[asterisk-users] Android Phones ;-)
FWIW, just received an android-based phone and after installing sipdroid found that it works very well with asterisk ;). It automatically dials numbers through asterisk if available and otherwise through the gsm network. Contacts integrate well too. No ties to any telco or to google, just a happy user ;) Conrad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snom mass deploy help
I fail to see how this script is useful in order to use Snom's Plug'n'play config. Who said it does? The Topic is snom mass deploy - not Plug'n'play config. It does not use snoms Plug'n'play config, but it still provides for snom mass deploy using the phones' built-in dhcp/http mechanism. Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snom mass deploy help
On Thu, 2009-06-18 at 14:21 +0200, Philipp Kempgen wrote: On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote: I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. FWIW I use a home-grown cgi script to configure the mass-deploy. (attached) Conrad snom Description: Perl program ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64bit: any problems with asterisk?
On Sat, 2009-04-25 at 06:03 -0400, sean darcy wrote: We're getting a new server. I'm considering installing 64bit fedora rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any issues we should expect? FWIW I am using 64Bit Debian all the time - works like a charm. Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anonymous callerid
On Sat, 2008-11-29 at 11:26 -0600, Tilghman Lesher wrote: On Friday 28 November 2008 08:17:24 Philipp Kempgen wrote: Max Alex schrieb: I have one issue regarding override callerid when i have anonymous call. I have added PAI in sip header and also set sendrpid = yes in sip.conf but the callerid is not overriding while i am sending call to three digit calling like 911. Perhaps you need to use SetCallerPres(allowed) ? Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT Astricon/Digium Beach Ball Mailing
On Sat, 2008-07-19 at 03:40 -0400, Alex Balashov wrote: Steve Totaro wrote: Just an FYI for Digium. I received a mailing today from you guys which was nice. The price of mailing was ~$1.60 and inside was an inflatable beach ball. Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. I wonder if the sick heat had anything to do with it, was mine just bad, or should Digium get a refund from the promotion company for providing garbage? Anyone else get one? Was it OK or junk? I didn't get one? Where do I sign up to receive these balls (preferrably working ones) and pens? I keep buying digium stuff already ;) Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dimensioning
On Wed, 2008-07-09 at 10:17 +0200, voip crazy wrote: Maybe 400 calls at one time. By the momento there aren`t voip trunks maybe in the future. [snip] I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? If most of the calls are SIP-to-SIP you could have them reinvite and thus the load on the asterisk server is minimal. Unless I am mistaken and there *is* some way to run 400 simultaneous calls over 2 PRIs... Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK FXO hangup detection with a twist
If we accept a call originated elsewhere, then we cannot hang it up. Only the call originator seems to be able to do that. The upshot is that if asterisk hangs-up a line, and then tries to re-use it for an outbound call before the remote has disconnected, we are simply re-connected to the original caller, and start to play DTMF at them! Has anyone experienced this before? Anyone found a solution? Yes I have seen that with many analogue lines in the UK. This behaviour is somewhat 'by design' - it occurs even if you just use 2 plain old telephones (and no asterisk). I always forced it on-hook for (I think it was) 15 seconds before attempting a new call on the same line, but I always had spare lines to dial out on, so no real need to dig deeper into this ;-) Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Star Wars Echo Sound
them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. I had a similar experience where people claimed it sounded like a 'Dalek' (yes, in UK). The sound is somewhat similar to Darth Vader, I suppose. This was down to a buggy Echocancellation/Silence Detection implementation in the softphone (iaxcomm). Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_sms and smsq in germany
On Mon, 2008-03-24 at 15:03 +0100, Tobias Wolf wrote: Hi, i've been trying to get fixed line sms working for some time now. Can anybody tell me, if he is actualy using this with asterisk in germany? I _was_ using with Deutsche Telekom (dialing 0193010), but the message delivery was so darn unreliable (sometimes it took several DAYS before SMS were forwarded on, and then they all went in a bunch. yak). However, if you want to try, you might be able to dial 0193010 on your DOKOM pri to transmit sms. Nowadays, for transmission of sms, I use kapow (http://www.kapow.co.uk) which allows you to set the sender-number to a german number if you register. I also use sipgate (http://www.sipgate.de) scripts. They also work very well. For receiving sms on my landline, I still use telekom, but as I said, the service is so lousy I don't rely on it anymore. Just didn't find anything better in Germany yet ;( If you want more details of this setup, let me know, happy to provide. Conrad I have followed the instructions found on voip-info. I was successful a couple of years ago with asterisk 1.0.7 and an normal telekom isdn line. Now i want fixed line sms over an Dokom PRI with Asterisk 1.2.9. Here in Germany the Materna AnnyWay Service has to be used, its service number has changed to 09003266900 some time ago. As i understand it you have to send an SMS first, before actually able to receive SMS, so i tried to use smsq with these parameters: smsq --queue SMS-outgoing --motx-channel Zap/g1/09003266900 --motx-callerid XXX --da YYY -m Dies ist eine Testnachricht XXX = The Number from which the sms should be originating and to which i want to receive SMS YYY = The Number i want an SMS deliviered After issuing the command an call file is generated and asterisk dials, but the call retries 10 times without transmitting anything. For a try i set the motx-channel to a phone of mine a listened to the audio that gets send. Now i am curious if i am still make something wrong, if i should contact my telekom provider or Marterna. Has anybody some hint for me? If i get it working stable i will update the wiki page ob voip-info Regards, Tobias Wolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage
I second Sun and supermicro. Sun was really cool on the management facilities, the linux compatibility and the speed was nice too. Supermicro (opteron series) always amazes me how fast they are. They really *feel* fast ;) Only ever used support on supermicro and it was excellent. My box froze and after I send them the opteron built-in exception log they identified a problem with one of the DIMMs, told me which one and sent me a replacement. No fuss with the old one either, I threw it away. Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco phone 79xx get database information
On Sat, 2008-02-09 at 12:51 -0500, Doug Lytle wrote: Javier Temponi wrote: Hi, may be this question is a bit silly, but I couldn’t find any document or post or anything that say that if this is possible or not. I want to show information on my phones cisco 7960/40 when a call arrive. May be a bit more than a caller ID, show more detail level, if is that possible. I already have an asterisk and the phones registered there, and I need to show on the phone display, when the call is ringing, the customer information.. Something like this: source number: xx Customer Number: x Name of Customer: x You may want to check How about doing something like that in your dialplan? Set(CALLERID(name)=source number: \nCustomer Number: \nName...)) I know the ciscos display the callerid-name, so you could put all the information in the callerid-name field and the phone will display it as-is. It might take a bit to make it look pretty and it might affect your CDR as well. But it might also make your microwave shrink, who knows? ;) Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private IP-Adresses of Int1/2
Asterisk can't give out a public IP-address for Int1/2. Where
would it get one from?
