't load match `ipp2p':/lib/iptables/libipt_ipp2p.so: cannot open shared object file: No such file or directory Try `iptables -h' or 'iptables --help' for more information. iptables: No chain/target/match by that name Any suggestions? Doesn't look like iptables is installed properly. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Julius Igugu SouthWork Co. Ltd. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Affecting overhead with Runlevel?
Robert Goodyear wrote: So: knowing that the X11 window GUI is a resource hog, is it appropriate to use the GUI to install and configure various components, then set RUNLEVEL to 3 once all is nicely set up and running cleanly? Would this give the same effect as doing a minimal install or is the mere presence of the installed (yet not inited?) packages too heavy? That would work fine. You could still log in to the console and run the Asterisk console (asterisk -R) to watch things work, which is instructive. Corollary: if Asterisk is running as ROOT, is there any benefit to booting at RUNLEVEL 1 to prune the overhead down even further? Or is that really only for debugging or administrative issues? Yes, runlevel 1 (or single user mode as it is often called) is primarily for debugging or administration, and is set to run as few background programs (daemons) as possible. Asterisk may depend upon some of these daemons (email, cron, syslog, to think of of a few), so runlevel 3 is still your best shot. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Spanish Voice Messages
On Sat, 2005-05-21 at 11:08 -0500, Moises Silva wrote: other tip. you can use a text to speech software like naturalvoices from ATT to develop your own sounds. Asterisk will try to look for them in /var/lib/asterisk/sounds/ if the default language is set to english, for spanish you can use the folder /var/lib/asterisk/sounds/es/ More info is available in the voip-info international page. Thank you. I'll be trying the tip. On 5/21/05, Moises Silva [EMAIL PROTECTED] wrote: Hi Guillermo. Currently im using several sounds that you can download in: http://voip-info.org/wiki-Asterisk+sound+files+international there are other good links there to explain how to make it work. But actually is very easy. If you have troubles making it work, let me know. Thank you very much Moises . Best Regards. - moy On 5/21/05, Guillermo Salas M [EMAIL PROTECTED] wrote: Hi, It is possible to change the operator voice announcements to spanish? What files do i need to replace o record new ? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.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; -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.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] LOOKING TO HIRE
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Thursday, May 19, 2005 4:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] LOOKING TO HIRE Or rather, let me take that back. If you do not recognize the value of Perl or Python as appropriate, valid programming tools for certain scenarios (for example, prototyping AGI scripting with Perl), I doubt that /you/ are what I would consider a good programmer. I'm building something around an industrial SBC with the built-in tiny basic intepreter. I guess I'm not a good programmer, huh? Why..., you're no programmer AT ALL! mutter ...stinking hardware-hacking cretins... /mutter -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Spanish Voice Messages
On Sat, 2005-05-21 at 12:51 -0500, Guillermo Salas M wrote: On Sat, 2005-05-21 at 11:08 -0500, Moises Silva wrote: other tip. you can use a text to speech software like naturalvoices from ATT to develop your own sounds. Asterisk will try to look for them in /var/lib/asterisk/sounds/ if the default language is set to english, for spanish you can use the folder /var/lib/asterisk/sounds/es/ More info is available in the voip-info international page. Thank you. I'll be trying the tip. I've copied the es files to /var/lib/asterisk/sounds and the letter and digits to /var/lib/asterisk/sounds/es What is the .conf file where I define the lenguaje=es ? Regards, On 5/21/05, Moises Silva [EMAIL PROTECTED] wrote: Hi Guillermo. Currently im using several sounds that you can download in: http://voip-info.org/wiki-Asterisk+sound+files+international there are other good links there to explain how to make it work. But actually is very easy. If you have troubles making it work, let me know. Thank you very much Moises . Best Regards. - moy On 5/21/05, Guillermo Salas M [EMAIL PROTECTED] wrote: Hi, It is possible to change the operator voice announcements to spanish? What files do i need to replace o record new ? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.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; -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Spanish Voice Messages
On Sat, 2005-05-21 at 13:17 -0500, Guillermo Salas M wrote: On Sat, 2005-05-21 at 12:51 -0500, Guillermo Salas M wrote: On Sat, 2005-05-21 at 11:08 -0500, Moises Silva wrote: other tip. you can use a text to speech software like naturalvoices from ATT to develop your own sounds. Asterisk will try to look for them in /var/lib/asterisk/sounds/ if the default language is set to english, for spanish you can use the folder /var/lib/asterisk/sounds/es/ More info is available in the voip-info international page. Thank you. I'll be trying the tip. I've copied the es files to /var/lib/asterisk/sounds and the letter and digits to /var/lib/asterisk/sounds/es What is the .conf file where I define the lenguaje=es ? Solved: Added language=es in sip.conf Thank you. Regards, On 5/21/05, Moises Silva [EMAIL PROTECTED] wrote: Hi Guillermo. Currently im using several sounds that you can download in: http://voip-info.org/wiki-Asterisk+sound+files+international there are other good links there to explain how to make it work. But actually is very easy. If you have troubles making it work, let me know. Thank you very much Moises . Best Regards. - moy On 5/21/05, Guillermo Salas M [EMAIL PROTECTED] wrote: Hi, It is possible to change the operator voice announcements to spanish? What files do i need to replace o record new ? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.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; -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Uncommon callback
Hello! I got an interesting task to make with asterisk: pstn--- * ---sip--- * pstn This sounds common till now. What I have to make is: 1.the call is routed through PSTN to asterisk1 (#1) which has ISDN PRI interface(s) - leg1 2.#1 doesn't pick up the call, neither rejects, it just place into state CALL PROGRESS (?) [maybe gives back alerting tone? probably not] 3.#1 notifies that there is a call to number B from number A to system asterisk2 (#2) - through non-voip protocol [xml-rpc, anything else, doesn't matter here] 4.#2 dials number B through PSTN - leg2 5.#2 dials #1 (on number #1 sent in notification) through SIP - leg3 6.#2 interconnects leg2 and leg3 without ISDN CONNECT 7.#1 interconnects leg1 and leg2 without ISDN CONNECT 8.when B picks up the phone, channels are getting CONNECT message I know that my explanation can be a bit mess, but I wasn't able to write it down better. I was thinking about putting the incoming call on #1 into meetme, but meetme looks to establish the call, so it is charged from the 1st second - that's not good in this case. In #2 I thought, .call feature can initiate the call out to number B (leg2) which will dial to #1 using SIP. I have many black holes right now. Any help appreciated! Kind regards, Tamas ___ Asterisk-Users mailing list Asterisk-Users@lists.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 on NetBSD
I was reading on the wiki that Asterisk runs very solid on NetBSD. Can anyone comment? What is the definition of solid? Who is running Asterisk on NetBSD and which version of Asterisk are you running? Also, I know there is limited support for Digium cards on NetBSD, but is there any support at all? Would a TE410P work in NetBSD? I want to build a very simple VoIP to TDM gateway. My idea is only to have the box connect to the PSTN with a TE410P and send or receive calls to other asterisk boxes using IAX2. The communication with the other boxes should be limited to only ulaw since they will be connected via, at the very least 100Mbps FastE. It doesn't even have a need to do any SIP or any other protocols, other than IAX2. Which version of Asterisk would you recommend would be most stable for this? Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Working Xten, Asterisk, double-NAT configs out there?