Correct that it doesn't. But some kind sould could indeed code a
variety of techniques to get it, such as:
Again: My Problem is not Intern-to-Extern (NAT,Stun). My Problem is
Extern-to-Extern, that the external phones are not talking RTP
*directly* to each other. This is bad, when Asterisk is in Europe and
the Phones are in Asia.
___
Just an idea, it's completely unverified and if I missed the point
somewhat, please excuse me ;) - but maybe this approach leads to the OPs
desired result if thought through further.
I wonder wether some clever dialplan constructs couldn't help.
I'm thinking along the lines of:
[globals]
ALLOWDIRECT=
REMAININPATH=tTwW
[internalphones] - registration context of internal phones
exten = extern1,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = extern2,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = intern1,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
exten = intern2,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
[externalphones] - registration context of internal phones
exten = intern1,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = intern2,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = extern1,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
exten = extern2,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
This assumes all sip phones are set to reinvite=yes.
I expect (one of) the options to dial (tTw or W) to force asterisk to
remain in the media path. This way *only* if it's int-int or ext-ext
will it send sip reinvite, right?
I have another idea just now, but that's even weirder ... ;)
Conrad
___
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Re: [asterisk-users] Re: NAT: RTP Path Optimization
On Wed, 2007-01-31 at 08:42 -0500, Andrew Kohlsmith wrote: On Wednesday 31 January 2007 8:28 am, Conrad Wood wrote: This assumes all sip phones are set to reinvite=yes. I expect (one of) the options to dial (tTw or W) to force asterisk to remain in the media path. This way *only* if it's int-int or ext-ext will it send sip reinvite, right? Yes, but now you have to be careful of unintended consequences when people are trying to use IVRs. I wouldn't take it live as is without further testing, but I guess the idea was worth adding to the thread. What's wrong with having two peers, one with canreinvite=no, and Dial() using the appropriate one? (I haven't been following the thread, so this may have already been discussed, and discounted.) That was the other idea I had - but think different Dial parameters are less problematic. For 2 peers per phone, the phones either need to have a static public IP or need to be able to register with 2 credentials (e.g. snom/cisco). Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] more than 32 callgroups pickupgroups
On Thu, 2006-12-21 at 12:07 -0700, Douglas Garstang wrote: I'm no C programmer, but is this 32 limit just an array definition somewhere? Wouldn't it be a no brainer to track it down and increase it so some very large number? I think pickupgroup is defined as 'unsigned int' somewhere in channels.h. 32 is the number of bits in a 4-byte integer, so it's probably using a bitmask to define which pickupgroups a channel belongs to. I suppose if you are on a 64bit machine/os you /could/ try to make it a 64 bit pointer, but you should really check the source a bit more to see how exactly it's accessed (I didn't!) I don't know any .32bit integers on 32bit machines. -Original Message- From: John Harragin [mailto:[EMAIL PROTECTED] Sent: Thursday, December 21, 2006 11:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] more than 32 callgroups pickupgroups callgroups pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] record time with phones option buttons
On Wed, 2006-12-13 at 12:28 -0500, Matt Van Alst wrote:
Anyone able to point me the right direction for the following would be
helpful.
[..]
Say we have Cisco 7940’s or 7960’s or any phone that has the
additional buttons other than call appearance. Can we program those
buttons to start recording that reps time to the correct division.
FWIW here's an idea:
My first shot at it would be to try it with a snom phone and it's
programmable buttons.
In theory (haven't tried) you should be able to program the snom phone
to send some digits via it's buttons. These digits can match those in
features.conf which in can run some AGI (without interrupting the call
itself).
Example:
snom phone button #1 sends **601,
features.conf:
[applicationmap]
productone = **601,Macro,logproduct,1
producttwo = **602,Macro,logproduct,2
productthree = **603,Macro,logproduct,3
[macro-logproduct]
exten = s,1,Agi(logproduct,${ARG1})
or so..
of course this can be a pain to handcode should you have loads and loads
of products, but it might work.
Let me know!
Conrad
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Re: [asterisk-users] agi scripts running slowly
On Thu, 2006-12-14 at 13:28 +, Richard Smith wrote: Hi all, I recently installed asterisk 1.2.4 on a HP DL140 G2 server and co-located it. My only problem with the box is that there is a noticeable delay in the processing of agi scripts compared to any other install of asterisk I have. Has anyone got any ideas why this is happening and any guide to tweaking the agi to run faster? did your harddrive spin down and the delay is caused by the time it takes to wake up? ;-) More seriously, is it taking long to start the agi or is the agi itself running slowly? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of VPNs
On Tue, 2006-11-28 at 07:18 -0500, Barry Fawthrop wrote: Hi all Is the use of a VPN between IP-PBX and VoIP Provider a useful tool? Since the QoS and general traffic of the Internet can never be predicted, would the implementation of a VPN between Client and VoIP Provider increase voice quality and/or security or is the converse true ? if you do it right then yes, you'll get extra security. Voice quality is likely to degrade due to the added overhead. However, some people reported ISPs mucking about with VoIP traffic and encapsulating that into say IPSec or OpenVPN traffic should help. I run VoIP traffic over OpenBSD IPSec connections quite happily (in UK, using Mistral/Netkonect DSL) and are confident they don't get intercepted or abused. I encapsulate on a seperate box, not on the asterisk box directly to avoid cpu spikes and to be able to shape QoS in my VPN. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] about voicemail setting
On Wed, 2006-11-22 at 18:17 +0800, rilawich ango wrote: As I know, the voicemail will be sent using localhost smtp. I want to use another smtp server for sending voicemail to the users. Is it possible to set it, where to set it? ___ it does not use smtp. If it did use smtp it would need to handle errors and queuing in app_voicemail. It pipes it to /usr/sbin/sendmail -t. I use exim (which installs a link /usr/sbin/sendmail) and tell exim to route it to another mailserver via smtp. So you need to configure whatever mta you have installed on your system to route it to the other smtp server. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording outbound analog calls with X100P
On Thu, 2006-11-16 at 08:29 +1300, Hadley Rich wrote: On Thursday 16 November 2006 06:44, Conrad Wood wrote: On Thursday 16 November 2006 06:42, Matthew J. Roth wrote: As per ManxPower at #asterisk, it is not possible to record a call dialed from an analog phone connected to the Phone In port of an X100P because the two ports on the card are hard-wired together. A bit off-topic maybe, but does that then mean you can't make 2 simultaneous calls through the card? E.g. 1. Call: pstn-phone - asterisk - sip... 2. Call: sip-phone - asterisk - pstn... As he said above, the ports are wired together. There is no FXS device on that card. ah. fair enough ;) thanks for clarifying. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting the CallerID