All, I have my * box NAT'd with all ports forwarded that are SIP related (based on Wiki). I also have nat=yes, externalip=WAN address of firewall, internalip=LAN network of *. I have my Xten soft phone on a PC which is NAT'd behind firewall with ports forwarded. I have also followed instructions on Wiki for Xten. I can authenticate fine, and sip show peers shows my extension is OK, but whenever I dial another SIP or zap channel, I can a max retries exceeded on my * box. Can somebody post a working config with * behind firewall w/ports forwarded and xten behind firewall w/ports forwarded? Thanks, -- Aaron ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Confirmation Of Extension Before Transfer?
Try to use macro's I am not the one to ask about them, I couldn't give you an example off the top of my head. But read up on them on the wiki, and i'm sure they can do what you want very easily. Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Stearne Sent: Saturday, May 21, 2005 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Confirmation Of Extension Before Transfer? Is there any way to have the user confirm the extension they are looking to go to before transfering? i.e. You pressed 5 4 3 3 2. Is this correct? 1 - GoTo extensionPressed 2 - Enter extension again Thanks! Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX-IAX Trunking not works
Okay sounds like a stupid question but just to be clear do you have some sort of timer on both machines? Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adnan Ahmed Sent: Saturday, May 21, 2005 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX-IAX Trunking not works Hello , I want some tips guidance i am sure this topic discuss alot in list,i try my best to solve it by myself try googling looking wiki everywhere but no luck question is iax-iax trunking not working setting,trying each n every option server2 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [saim] username=saim secret=saim type=friend host=dynamic context=from-sip disallow=all allow=gsm [noman] username=saim secret=noman type=friend host=dynamic context=from-sip disallow=all allow=gsm [asteriskser1] type=friend ;auth=md5 ;secret=qwerty context=local ;host=dynamic defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no server1 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [user1] username=user1 secret=user1 type=friend host=dynamic context=from-sip disallow=all allow=gsm [user2] username=user2 secret=user2 type=friend host=dynamic context=from-sip disallow=all allow=gsm [test2] type=friend context=local defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no I am using Kiax soft phone on both servers using codec GSM asterisk latest stable version OS SLES9 ,any help is highly appreciated i had look almost every place in wiki regarding iax trunking but all in vein. Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.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: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Working Xten, Asterisk, double-NAT configs out there?
I have (had) a similar setup at one time. I'm running freebsd/pf on each nat box. Asterisk is behind one, an xten softphone behind the other. I watched the SIP traffic on both the incoming and outgoing interfaces (pre/post nat) of each box. You can then generally see whats wrong, and as a huge plus, learn a lot more about how SIP/RTP actually works .. That coupled with the firewall log (I deny/log all by default) and the firewall debut output ( I have had, and am still having in a couple of weird cases, state clashes) you can usually identify the problem .. Or you could post your details and let someone have a stab at helping, unless there are any psychics here :-) Tim Aaron O'Hara wrote: All, I have my * box NAT'd with all ports forwarded that are SIP related (based on Wiki). I also have nat=yes, externalip=WAN address of firewall, internalip=LAN network of *. I have my Xten soft phone on a PC which is NAT'd behind firewall with ports forwarded. I have also followed instructions on Wiki for Xten. I can authenticate fine, and sip show peers shows my extension is OK, but whenever I dial another SIP or zap channel, I can a max retries exceeded on my * box. Can somebody post a working config with * behind firewall w/ports forwarded and xten behind firewall w/ports forwarded? Thanks, -- Aaron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX losing registration
The firewall I'm using is a Linksys BEFSR41 V3 it says that it uses The router uses NAT and TCP/IP port inspections not stateful inspections. Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Francisco A. Lozano Sent: Saturday, May 21, 2005 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX losing registration Maybe you connections pass through a stateful firewall , and these states die after some inactivity time... Check it. - Original Message - From: Joel Duffield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 21, 2005 3:45 PM Subject: [Asterisk-Users] IAX losing registration My * box keeps losing its registration to all the servers it is registering to, the only way to fix it is to restart asterisk and then it works fine for another 2 hours or so. I'm on a static IP, but this happens like clockwork every time. I have seen other people that have this problem but never an answer. Please can any guru out there help me. This is the only problem with this system that is keeping me from going live with it. Here is my IAX2 show registry. Host UsernamePerceived Refresh State 216.94.102.***:4569 ** Unregistered 60 Request Sent 139.142.184.***:4569 ** Unregistered 60 Request Sent * them out just for security but these fields show up fine. Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] having asterisk play music on hold to callers while phone rings?
hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ Asterisk-Users mailing list Asterisk-Users@lists.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 on NetBSD
On 5/21/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I was reading on the wiki that Asterisk runs very solid on NetBSD. Can anyone comment? What is the definition of solid? Who is running Asterisk on NetBSD and which version of Asterisk are you running? Also, I know there is limited support for Digium cards on NetBSD, but is there any support at all? Would a TE410P work in NetBSD? I want to build a very simple VoIP to TDM gateway. My idea is only to have the box connect to the PSTN with a TE410P and send or receive calls to other asterisk boxes using IAX2. The communication with the other boxes should be limited to only ulaw since they will be connected via, at the very least 100Mbps FastE. It doesn't even have a need to do any SIP or any other protocols, other than IAX2. Which version of Asterisk would you recommend would be most stable for this? Couple of thoughts. First off, go with what is known to work. In a business critical application I would use linux. That said, I run asterisk on Freebsd with the zaptel ztdummy driver and everything has worked fine. I've been running CVS HEAD for a couple of months with the following obervations which probably also apply to NetBSD. This is on my personal at home setup though, not for our business. I run Debian for the business. 1. You can't compile zaptel from cvs, you have to use a bsd port. Not sure if the Freebsd port would work with Netbsd, but it should without too much tweaking. 2. mpg123 always hangs on freebsd. Use madplayer instead. 3. Zaptel on freebsd has limited hardware support. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Working Xten, Asterisk, double-NAT configs out there?
Tim, Aside from the firewall logs in /var/log/messages, what tools did u find most helpful for seeing SIP/RTP traffic? What are some of the key things to look for to see if there's a problem? Aaron On Sat, 2005-21-05 at 14:04 -0600, Tim Pushor wrote: I have (had) a similar setup at one time. I'm running freebsd/pf on each nat box. Asterisk is behind one, an xten softphone behind the other. I watched the SIP traffic on both the incoming and outgoing interfaces (pre/post nat) of each box. You can then generally see whats wrong, and as a huge plus, learn a lot more about how SIP/RTP actually works .. That coupled with the firewall log (I deny/log all by default) and the firewall debut output ( I have had, and am still having in a couple of weird cases, state clashes) you can usually identify the problem .. Or you could post your details and let someone have a stab at helping, unless there are any psychics here :-) Tim Aaron O'Hara wrote: All, I have my * box NAT'd with all ports forwarded that are SIP related (based on Wiki). I also have nat=yes, externalip=WAN address of firewall, internalip=LAN network of *. I have my Xten soft phone on a PC which is NAT'd behind firewall with ports forwarded. I have also followed instructions on Wiki for Xten. I can authenticate fine, and sip show peers shows my extension is OK, but whenever I dial another SIP or zap channel, I can a max retries exceeded on my * box. Can somebody post a working config with * behind firewall w/ports forwarded and xten behind firewall w/ports forwarded? Thanks, -- Aaron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on NetBSD
Additionally you may want to check http://www.pkgsrc.org/ to see if there is a package for NetBSD [pkgsrc related tangent] Sadly there is not one yet for interix the posix subsystem that is not a sandbox (cygwin) not an emulator (bochs) not a virtual machine (vmware, virtual pc) it runs side by side windows and directly interfaces with the kernel, so 'ps' shows all processes windows and interix, processes can talk to each other, etc. With pkgsrc you have a weaker darwin type thing for windows, with more software it may even be as good! I wonder if anyone has tried to compile asterisk for interix (a COM (most likely) connector would probably be needed to interface with real hardware as the driver would be windows). http://www.microsoft.com/windows/sfu (free) -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.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] having asterisk play music on hold to callers while phone rings?
use option m in the cmd dial. Cheers, Jon. On Saturday 21 May 2005 03:26 pm, hank smith wrote: hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Affecting overhead with Runlevel?
On May 21, 2005, at 10:29 AM, Johnathan Corgan wrote: Robert Goodyear wrote: So: knowing that the X11 window GUI is a resource hog, is it appropriate to use the GUI to install and configure various components, then set RUNLEVEL to 3 once all is nicely set up and running cleanly? Would this give the same effect as doing a minimal install or is the mere presence of the installed (yet not inited?) packages too heavy? That would work fine. You could still log in to the console and run the Asterisk console (asterisk -R) to watch things work, which is instructive. Noted. To clarify, will dropping back to runlevel 3 still ensure a smaller set of processes that would be as non-intrusive as if I had installed Linux with console/command line support only or would there still be stuff hanging around that's inextricably there because I had _at one time_ installed and run the GUI? thx /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.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] having asterisk play music on hold to callerswhile phone rings?
Edit the extensions.conf and put an m at the end of the dial line. Do you have hold music otherwise? Sincerely; Gary Lawrence ITcom.Net 866.4ITcom1 866.448.2661 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smith Sent: Saturday, May 21, 2005 4:26 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] having asterisk play music on hold to callerswhile phone rings? hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] having asterisk play music on hold to callerswhile phone rings?
what config is this found in? email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: Jon Gabrielson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 21, 2005 2:05 PM Subject: Re: [Asterisk-Users] having asterisk play music on hold to callerswhile phone rings? use option m in the cmd dial. Cheers, Jon. On Saturday 21 May 2005 03:26 pm, hank smith wrote: hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] having asterisk play music on hold tocallerswhile phone rings?
yep I have hold music other wise looks like I am going to have to go in to the [EMAIL PROTECTED] and configure it via that method can you give me pointers on what the dial line lookslike so I dont screw this thing up?? they dont recommend editing this stuff bye hand unless you know what you are doing. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 - Original Message - From: Gary Lawrence To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Saturday, May 21, 2005 2:09 PM Subject: RE: [Asterisk-Users] having asterisk play music on hold tocallerswhile phone rings? Edit the extensions.conf and put an m at the end of the dial line. Do you have hold music otherwise? Sincerely; Gary Lawrence ITcom.Net 866.4ITcom1 866.448.2661 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smithSent: Saturday, May 21, 2005 4:26 PMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] having asterisk play music on hold to callerswhile phone rings? hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___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 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] having asterisk play music on holdtocallerswhile phone rings?
Yours could look totally different than
mine depending on how you route calls.
It will start with exten and
have the word Dial in it. You may have several lines that you
need to change...
In the below example change the r at the
end to an m.
exten = _NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}||r
yep
I have hold music other wise
looks like I am going to have to go
in to the [EMAIL PROTECTED] and configure it via
that method
can you give me pointers on what the
dial line lookslike so I dont screw this thing up??
they dont recommend editing this
stuff bye hand unless you know what you are doing.
thanks
hank
email:
[EMAIL PROTECTED]
gmail:
[EMAIL PROTECTED]
msn messenger:
[EMAIL PROTECTED]
aim:
hanksmith5
skype:
hanksmith5
- Original Message -
From: Gary Lawrence
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Saturday,
May 21, 2005 2:09 PM
Subject: RE:
[Asterisk-Users] having asterisk play music on hold tocallerswhile phone rings?
Edit the extensions.conf
and put an m at the end of the dial line.
Do you have hold music
otherwise?
Sincerely;
Gary Lawrence
ITcom.Net
866.4ITcom1
866.448.2661
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank smith
Sent: Saturday, May 21, 2005 4:26
PM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] having
asterisk play music on hold to callerswhile phone rings?
hello how do I set up asterisk to
play music on hold to callers while it rings my phones?
I am using the amp portal to
configure the asterisk pbx just to let you all know.
thanks
hank
email:
[EMAIL PROTECTED]
gmail:
[EMAIL PROTECTED]
msn messenger:
[EMAIL PROTECTED]
aim:
hanksmith5
skype:
hanksmith5
___
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[Asterisk-Users] zaphfc error
I have problems to install zaphfc on system base on knoppix 3.8. with kernel update to 2.6.11.8 with bristuff-0.2.0-RC8d-CVS please help !! ztcfg ZT_SPANCONFIG failed on span 1: No such device or address (6) lspci :00:08.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) :00:09.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) download bristuff-0.2.0-RC8d-CVS update: /usr/src/linux/include/asm/irq.h In line 16 change: #include irq-vectors.h to: #include mach-default/irq-vectors.h /usr/src/linux/include/asm/mpspec.h In line 6 change: mach_mpspec.h to: asm/mach-default/mach_mpspec.h /usr/src/linux/include/asm/smp.h In line 71 change: mach_apicdef.h to: asm/mach-default/mach_apicdef.h - cd zaptel make linux26 make install (no compile errors) cd libpri make make install (no compile errors) cd zaphfc make linux26 make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/bristuff-0.2.0-RC8d-CVS/zaphfc ZAP=-I/usr/src/bristuff-0.2.0-RC8d-CVS/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.11.8' Building modules, stage 2. MODPOST *** Warning: zt_register [/usr/src/bristuff-0.2.0-RC8d-CVS/zaphfc/zaphfc.ko] undefined! *** Warning: zt_transmit [/usr/src/bristuff-0.2.0-RC8d-CVS/zaphfc/zaphfc.ko] undefined! *** Warning: zt_receive [/usr/src/bristuff-0.2.0-RC8d-CVS/zaphfc/zaphfc.ko] undefined! *** Warning: zt_ec_chunk [/usr/src/bristuff-0.2.0-RC8d-CVS/zaphfc/zaphfc.ko] undefined! *** Warning: zt_unregister [/usr/src/bristuff-0.2.0-RC8d-CVS/zaphfc/zaphfc.ko] undefined! make[1]: Leaving directory `/usr/src/linux-2.6.11.8' [EMAIL PROTECTED]:/usr/src/bristuff-0.2.0-RC8d-CVS/zaphfc# make install install -D -m 644 zaphfc.ko /lib/modules/`uname -r`/misc/zaphfc.ko my /etc/zaptel.conf loadzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 my /etc/Asterisk/zapata.conf [channels] language=en switchtype=euroisdn ;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier to unknown. pridialplan=unknown prilocaldialplan=unknown signalling = bri_net_ptmp usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 language=en switchtype=euroisdn ;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier to unknown. pridialplan=unknown prilocaldialplan=unknown signalling = bri_net_ptmp usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes [channels] language=en switchtype=euroisdn ;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier to unknown. pridialplan=unknown prilocaldialplan=unknown signalling = bri_net_ptmp usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 lsmod Module Size Used by zaphfc 15892 0 zaptel225284 1 zaphfc crc_ccitt 5888 1 zaptel --- ___ Nuovo Yahoo! Messenger: E' molto più divertente: Audibles, Avatar, Webcam, Giochi, Rubrica Scaricalo ora! http://it.messenger.yahoo.it ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Working Xten, Asterisk, double-NAT configs out there?