Out telco assigned us a number range, say from 0231 - 555 to 0231 - 555 . These are number wich are routed to our asterisk server. This will make 0231 - 555 my base number, sorry if i have chosen a name with more an one definition. On the other hand, i can set whatever number i like for outgoing calls. Usually i have something like this in my dialplan: Set(CALLERID(number)=X), where is the number i want to be shown. If i dial a national number (landline or mobile) the number i have set is shown properly. If i dial a number in England the called sees my base number, but in correct international format, +49 231 555 . My Provider i called Dokom, it is a small local telco. I think they only have lines in Dortmund, and beyond that they route through German Telecom lines, but i am not sure of this. The Number I call is a mobile number from Manx Telecom (Isle of Man). I have made the calls with pri intense debug set on, but from my side a can see no difference between calls where the number is set successfully and calls the my number does not show up. I have spoken with Dokom too, and they tell me that they don't mess up with my calls, only passing through, and that they are not sure if the number i have set, will be able to show up at the other side. I have also tried to set number within my assigned number range and outside of my number range. Here the same: To national calls i can set whatever i like and to international calls only my base number is shown. I guess that Dokom bought or rented or whatever a certain range of numbers from 'Big German Telco GmbH'. As long as you remain within the Dokom network you (dokom) can retain the callerid. As soon as you route your call out to another network, the peer will see that the callerid doesn't match the account and reset it. If your call gets routed through various networks I guess any network could decide to reset the callerid. Here in UK any DDIs are not automatically added to the list of allowed out callerids unless you specifically ask, so they get reset to the base number as well. Now, that's only a *guess*. And the bad news is, *if* I'm right, there is little you can do except change provider, route or persuade dokom to speak to their upstream peers. You might have more luck asking on a telecom specific list, rather than here. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] A question on ISDN cards... (in the UK)
On Wed, 2006-11-15 at 11:23 +, Senad Jordanovic wrote: Would anyone like to recommend a good and reasonable quality ISDN card for use in the UK, as after a lot of good results with TDM400P cards with several systems installed now, I need to look at a few ISDN BRI (old business highway about to move to ISDN2) and possibly a single-line PRI (ISDN-30) system. I'd say the most important distinction is the choice between HFC based ISDN cards (starting around £9) and active cards, like Diva, Digium etc (~£300). Whilst the HFC cards work (with bristuff) you need to be prepared to reload modules regularly and go through other hoops. I used to work in a IT company, and there it's perfectly allright to use cheap cards, because the skills to reset modules etc are available at all times. For a clients' system I wouldn't go down that route and spend the money. I have no complains on call quality or dropped calls on cheap cards nor on expensive cards, it's the administration and 'shinyness of the product'. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] some questions about atxfer usage
On Wed, 2006-11-15 at 13:54 +0100, Antonio Almodóvar wrote: Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the transferer. Is there any possibility to recover the call before the timeout of 15 seconds expires? I just press * to retrieve the caller again - Have you tried that? I mean, I would like to personalize the way of making transfers using the feature of atxfer. How can I do that? anything in particular? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting the CallerID
On Wed, 2006-11-15 at 17:50 +0100, Tobias Wolf wrote: Hi, I have some trouble with setting my CallerID if i make an international Call. No Problems with National Calls, i can set whatever I want. We pay for this service but our telephone provider was not able to state clear, wether the number we set on an international call should be shown on the other side. Actually only our base number shows up. With that, do you mean the number without international prefix? That's odd. What's the provider you are calling here in uk? I regular get calls from Germany with the correct international callerid showing. If I understand it correctly, in every call the base number is embedded as ANI, so that we can be billed. Is it possible, that, if a calls goes international, they only refer to the ANI and forget the set number ? The only route I am trying is from Germany to England, maybe it is a problem between the providers and not of my setup. I don't receive any callerid from some (cheaper) german telco providers (but most work correctly) I vaguely remember that there are some pretty dodgy contractual agreements lingering around. You might want to google for 'europe callerid', specificially [1]. Are you sure your calls goes straight from Germany to UK? I found that many German telecom providers terminate through the US, particularly to landlines. Maybe you could route your call to via sip to a uk voip provider and persuade them to set the callerid to whatever you have in Germany? or simply get a UK number routed via sip ? ;) [1] http://www.ainslie.org.uk/callerid/cli_faq.htm In the UK, Oftel will allow European Caller ID if the other country has implemented the Telecoms Privacy Directive ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording outbound analog calls with X100P
As per ManxPower at #asterisk, it is not possible to record a call dialed from an analog phone connected to the Phone In port of an X100P because the two ports on the card are hard-wired together. A bit off-topic maybe, but does that then mean you can't make 2 simultaneous calls through the card? E.g. 1. Call: pstn-phone - asterisk - sip... 2. Call: sip-phone - asterisk - pstn... Because if you could, you could try some trickery with Meetme or Local channels but it sounds like a pretty big limitation of the card, right? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - big installation
On Wed, 2006-11-15 at 14:41 +0100, doki_cti wrote: Hello I want build big asterisk server. Server will be work as gateway between PSTN and VoIP network. I think about 1000 SIP users accounts and 4 E1 ports. I know that preformance in this case depend on codeck which will be use. I want use card with CAPI interface. Can you describe me your experience with this? If you have some big installaion, please wriete some info about server (procesor, ram etc), numbers of user and simultaneous calls beetwene VoIP and PSTN. How often server crash? Regards Doki _ why capi? why not a sangoma or digium card? If you have a 1000 Sip users you'll need to do more than just a 'big' server - google and browse this list, there are plenty of people who published their server hardware specifications and call lists. Have you looked at sip proxying? Maybe use multiple smallish servers? The server should not crash more than you want it to, whatever that is. If you plan to run it non-stop for say 4 years, plan it out accordingly and it can be done. If you run multiple small ones, maybe you can accept 1 crashed server every 3 months. Unless you have lots of time and patience, your best next step is to ask on the -biz list for someone to help with this installation. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VLANs and Quality