Aaron O'Hara wrote: Tim, Aside from the firewall logs in /var/log/messages, what tools did u find most helpful for seeing SIP/RTP traffic? What are some of the key things to look for to see if there's a problem? Oh, I generally use tcpdump to grab the packets and save them to a file, then load them with ethereal at a workstation to analzye them. You can also use ngrep to watch sip traffic. Some things to watch for: - make sure all SIP traffic on the outside interface of each firewall advertises its external IP to the other side - make sure that the host and port in the SDP payload of the SIP packet looks sane (port within redirection range, host is external IP) For example, here is a SIP problem I just troubleshot: I have a friend with a d-link router and a sipura 2000 behind it, connecting to free world dialup. I have an * server behind a NAT and *thought* everything was working ( I have a few SIP connections that all seem to work). When he called me, everything was fine When I called him, we had one way audio. one way audio + NAT's on both sides = 99% Probability it is NAT related I started by watching the traffic arrive on my external interface (using ngrep) . I made sure the INVITE from him (actually from free world dialup) looked sane (contained his external IP address in the SDP payload, and the port was withing the range programmed into his sipura 2000 - so it would be forwarded properly by his redirection rules on the d-link). Then I watched my response, and make sure again that the SDP payload in my response was advertising my IP address and a port within the range secified in my rtp.conf. Everything seemed ok. Just to make sure, I grabbed a bunch of packets with tcpdump and loaded them up in ethereal. Everything looked like it should work. So next, I watched my firewall log in realtime (the drop log) to see if I was dropping anything inadvertently on the firewall. Nothing. I open two sessions into the firewall, and watch all communication to either fwd or my friend (in just a one line summary per packet) on both the Internal and External interface. This shows a different picture. I can follow the SIP conversation coming from FWD and my response. It is when we begin trying to talk (using RTP) that I see that his RTP packets are in fact coming to me, but I don't see them cross the firewall (I see them on the outside, and not on the inside). I know that my firewall is not dropping them, so it must be something else. I enabled debugging on my firewall (pfctl -x loud) and watched my system log. Sure enough, I am getting a state error on every inbound RTP packet from him. I have had this problem before, and it has to do with pf using the state table to handle redirects. I had established a state already for some reason (probably a keep-state rule somewhere) that is clashing with the inbound rdr. I just found the problem, I havn't fixed it yet, but at least I know where it is. Interestingly it wasn't really a NAT problem, but more how I've decided to implement my firewall. Sorry if this is long and maybe obvious to you, but this is basically the process that I follow whenever I have SIP or NAT problems. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Affecting overhead with Runlevel?
Robert Goodyear wrote: Noted. To clarify, will dropping back to runlevel 3 still ensure a smaller set of processes that would be as non-intrusive as if I had installed Linux with console/command line support only or would there still be stuff hanging around that's inextricably there because I had _at one time_ installed and run the GUI? No, runlevel 3 typically doesn't include any graphical console processes. Having them installed but not running only wastes disk space; there would be no difference to Asterisk from a CLI only installation. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Sipura 3000 Question
[EMAIL PROTECTED] is believed to have said: I don't know if it is a phone like issue or not, but try the SPA-3000 setup at http://geekgazette.com. -Kerry Kerry, thanks for the hint. A first try did not get better results, but I was doing it very quickly.. Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.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: Sipura 3000 Question
[EMAIL PROTECTED] is believed to have said: In the advanced options there are a few options for hang-up detection including tone detection, and silence detection. They also have parameters to adjust timing and sensitivy. IIRC, they are not enabled by default. Nathan, thanks: this is something I still have to try systematically. Has anybody hints to give as where to find understandable (by the uninitiated..) documentation of what the telecoms do in Europe with regards to signaling the hangup condition? Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAXTEl down
Is iaxtel down? Ive been getting this: May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest: Auto-congesting call due to slow response -- IAX2/Iaxtel-12 is circuit-busy -- Hungup 'IAX2/Iaxtel-12' is it down or am I doing something wrong? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to create channel of type 'IAX2' (cause 3)
What does it mean? How to solve it? -- Executing Dial(Local/[EMAIL PROTECTED],2, IAX2/[EMAIL PROTECTED]/011886229xx) in new stack May 20 18:02:17 NOTICE[31410]: app_dial.c:973 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.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] realtime app data formatting
On the wiki it say's that if you use the Goto commands you need to replace ',' with '|' in the app data field. But in the examples it uses '|' in place of ',' in the Dial command also in a couple of places. Is it safe to replace ',' with '|' everywhere in the app data field when using realtime? Or should I still to substituting ',' with '|' only for the Goto commands? I'm putting together a quick perl script that parses extensions.conf and any include files, then writes out the sql commands to put everything into the database. I'll post it on the wiki when it's finished. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Uncommon callback
Your explanation is really messy but from what I understand it seem like you just want to be able to take incoming call from one asterisk server and forward it to another asterisk sever over IP that would terminate the call but dont connect calls untill B party answers the phone on second asterisk server. If thats the case its very simple to do with just standard iax2 and proper extensions regards m. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] realtime app data formatting
Snacktime, I've found that the pipe thingy ( | ) is needed anywhere in your extensions table that a comma ( , ) would normally be. In your SIP peers/users/friends table/s, you need a semicolon ( ; ). This is as much as I know for sure at present :-) snacktime wrote: On the wiki it say's that if you use the Goto commands you need to replace ',' with '|' in the app data field. But in the examples it uses '|' in place of ',' in the Dial command also in a couple of places. Is it safe to replace ',' with '|' everywhere in the app data field when using realtime? Or should I still to substituting ',' with '|' only for the Goto commands? I'm putting together a quick perl script that parses extensions.conf and any include files, then writes out the sql commands to put everything into the database. I'll post it on the wiki when it's finished. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers, Mattt. VoIP made easy - http://voip.abovenetworks.net Convergent network specialists - http://abovenetworks.net I have an inferiority complex, but it's not a very good one... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFCR2 Venezuela with libunicall