On Wed, 2006-11-08 at 10:15 -0500, Barry Fawthrop wrote: Hi all How much does configuring a network with VLANs improve or effect quality ? Is there much reason to justify the configuration of VLANs ( I know networking, but not VLANs at all) Would it not be better to find high traffic users and determine why? That goes for any network ;-). What you _really_ don't want is some guy uploading the lastest holiday movie to your fileserver and bringing down your entire companies' telephone system. However, you might care less about your fileserver running slow during that time. Or: someone plugs in a Apple Laptop with DHCP server enabled and your phones suddenly all get new IP-Addresses. So, you essentially build 2 seperate networks with (almost) seperate levels of bandwidth available. I prefer to use a seperate switch(es) for phones than for data but settle for vlans. I might even use multiple vlans for phones and multiple vlans for data, depending on topology and usage. So I say: yes - there is reason for configuring vlans. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Out Dial Interface for Asterisk
On Thu, 2006-11-02 at 12:31 -0600, Shawn Kelley wrote: Hi All, I sent this a while back but never received any replies. My deadline is fast approaching so I thought I'd throw it out there again in hope of some advice. I need the ability to automatically out-dial and play a dynamically generated message. I then need the ability for the answering party to give feedback via touch tone. I am a .Net Programmer and I have looked at the Asterisk.NET examples, but all I see there is creating calls and sending them to system phones, etc. I don't see anyway of capturing responses back from the answering party, or how to play dynamically generated messages. Does anyone know if this is possible with the Asterisk.NET interface? Or does anyone know of another way to accomplish my needs? I'd drop a call file into asterisks spool dir: call file begin --- Channel: Zap/g1/phonenumber Context: playbackmenu Extension: main Priority: 1 call file end --- in extensions.conf: [playbackmenu] exten = main,1,Background(your-announcement) exten = 1,1,NoOp(User Pressed 1) exten = 2,1,NoOp(User Pressed 2) exten = 3,1,NoOp(User Pressed 3) ... Is that all you need? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API - Originate Call - Need Help
On Wed, 2006-11-01 at 03:23 -0800, Ehsan Khosrowshahi wrote: Hi all, How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command : Channel = SIP/0041435215301 Context = default Exten = 00982166501553 Priority = 1 CallerID = 0041435215301 this works with out any problems I initiate a call from one of my network sip clients (0041435215301) and call someone at anyside of the world, but Can I initiate a call from (00982166501553) to one of my sip users? Why not do this: Channel = ZAP/g1/00982166501553 Context = default Exten = whateveryoursipphoneis priority =1 CallerID = whateveryouwant If you don't have an extension for your sip phone, add this in context default: exten = whateveryoursipphoneis,1,SIP/SIP/0041435215301 Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ${CALLERIDNUM}
On Wed, 2006-11-01 at 11:53 +, Scott Pinhorne wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the
phones ring one way if it is more than 3 digits it rings another i.e.
internal calls and external calls.
exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5)
I'm sure on the wiki (http://voip-info.org) is a list of functions, including
one to
determine length of strings, but you could also do something like:
exten = ,1,Goto(${CALLERIDNUM},1)
exten = _XXX,1,dostuffwith3digits
exten = _.,1,dostuffwithmorethan3digits
Conrad
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Re: [asterisk-users] Server Recommendations
On Tue, 2006-10-31 at 13:52 -0500, Carlos Rojas wrote: Hello, I'm working with supermicro servers, for the irq problems with Dell, any people have problems I second the supermicro servers - particularly the opteron range based on Serverworks HS1000 chipset. Excellent stuff. Well designed, no irq problems and no timing problems. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom or Cisco Phones?
On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote: Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia cars Not really. Both are very good phones. * My Clients prefer cisco because it looks more business-like. - The new snom phones do look better though and the side car rules. * The Cisco phone 'feels' very good in your hand, and the voicequality is superb. (I'd say slightly better than that of the snom 360) * Technically, I find the snom phone more advanced and I can do more cool stuff with it - Cisco doesn't seem to like giving features away in SIP. * Snom phones, for example, have freely programmable buttons that can park/retrieve/transfer calls, show line status etc. I can't get that to work with Cisco phones at all. * Putting custom ringtones (and choosing which ones to use) is a no-brainer with snoms and real trouble with ciscos. * On ciscos, I find the upgrade path from sccp to sip a totally unnecessary annoyance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] porting numbers in UK telewest/bt/adept
On 29 Oct 2006, at 11:02, Matthew Thompson wrote: On 26 Oct 2006, at 11:59, Conrad Wood wrote: A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. What is the old PBX and how are Telewest presenting? We had Telewest lines once and they were the same RJ-45 ISDN 30 as BT. Would it not be possible to use a dual port card and use Adept for the outgoing and Telewest for the incoming service? Ah - I forgot to mention that there are 2 offices involved. The client moved to new premises and the telewest lines are in the old office, Adept in the new office. Otherwise I would do as you suggest, yes. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No zap* commands?
On 29 Oct 2006, at 20:24, Jim Lynch wrote: I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? zap modules not loaded? try: load chan_zap.so on the console and/or put that into modules.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] porting numbers in UK telewest/bt/adept
A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. On the adept line I got a digium card in an opteron supermicro server. ztcfg gives me over 99.99 pretty much all the time. Client complains about crackling, echo and faint on *some* of the calls, mostly incoming. I think the old pbx converts the calls into analogue and back into digital and that's where a lot of trouble comes from. Does anyone have any ideas on a) how to get the numbers ported and from who to whom? b) does adept really resell bt or have they got their own equipment? Are numbers ported to adept or to bt? c) are there any other ideas on how one could cut out the old pbx? Any experts on porting numbers in the uk here? ;-) Thank you, Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ECHO Cancellation in SIP Calls
On Thu, 2006-10-26 at 12:18 +0200, Stefan Agethen wrote: Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS Can i control the cancellation with the zapata.conf ? The snom phones are pretty decent devices and shouldn't introduce echo. Your latency might be too high between asterisk + voipprovider introducing a delay that is noticed as echo. You are hearing the echo that is introduced on the callers side. As I understand it, when you are calling someone there is no zap involved and thus you can't cancel it with zapata.conf. if you look at voip-info.org [1] you'll find a good explanation why you can't use an echo canceller to cancel that sort of echo. So, check the path between you and the voipprovider, e.g. connection saturation, ping times etc (This is assuming you have a proper lan connection between asterisk/snom) Conrad [1] http://www.voip-info.org/wiki/index.php?page=Asterisk+Echo +Cancellation ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP v IAX2
On Thu, 2006-10-26 at 06:51 -0400, Al Bochter wrote: Lets talk about SIP and IAX2 1. The good and bad of both 2. What is the better one and why 3. and any other information that maybe use full like this? http://www.voip-info.org/wiki-IAX+versus+SIP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ECHO Cancellation in SIP Calls
I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS Can i control the cancellation with the zapata.conf ? Sure, but only for Zaptel channels. Not for mISDN channels. If you use ZapBRI, this would be the place to configure echo cancelling. The important bit is: I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people[...] If *he* calls other people he's not going to use zap or misdn, as far as he is concerned it is SIP all the way. (the voip provider puts it into pots) Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] porting numbers in UK telewest/bt/adept
On Thu, 2006-10-26 at 15:40 +0100, Tim Panton wrote: On 26 Oct 2006, at 11:59, Conrad Wood wrote: A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. On the adept line I got a digium card in an opteron supermicro server. ztcfg gives me over 99.99 pretty much all the time. Client complains about crackling, echo and faint on *some* of the calls, mostly incoming. I think the old pbx converts the calls into analogue and back into digital and that's where a lot of trouble comes from. Does anyone have any ideas on a) how to get the numbers ported and from who to whom? b) does adept really resell bt or have they got their own equipment? Are numbers ported to adept or to bt? c) are there any other ideas on how one could cut out the old pbx? Any experts on porting numbers in the uk here? ;-) Yep, it is your legal _right_ to have the numbers ported in a reasonable time/cost. Point this out to them and ask what the complaints escalation procedure is. That should get their attention. I admit that moving twice is a bit unusual, but I successfully moved some BT basic rate numbers from BT to NTL (who now own telewest). It took a couple of weeks and a few faxes. Hm, well according to ofcom the numbers were originally given to BT. Would it then be BT who ports them back to themselves (or to adept)? Or do I hassle Telewest? thank you for your time! Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choice of soundfile format
On Wed, 2006-10-25 at 11:24 +0200, Jon Schøpzinsky wrote: Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? The one that is encoded in the same codec as the codec of the channel. On zap it's often alaw or ulaw so you can encode your files like that. You can encode the same file with different codecs and save it with different extensions (matching the codec) and asterisk will pick the most suitable one. If the channel is gsm, a gsm encoded file would be most efficient, as it doesn't need transcoding at all. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choice of soundfile format
On Wed, 2006-10-25 at 12:15 -0400, Noah Miller wrote: What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? When you get down to it, the asterisk native format is slinear. Fortunately, you're in luck as Kristian Kielhofner did the asterisk community a big favor and had Alison re-record all the asterisk sounds, and he put them in slinear format. You can find them on the astlinux website (there's other formats, too): http://www.astlinux.org/index.php?option=com_contenttask=viewid=38Itemid=43 - Noah As I understand it, if you have a channel that has a given codec the least amount of cpu power is required if the voiceprompt is recorded in that same codec because then asterisk doesn't transcode. Slinear is good, because you can re-encode them without loss of quality. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
On Tue, 2006-10-24 at 12:05 +0100, Faris Raouf wrote: Do any *UK* users have an SPA3102 (the newer version of the SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call has hung up? I've read everything I can find, including an SPA3000 UK setup PDF that lists UK ring etc tone settings, port impedances, disconnect tone settings and so on, but I'm still not getting PSTN hangup detection to work. I got an spa-3000 that works perfectly well now. (UK) I had some trouble at first though. What firmware are you using and what's the symptom? does it not hang up or does it hang up during calls? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
It is brand new so I assume the firmware is the latest?: Software Version: 3.2.6(GWa) Hardware Version: 1.1.5. It just doesn't detect real hangups at all. If the person calling hangs up, either before and after the call is answered, the SPA will eventually timeout after about 30 seconds and hang up - in other words it does not detect the disconnect tone like it should. Interesting, I had it the other way round. It detected the disconnect tone during conversations. I disabled disconnect tone detection. *I think* it detects a polarity reversal instead. (It's been a while and once it worked I forgot about it) I posted my settings here http://www.conradwood.net/sipura.pdf Are they any different from yours? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
You have polarity reversal detection and I do not (I did try with it on, but it didn't help even though there I have measured a polarity reversal on disconnect) FWIW: I once had a nasty DSL filter that broke polarity reversal detection. You have 3ms On hook speed, I have less than 5ms. You have Line In Use Voltage 30 and I have 25. You have Ring Validation 100Ms and I have 256. You have Ring Indication Delay of 256 and I have 0. I had problems with my (old) phone ringing briefly at some stage, so I experimented a little. I will now try your settings to see if it helps with my next big problem --- I'm not getting a CLI number. Instead I get the Username I've allocated to my SPA. ah. Do you have callerid from BT (bt line?). I signed up for something called BT Privacy or so which is free and gives you callerid. If you turn on logging (debug) on the sipura it'll log the received callerid via syslog. Also helpful to check under info Last seen number or so. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
I'm not seeing any caller id in the syslog nor the last seen number thing. (which helpfully just says , :-) I'd be pretty sure that the device doesn't detect the cli. My one does list the number under the 'last seen number thing'. What sort of line is it? Straight BT? telewest? Some converter? I thought your suggestion about the filter was excellent so I tried a few different ones (we are an ISP so I have a few hanging around ;-) ) but to no avail. Thanks. It's unlikely that it would affect cli. It was meant as a possible explanation for the disappearing polarity reversal. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] update_header: Unable to find our position
On Wed, 2006-10-25 at 04:44 +0800, Mark Quitoriano wrote: Hi i got lots of this from the asterisk console what does this mean? format_wav.c:247 update_header: Unable to find our position asterisk console: Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position my first guess would simply be a wav file that's broken. You could try to re-encode it with sox and see if that fixes it. But really, that's more instinct than anything. Haven't looked at that code at all. conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unique call ID's across several systems
On Sat, 2006-10-21 at 19:16 +0100, Julian Lyndon-Smith wrote:
hi guys. Is there anyway of generating a universal / global unique id
from the dialplan (A uuid or guid). I want to have several asterisk
servers sharing a cdr database, and want a unique reference for each
call. Obviously, ${UNIQUEID} doesn't work across several * systems/
couldn't you set a variable in the local dialplan and combine ith with
uniqueid?
e.g.
LOCALID=asterisk1
and then
Set(GLOBALID=${LOCALID}-${UNIQUEID}) or so?
or even use the hostname of the machine?
Conrad
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Re: [asterisk-users] considering purchasing a t1 card, any recommendations?