I have a setup for a 30 incoming channels with telcel. The incoming is R2, they told me the outgoing is MF not R2. If the other channels are fxo, you should change your zaptel.conf so you can use zapata.conf and comment out those channels on unicall.conf. ia On 5/20/05, Andres Maduro [EMAIL PROTECTED] wrote: Hi, I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code. All libs compiled successfully and the E1 have a green light! I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working. My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P E1/T1. I have loaded in the followin order the kernel modules: 1. zaptel 2. wcte11xp 3. wcfxs The E1 is configured as this: the first 15 channels are for incoming calls using MFC/R2 Venezuela protocol, the last 15 channels are configured as a normal analog line, you pick up the channel and hear dial tone, you then only need to send dtmf to place a call (how I configure this channels ??), I guess with fxo loop start in the E1 channel ?? The problem I have is that I can't bring up asterisk, when I try, I receive the following error on the logs about being unable to load chan_zap: May 20 12:01:05 ERROR[13693]: Signalling requested is FXO Kewlstart but line is in R2 Signalling signalling May 20 12:01:05 ERROR[13693]: Unable to register channel '1' May 20 12:01:05 WARNING[13693]: chan_zap.so: load_module failed, returning -1 May 20 12:01:05 WARNING[13693]: Loading module chan_zap.so failed! I am including zaptel.conf, zapata.conf and unicall.conf to see if you can help me. zttool shows both cards in an OK status. -- zaptel.conf start -- # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 # Span 2: WCTDM/0 Wildcard TDM400P REV E/F Board 1 fxoks=32 fxoks=33 fxsks=34 fxsks=35 # Global data loadzone= us defaultzone = us --- zaptel.conf end --- --- zapata.conf start --- ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf ; (extensiones) signalling = fxo_ls channel = 32-33 signalling = fxs_ls channel = 34-35 --- zapata.conf end --- --- unicall.conf start --- ; ; Unicall telephony channel driver ; ; Sample configuration file ; [channels] ; ; Default language ; ;language=en ; ; Default context ; context=default ; ; Whether or not to use caller ID ; usecallerid=yes ; ; Whether or not to hide outgoing caller ID ; hidecallerid=no ; ; Whether or not restrict outgoing caller ID (will be sent as ANI only, not available for the user) ; Mostly use with FXS ports ; ;restrictcid=no ; ; Support Caller*ID on Call Waiting ; callwaitingcallerid=yes ; ; Support three-way calling ; threewaycalling=yes ; ; Support flash-hook call transfer (requires three way calling) ; transfer=yes ; ; Support call forward variable ; cancallforward=yes ; ; Whether or not to support Call Return (*69) ; callreturn=yes ; ; Enable echo cancellation ; Use either yes, no, or a power of two from 32 to 256 if you wish ; to actually set the number of taps of cancellation. ; echocancel=yes ; ; Generally, it is not necessary (and in fact undesirable) to echo cancel ; when the circuit path is entirely TDM. You may, however, reverse this ; behavior by enabling the echo cancel during pure TDM bridging below. ; echocancelwhenbridged=yes ; ; In some cases, the echo canceller doesn't train quickly enough and there ; is echo at the beginning of the call. Enabling echo training will cause ; asterisk to briefly mute the channel, send an impulse, and use the impulse ; response to pre-train the echo canceller so it can start out with a much ; closer idea of the actual echo. Value may be yes, no, or a number of ; milliseconds to delay before training (default = 400) ; ;echotraining=yes
[Asterisk-Users] Erissson Webswitch 100
Somebody have a Ericsson WebSwith 100? I had 2 of them, but, unfortunatelly,I have only one CD that broked inside the drive. If somebody have the mencioned CD that can send me, I'll be grate. Bernardino Campos [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: failure notice
You are in luck: http://bugs.digium.com/view.php?id=4037 -Matthew From: Richard Z [EMAIL PROTECTED] Reply-To: Richard Z [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 20 May 2005 21:17:11 -1000 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] acd with mysql or ast_data support Hi, I am using ACD, i.e. application Queue(). Is there a way to use mysql for the configuration file? Thanks, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX losing registration
Joel Duffield wrote: The firewall I'm using is a Linksys BEFSR41 V3 it says that it uses The router uses NAT and TCP/IP port inspections not stateful inspections. Make sure that your are using qualify=xxx for your IAX2 peers. For example, if you set it to 400 (this is in iax.conf in the definition for a particular account), it would send a request every 400ms (and mark the peer as unreachable if it goes over this amount). If you still get problems, lower the number. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX provider using Broadvox's network?
Hi. I'd really like to start using Broadvox or one of the many companies that resell connectivity to their network, since they are the only VoIP provider out there that solidly advertises full support for T.38 (I'd be using the openh323 stuff for faxing, since Asterisk doesn't do T.38). However, I really like using IAX for my voice calls. Is there any way to have both? Ultimately I'd like to be doing fax detection in Asterisk with a handoff to the inbound fax receive via T.38, but I don't think Asterisk is ready for that yet. - a ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mysql CDR
Rodrigo Otavio de Fraga wrote: Hi, When I finished a call, the asterisk give a message : FAILED TO INSERT INTO DATABASE. Make sure that the details inside cdr_mysql.conf are correct. I.E. if it has username bob, password fred, host 127.0.0.1, run mysql -u bob -p Then it will ask you for a password. Type fred (or whatever your password is). Then it should connect (if the user/pass is in the db). If you then type use cdr (where cdr is the name of your cdr database) it should work. So, if you go through these steps, you should be able to find out where the problem is. (BTW: you did create the tables didn't you?) :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] realtime app data formatting
Got another question now after digging into this. How are regular include statements and ignorepat implemented in realtime? Do I just add them as additional fields to the extensions table I am using? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.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] CallerID
What re you guys doing for windows callerid from Asterisk besides using yac? Any other working software? I have the tapi driver installed but all software I have tried doesn't seem to work or doesn't support the asterisk tapi driver. Any suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] realtime app data formatting
On 5/21/05, snacktime [EMAIL PROTECTED] wrote: Got another question now after digging into this. How are regular include statements and ignorepat implemented in realtime? Do I just add them as additional fields to the extensions table I am using? Chris Well that doesn't make any sense now that I look at it. Supporting them in realtime doesn't make any sense either really, so never mind on that question. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Affecting overhead with Runlevel?
On May 21, 2005, at 3:46 PM, Johnathan Corgan wrote: Robert Goodyear wrote: Noted. To clarify, will dropping back to runlevel 3 still ensure a smaller set of processes that would be as non-intrusive as if I had installed Linux with console/command line support only or would there still be stuff hanging around that's inextricably there because I had _at one time_ installed and run the GUI? No, runlevel 3 typically doesn't include any graphical console processes. Having them installed but not running only wastes disk space; there would be no difference to Asterisk from a CLI only installation. Outstanding. Thanks for your time. So that's the answer to my question; ensuring that unneeded stuff will be dormant after I'm done. /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.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] CallerID
Anton Krall wrote:
What re you guys doing for windows callerid from Asterisk besides using yac?
Any other working software?
I use:
MSN Messenger (this is a bit slow - uses centericq)
===
exten = s,2,System(/bin/echo -e 'Incoming Call From: ${CALLERIDNAME},
${CALLERIDNUM} To: ${ARG3} Received:
${DATETIME:0:2}/${DATETIME:3:2}/${DATETIME:4:4} at ${DATETIME:9}' |
centericq -s msg -p msn -t [EMAIL PROTECTED])
Windows Popup (fast but missing on XP - uses smbclient)
===
exten = s,1,System(/bin/echo -e 'Incoming Call From: ${CALLERIDNAME},
${CALLERIDNUM} To:${ARG3} Received:
${DATETIME:0:2}/${DATETIME:3:2}/${DATETIME:4:4} at
${DATETIME:9}'|/usr/bin/smbclient -M ${ARG4})
SMS to my cellphone (only if I missed a call - uses smsx AGI)
=
exten = s,10,AGI(smsx|64211387245|txt|You missed an incoming call from
${CALLERIDNAME} - ${CALLERIDNUM} to ${ARG3} on
${DATETIME:0:2}/${DATETIME:3:2}/${DATETIME:4:4} at ${DATETIME:9})
Sorry about the line splits :)
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Boosting Shared Internet Bandwidth for Asterisk
Robert Goodyear wrote: On May 20, 2005, at 8:11 AM, chawki hammoud wrote: --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: http://www.mixdown.ca/~andrew/dump/rc.tc. It's what Could you please tell me where and how to install it Thanks. GOOGLE. LEARN. DEPLOY. You need a primer in IP networking before you endeavor to play with packet shaping or you'll be stabbing in the dark. You also need to ascertain whether or not it will be a complete waste of time if/when your provider completely ingores QoS. Assuming your provider completely ignores QOS, it is still not a complete waste of time. If for example you have 5 people on the LAN, 4 uploading files to a remote server and 1 trying to make a phone call. If you were to put the packets for the uploads into say bin 3 and Voip into bin 1, your customers would definitely notice an increase in quality. Agreed that's it's not going to work for inbound if your provider ignores it, but you still control outbound (and on asymmetric connections, outbound usually has the least bandwidth). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX-IAX Trunking not works
Joel Duffield wrote: Okay sounds like a stupid question but just to be clear do you have some sort of timer on both machines? Joel And of course you would want more than one channel to see the benefits of trunking. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] realtime app data formatting
Here is a quick script that will parse extensions.conf, any files
included via #include, and print out the sql commands to put them into
mysql.
I'll add on routines to do the same for sip, iax, and voicemail when I
get the chance.