On Thu, 2006-10-19 at 22:39 +0200, Dovid B wrote: Can I now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. I strongly suspect digium is painfully aware of the problems with some combinations of mboards and their cards, but given limited resources, what would you focus development efforts on? Creating echo-cancellers and bri cards or designing a PCI interface that works even with the most broken motherboard chipset? Most people who have digium cards working seem to have them working extremely well and some can't get them to work at all. Frankly, I have seen so many motherboards that I consider outright broken, I wouldn't blame a digium card for not working in any of those. In fact, those motherboards tend to have problems with most cards, it just happens to show much more when you try and do voice traffic (e.g. isdn) rather than say IP traffic. I'm not qualified to judge digium vs sangoma isdn facing interfaces, but I'd be interested how that compares, if someone can shed some light on that, I'd read it with interest. E.g., how do digium cards perform on substandard isdn lines? How do they handle faults on the line? How compatible (compared to sangoma) are they with different exchange equipment? Last but not least, might that be why digium secured some funding recently, to increase the resources for development on these cards? just my 2 pence worth ;) Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Locking phones at night...
On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. asterisk -rx could also be used. Or a phone menu. Problems with a phone menu: how can you tell the status? asterisk -rx requires access to the asterisk console which throws its own bunch of problems with permissions and scalability. I'd then prefer to code it through the manager interface but that seems like a terrible overkill here ;) How would you use a phone menu for that? That sounds interesting. Our users here like doing phonestuff on their phones rather than on websites etc. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gotoiftime and Macro question
On Wed, 2006-10-18 at 13:39 +0200, [EMAIL PROTECTED] wrote:
Is there a way to run a macro in a GotoIfTime statement ??
from the wiki documentation it seems not, but..
I would like to do something like this:
.
554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?Macro(exten-vm,novm,567))
it does not work, as expected from documentation
any workaround to call an extension WITHOUT vm (also if vm for that
extension is present...) as a consequence of a Time condition?
I presume you can't move the GotoIfTime into the macro itself?
Would something like that work for you? (You might need to doublecheck
the exact syntax!)
exten = 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?novm,567,1)
...
[novm]
exten = _X.,1,Macro(exten-vm,novm,${EXTEN})
...
Otherwise, can you post the rest of your dialplan and/or describe in
more detail what you're trying to achieve?
Conrad
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Re: [asterisk-users] Monitor stops recording midstream?
On Mon, 2006-10-16 at 08:00 -0500, Tim Connolly wrote: Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686 running Linux on 2006-06-17 When I used monitor, I seem to get most calls cut off if they run very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any ideas what might kill the recording process? I'm beginning to wonder if soxmix is truncating the file when it blends the in/outbound streams together due to bad data or something. Same here. I have kept the original wav files from asterisk to check wether it is sox that truncates it and can confirm it is not sox but asterisk cutting them off. There also appears to be a timing problem, because left+right channel become out-of-sync. This is even in the wav files created by asterisk - before I run them through sox. I didn't spend more time on it because it's only a nuisance not a big problem for us, but now at least you know you're not alone ;) Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording from a script
On Tue, 2006-10-17 at 16:00 +1000, Nikolai Lusan wrote:
Greetings,
I have been asked to provide a one off solution for someone. They would
like to take a message left on a remote voicemail system (with their
mobile phone provider) and get it to a wav/mp3 file. There is a number I
can call from my Asterisk system that would allow playback of the
message, but it would require sending some DTMF tones to do it
(traversal of the remote IVR on the voicemail system) I would then have
to record the resulting message (even if I can just use record() and get
it to GSM I can the transcode it to what I want).
In short I would like to know:
a) if this is actually possible
b) if anyone can give me some pointers on how I might go about
automating this with a script.
Thanks in advance.
My first shot at it would be
exten = getmsg,1,Monitor(wav,/tmp/msgdir/${UNIQUEID},m)
exten = getmsg,2,Dial(${VOICEMAILNUMBER}www${VOICEMAILDTMF})
I think there was an option for Dial() to hangup after some time
elapsed, but ideally you would detect end-of-voicemail tone if there is
one.
Conrad
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Re: [asterisk-users] Why the MusicOnHold sound so soft?
On Tue, 2006-10-17 at 17:18 +0800, Xue Liangliang wrote: My MusicOnHold sound is very soft, but when I hear it directly from mp3 playe on desktop, the loudness is quite ok. Wonder whether there is any configuration to change the loudness of MusicOnHold. If you play it with mpg123 you can try the -g option. Alternatively you can change the volume of the file(s) itself with sox. I'm not aware of any volume setting to musiconhold. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what hardware and is it possible
On Tue, 2006-10-17 at 20:38 +0700, Ady Wicaksono wrote: Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? did you look at http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out ? what is the correct hardware for this? if you want to do just that, then pretty much any odd box that runs asterisk will do. There is very little required in terms of cpupower or memory. I'd probably choose an embedded system like soekris for that sort of stuff. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Locking phones at night...
On Tue, 2006-10-17 at 10:25 -0500, Carlos Chavez wrote: I have a customer that wants to lock his phone when he goes home at night so no one else can use it. What would be the easiest way to do this? To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR problem
On Tue, 2006-10-17 at 11:12 -0700, Jack Morgan wrote: All, I'm not able to play background files since this morning. I'm seeing this error message in the logs: [Oct 17 10:23:56] WARNING[4572] file.c: File custom/asterisk-prospectus_IVR-main-day does not exist in any format [Oct 17 10:23:56] WARNING[4572] file.c: Unable to open custom/asterisk-prospectus_IVR-main-day (format 0x4 (ulaw)): Permission denied [Oct 17 10:23:56] WARNING[4572] pbx.c: ast_streamfile failed on IAX2/teliax-2 for custom/asterisk-prospectus_IVR-main-day I know the file is there was working last week. I did update some files on the server over the weekend. I built Asterisk from SVN-trunk-r44731. Any help? Well, Permission denied does make it quite obvious ;-) Assuming it runs on linux try 'ps axu|grep asterisk', That will list the user id of the asterisk process, then run chown userid ..custom/asterisk-prospectus_IVR-main-day.* a google for unix file permissions might help too. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Locking phones at night...