Chris
-
#!/usr/bin/perl
# Copyright (c) 2005 Chris Ochs
# Permission is hereby granted, free of charge, to any person
# obtaining a copy of this software and associated documentation
# files (the Software), to deal in the Software without restriction,
# including without limitation the rights to use, copy, modify, merge,
# publish, distribute, sublicense, and/or sell copies of the Software,
# and to permit persons to whom the Software is furnished to do so,
# subject to the following conditions:
# The above copyright notice and this permission notice shall be
# included in all copies or substantial portions of the Software.
# THE SOFTWARE IS PROVIDED AS IS, WITHOUT WARRANTY OF ANY KIND,
# EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
# OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
# IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
# ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
# TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
# SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
use IO::File;
$path = /etc/asterisk;
my $database = asterisk;
my $table = extensions;
my $context;
$file = $ARGV[0];
if(!-e $path/$file || $file eq '') {
print File does not exist\n;
exit;
}
print USE $database;\n;
parse_config($file);
sub parse_config {
my ($file) = @_;
my $fh = IO::File-new($path/$file);
while ($line = $fh) {
chomp $line;
if ($line =~/^\[(.*)\]$/) {
$context = $1;
}elsif ($line =~/(^[\s]{0,9}#include[\s]{0,9}.*$)/) {
my $file_include = $1;
$file_include =~s/#include[\s]{0,9}//g;
$file_include =~s/\s//g;
parse_config($file_include);
}elsif ($line
=~/(^[\s]{0,9}exten[\s]{0,9}=[\s]{0,9})([_.XN0-9a-zA-Z]{0,20}),([_.XN0-9a-zA-Z]{0,20}),(.*)/)
{
my (undef,$exten,$priority,$action) = ($1,$2,$3,$4);
my ($app,$appdata);
if ($action =~/(^[a-zA-Z]{0,32})\((.*)\)/) {
($app,$appdata) = ($1,$2);
}elsif($action =~/(^[a-zA-Z]{0,32}),(.*)/) {
($app,$appdata) = ($1,$2);
}else{
$app = $action;
$appdata = '';
}
$appdata =~s/,/\|/g;
print INSERT INTO extensions
(context,exten,priority,app,appdata) VALUES ('$context', '$exten',
'$priority', '$app', '$appdata');\n;
}
}
$fh-close;
}
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[Asterisk-Users] IP header Bandwidth Reduction
Hi: Internet Bandwidth in my country is expensive so I am trying to figure out a way to use the most of what I have. All the calls are between two servers only. How can I reduce the ip header bandwidth to the minimuim whether I am making one call or multiple calls? __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set CallerID in zapata.conf with QuadBri or other solution with parallel call signalling
I can't tell you how to resolve your issue, but I can tell you about
mine. I was fighting for setting my outgoing number (MSN /
bri_cpe_ptmp), and showing or hiding the number, with Swisscom operator.
Showing or hiding the number is resolved by the CallingPres command. For
me, values 0 and 32 worked.
Reference:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20CallingPres
Setting the outgoing number was another headache.While incoming
CallerIDs had 7 digits, the ougoing MSN must be set to 9 digits. The
formerly called 'zone prefix', now fully included inside the number, has
to be sent as well.
I hope this can help you.
Jean-Christophe
my conf:
[pstn-out]
exten = _.,1,Macro(setmsn,${CALLERIDNUM})
exten = _.,2,CallingPres(32) ; 0 to show, 32 to hide
exten = _.,3,SetCallerID(${MSN})
exten = _.,4,Dial(Zap/g1/${EXTEN})
exten = _.,5,Congestion
[macro-setmsn]
exten = s,1,GotoIf($[${ARG1} = 120]?200)
exten = s,2,GotoIf($[${ARG1} = 121]?200)
exten = s,3,GotoIf($[${ARG1} = 122]?200)
exten = s,4,GotoIf($[${ARG1} = 123]?200)
exten = s,6,GotoIf($[${ARG1} = 131]?202)
exten = s,7,SetGlobalVar(MSN=3)
exten = s,200,SetGlobalVar(MSN=1)
exten = s,202,SetGlobalVar(MSN=2)
Companity a écrit :
Hi,
we are using asterisk with Junghanns QuadBri and some sip phones. 2
channels are configured in NT mode (ISDN PBX connected, internal ) and
2 channels are connected to the public ISDN network (bri-cpe). We use
Bristuff 0.2.0 RC8C from Junghanns.
When a call comes in from the public phone for a specific extension
(Hotline Number), we initiate a parallelcall to some SIP phones and
also to our PBX through the quadbri and we also do a signalling on one
mobile phone(through the second channel which is connected to the
oublic phone network).
Problem:
The sip phones and the internal phones on the PBX see the number of
the calling party correctly (e.g. 040-987654321). Cause we can´t set a
callerid to the public phone network (to show the calling party
number), we want to show an extension of our numbers on our isdn-bri
(asigned by Carrier, e.g. our numbers are 12345-0 to 12345-99). If we
use our current configuration, everythings works good, execpt the isdn
call to the mobile phone. As calling id it shows 12345 w/o an
extension. We would like to set a specific extension, so that for the
call to the mobile phone it is displayed 12345-88 (so we see that is a
call forwarded from asterisk for a specific extension) and the rest
(SIP an PBX) should display the ID of the calling party (e.g.
040-987654321). Does anybody has an idea ? I tried to set a callerid
in the zapata.conf for the channels dialing the mobile phone with
callerid= 1234588 and also same w/ setcallerid(1234588).
Result is that it´s shown 12345 to the mobile phone.
thanks for help
Andreas
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[Asterisk-Users] Re: Obtaining Cisco Firmware painlessly?
I know it can be a real pain in the butt getting hold of the firmware, so any help in obtaining it relatively fast and painlessly would be much appreciated. Can't help with the Cisco 7910, but I noticed that these two files are floating around on the Gnutella network (Cisco firmware is signed, so you don't have to worry about viruses/trojans -- the phone won't load the code if it's been tampered with). cisco.ip.phone.7970.firmware-sccp-6.0.3sr1.zip cisco.ip.phone.7940.and.7960.firmware-sip-7.1.zip - a ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP header Bandwidth Reduction
chawki hammoud wrote: Hi: Internet Bandwidth in my country is expensive so I am trying to figure out a way to use the most of what I have. All the calls are between two servers only. How can I reduce the ip header bandwidth to the minimuim whether I am making one call or multiple calls? Send the calls between the servers using IAX and trunk the calls. This can be achieved by putting trunk=yes in the definition in iax.conf for each server. The more calls you make the less header will come into the equation as trunking will pack multiple calls into the same packet. Combine this with a well compressed codec and you should be right! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Trouble getting a SIP phone to dial out through TE100P
Nick Crocker wrote:
We have a test asterisk box setup and can call each other on our sip
phones and receive calls in on the PRI to our phones no problem. Our
problem is getting asterisk to allow us to dial out using our PRI.
Digium has instructed us that we need to strip the leading 9 from the
digits dialed. Does anyone having a config that I might glean some lines
from to see where we are going wrong.
1. Don't post multiple times.
2. Don't post HTML
3. You should have a line which looks something like this:
exten = _9X.,1,Dial(Zap/g1/${EXTEN})
to remove the 9 when you dial simply change it to:
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
to remove the first digit.
--
Cheers,
Matt Riddell
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