On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. Obviously for performance and elegance the astdb is superior. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk upgrade
On Mon, 2006-10-16 at 13:13 +0200, Simone Ruffilli wrote: at the moment (fortunately) i'm not experiencing any kind of particular problem, do you suggest me to upgrade asterisk? #1 sysadmin rule: If it's not broken, just don't fix it. That will get you into trouble when it _does_ break. I rather *test* new versions, fix any configuration problems and then keep the live versions uptodate. It can be quite a nightmare to skip lots of versions, particularly under timepressure with a broken system at hand. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk crash in res_features.c
On my asterisk machines the following features.conf file crashes asterisk (core dump) This happens with 1.2.4, 1.2.10, 1.2.12, with or without bristuff. It's easy to work around, but broken nevertheless. Has anyone else experienced that or is it just me? ;) /etc/asterisk/features.conf [applicationmap] # THIS CRASHES asterisk: rateone = #1,caller,Macro,ratecall,1 # THIS LINE DOES NOT #rateone = #1,caller,Dial,Local/[EMAIL PROTECTED] EOF: /etc/asterisk/features.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building the Perfect Box
I'm sorry. You seem to have fallen into the sar-chasm. And I thought the smiley would be enough hint. :-) never. 2 Smileys: maybe ;) Yes of course. These notebooks tend to get forgotten in a cupboard until the day they're needed. And then they're so out of date that they're more damaging that useful. Do you update the text on the server itself somewhere? How do you go about keeping it up-to-date? Just discipline? Do you work in a team with others? I think it's possible to discipline yourself on this front, yes. How often does an operational *-box change enough that you have to modify your recovery procedures, anyway? They *shouldn't* change too much. But there are currently quite a lot of companies here in the UK calling up clients to persuade them to switch to another provider. If the client doesn't loud and clearly says No thank you they'll be migrated and their number ported. They only find out when someone digs up the road and breaks the PRI. Then they wonder why BT can't help them. ;( I wondered wether you do regular recovery tests. Like a firedrill, or so. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Forcing Transcode
On Thu, 2006-09-28 at 16:54 -0600, Colin Anderson wrote: Erm, I think what the OP was referring to was something like this: ____ _ A. SIP service--B. His Asterisk install-C. His customer's install--- Enduser handsets ____ _ A. uses G.711 B. uses G.711 C. uses G.711 What he wants to do: A. uses G.711 B. transcodes to G.729 C. uses G.729 He can put disallow=all allow=g729 into the sip.conf on B, section for either A or C or both ;) (strictly speaking if it's in both it probably wouldn't transcode though) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building the Perfect Box
1. Good box, see above We used IBM, HP/Compaq and Fujitsu Siemens. None of them came close to supermicro opteron servers. The Serverworks-HT1000 Chipset rocks (apart from the broadcom nic). Things just work and I tell it exactly which IRQs to use for which slot. And boy, do they feel fast when working with. Actual spec isn't that impressive but the whole board is designed so well it easily outperforms any HP, Fujitsu,Dell or IBM I've seen so far. 2. Good LAN - this is so critical and so often overlooked in the day and age of guys crimping their own cables and running $150 switches. You can't do that, and if you do, you do so at your own peril. Managed swiches, professional cable installation. This is not a problem for me since I *am* a professional cable installer but I have actually witnessed people making patch cables with a flat blade screwdriver and a hammer! Did we meet? ;-) I used to do that in the 90s with coax wiring until I finally saw the light ;) And then I wondered why my fluke tester said no. Seriously though, you're quite right decent cabling is _essential_ 3. Tuning of the LAN - VLAN's are good. QoS packets are good. Switches that honor the QoS packets are good. I tend to use a different switch altogether and lock the switch ports (because people do plug weird stuff in which suddenly acts as a dhcp server or does other annoying things) 4. Handset selection - this is another biggie. I've selected Snom 360's, and yes they have warts, but they are feature rich for the price and Snom is snom 360s are definitely the best, but our clients seem to prefer the look and feel of cisco 79xx. USERS INVOLVED. 5. Tuning of Asterisk box itself - this cannot be under emphasized. This is a very important step and tuning methodologies vary according to distro, skill of the admin, and particular circumstances. I've learned *way* more than I ever wanted to about processor affinity sinc I started using Asterisk. I install the box minimalistic to begin with (debian usually). Compiling a preemptive kernel helps too. Stop unnecessary daemons and off it goes. 6. Termination of PSTN. Basically I would never do an Asterisk install where I was forced to do something stupid like aggregate a dozen Centrex lines or some mickey mouse deal with FXO ATA's or whatever except for a hobby or prototype install. PRI, BRI, IAX or SIP, don't mess around with anything else. absolutely. 7. Relationship with provider. What is their SLA? Is it the incumbent or the clec? An incumbent will be more expensive and more difficult to deal with but they will tend to be more reliable. A clec will be cheaper and they will be way more accomodating but you will most likely not get five 9's from them. A VoIP provider should never be trusted, period. You will not get five nines from them, ever. Plan failover situations accordingly. SLA? What's a 'SLA'? :-) Service Level Agreement. Normally it means if your line fails you get £50 or so. Maybe £100. But *never* enough to compensate for the trouble. You have to have a backup plan. Amusingly, a client's * box went down this morning. I didn't get the washout, but the mitigation wasn't well planned either -- everyone with an Asterisk box should know what they're going to do if it falls over, in detail. In a notebook. Just like when the nuclear missles start going. Yes of course. These notebooks tend to get forgotten in a cupboard until the day they're needed. And then they're so out of date that they're more damaging that useful. Do you update the text on the server itself somewhere? How do you go about keeping it up-to-date? Just discipline? Do you work in a team with others? 8. Plan plan plan plan. A good install of ANYTHING is 80% planning 20% doing it. What is your plan when your primary PSTN provider fails? What is your plan if your Asterisk box goes pear shaped? My dialplan can survive either PSTN, WAN or LAN failure (albeit with reduced functionality). I also keep a cold spare, an identically configured box that I can literally throw into the rack, turn it on, plug in the PRI's and no problem. Planning is vital, but do you play through disaster recovery scenarios regularly ? I was musing on giving station users a list of pseudo-CLASS dialcodes they could punch to mark that there was a problem with a previous call, so it would go into the logs and could be checked latter. This is actually quite a cool idea. That's almost a MOS test ;) Press 1-5 to rate the call ;-))) Would be even better if I could do that *during* the call already. Maybe in features.conf...hm. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Backup
On Sun, 2006-09-24 at 16:47 +0100, adebayo omo-dare wrote: I don't know if this may at sometime help mr Wood, but BT, with their ISDN30* actually offer something called Site Assurance - the problem is that it does not automatically fail over, and according to the last memo I read - failover takes about 1 hr. Yes, I was thinking about ISDN2e. With ISDN30 BT gives you more options. On ISDN2e BT only offers Back in Business which essentially means you call them up and they divert the number to somewhere else. Helpful, but not really quick enough. A problem is that, due to outsourcing, product ranges, size issues, etc, a lot of people on BT's frontline are not really keyed up to their product offerings. Who knowns, maybe the failover process has been automated at this point in time. True. We buy PRIs from a wholesale retailer who resell BTs pris. It's technically the same thing and they seem to be quite clued on what the line can do and what it cannot. (Apart from being a fraction of the price). I found the experience dealing directly with BT quite frustrating. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forcing Transcode
On Thu, 2006-09-28 at 14:10 -0700, Mr. Jones wrote: Hi Folks, I'm curious if there's anyway to force Asterisk to transcode for certain handsets. Specifically we have an inbound SIP origination service which uses g711. We're having bandwidth issues with a client and would like to force Asterisk to transcode to g729 until we can get their T1 in place. I'm not entirely sure I understand. Do you want certain sip peers to only ever use g729 (no matter what the other leg of the call uses) ? Then asterisk would (have to) transcode. Have you tried setting disallow=all allow=g729 in the peers' section in sip.conf? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DSL router with integrated SIP proxy?
On 24 Sep 2006, at 13:47, Steve Kennedy wrote: On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote: Does anyone here know of an ADSL router with integrated SIP proxy? I use soekris boxes with openbsd on a flash card and a lot of scripting to gather statistics on all sorts of stuff. works very well too and gives all the stats one can wish for ;) conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Backup
making the call. I guess I could just add the call route to the other campus just below the my default call route. So if the primary call route fails, it will just go to the next line being the other campus. That's precisely what I do with the main route out on ISDN, if that fails, it switches over to various voip providers and even down to a bluetooth enabled mobile ;). it works quite allright for outgoing calls. I believe for incoming calls you need to persuade your isdn supplier do forward the call to ISDN-B if ISDN-A is hosed. Here in UK I couldn't persuade BT to do so yet ;( Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s
2 cents I would not mind paying a reasonable price for a single port BRI but buying a Quad-BRI to get a stable installation is a bit too much for most home installations. Then I will probably start using the old Digital-Analog adapter and use a TDM card. But I don't understand why it shouldn't work with a HFC-S bri card. The way I understand it is that HFC-S cards are quite dumb and a lot of work needs to be done with the main cpu. The ISDN signalling is quite sensitive to timing and if the main cpu is busy it's not going to be happy. It also appears that capi/i4l/misdn have a latency far higher than what is useful for asterisk. It is aimed at general-purpose use, including data transfer. At least, that was my experience. I tried visdn and the results where very promising but it doesn't seem to be quite ready yet. I suspect it's architecture will be much saner than bristuff patches. I also think you get what you pay for and I don't use hfc based isdn cards in production any more. Having said that, a small home installation isn't quite the same as a 30 user office environment. My home-pbx for example is quite happy reloading asterisk+zaphfc every night. Of course not something I'd accept in a production environment, but that's probably not what HFC-s cards are aimed at either, right? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Budgetone phones don't show
On Wed, 2006-09-13 at 10:57 +1000, Paul Hales wrote: From memory, it canalmost I used quite a few Grandstreams on a job a while ago, and my memory says that they will do alpha if you are lucky. If not, you get rubbish. My memory also tells me that UPPER CASE worked better than mixed case. It's actually the same as an old-style calculator display. something like: _ | | - |_| The phone *will* happily display any characters that it can with this combination. (Meaning the firmware has provisions for alphanumeric) e.g. u is fine: |_| and L | |_ etc.. but D is trouble for example. So if you *really* wanted to, I guess, you could translate callernames into a combination of displayable characters before passing it on. Cool hack, but it's probably lots cheaper to buy gxp-2000s ;) Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s
On Mon, 2006-09-11 at 21:14 +0200, Remco Barendse wrote: Hi list! I was using bristuff-0.3.0-PRE-1s with florz patch but where normally the TEI check request message were I was getting errors. Concerned about that I switched to plain vanilla bristuff. Now everything *seems* to be working without errors but I regulary get reports from people trying to call me that they get a signal that the number is not in use or is disconnected. Is anyone else experiencing the same? yep I had the same here with BRIstuffed-0.3.0-PRE-1l it seems to get progressively worse over time. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 Function Keys
2) One digium quadri primary ISDN interface (TE410P) 3) Two Rhino Channel Banks 4) 25 Analogue Phones on every channel bank How I can configure function keys on my SNOM 360 for monitoring analogue phone status? I haven't used the Rhino Channel banks yet, so I'm guessing to some degree here: I'm not exactly sure how you address each phone on the channel bank. Presumably it connects to the digium card. If so, don't you have something like ZAP/1 to dial first phone ZAP/2 to dial second etc? If so, you should be able to add hints to your dialplan for each phone and make the snom monitor those. The snom360 works rather well with hinting and allows you to call/transfer a call to the monitored phone when you press the button too. For example... in the dialplan: exten = 4101,hint,Zap/1 for the functionkey (type destination) put: sip:[EMAIL PROTECTED];user=phone Conrad. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using HINT with Cisco 7940/SIP
On Sat, 2006-06-17 at 15:49 -0500, Lacy Moore - Aspendora wrote: Can't be done using the 7960 with SIP, unless you are talking about just monitoring that phone. You can monitor a 7960, but you can't show the status of other phones on a 7960 with SIP. Do you know wether it can be done with a 7940(SIP)? Can it display status of (for example) 4205,hint,SIP/phone1 ? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a Blue tooth wireless headset that will work with asterisk?
On Mon, 2006-08-28 at 16:24 -0600, Chuck Bunn wrote: Hi, Does anyone know if there is a blue-tooth wireless headset that works with asterisk and/or a SIP software phone on the PC? I use a Motorola HS800 as an alsa device with iaxcomm under Debian GNU/Linux. Works well for me ;). It is presented as an alsa device, so it *should* work with linphone (sip), asterisk and the lot. I have no idea wether that works on windows, but there surely are blue-tooth headsets out there for windows that can do a similar thing. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 160 analogue phones..
On Sun, 2006-03-05 at 16:05 +0200, Tele Cost Price Reducer wrote: Conrad, i would go with following solution: 1. 6 sets of Audio Codes of 24 FXS ports conected by SIP accounts to the system. the type is MP 124. then you open the conector on the initial MDF and then the users have the same phone on their table 2. one dual Xeon system (or even stronger - 2 Dual Core system). such a configuration can take 60 calls at g711. 3. 16 IP phones for the medium up users I quite like the idea of the audio codes MP124 - it was my initial feeling to use something like that. I'll give that a go. Thanks a lot!! to save others some googling I found the product at: http://www.voipsupply.com/product_info.php?cPath=3_26products_id=207 Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with NAT!!!
On Fri, 2006-03-03 at 10:45 +0100, serge messa wrote: Hi all I'm a newbie in asterisk.I install asterisk server successfully. I configure this server to traverse NAT. Using Xlite clients, i make a call between 2 local networks through Internet.Asterisk server is installed on a host with public IP. client A (in the LAn A) and client B (in the LAN B) are registered. When i make a call from the LAN A to the LAn B, everything goes well.But, when i try to make a call from the Lan B to the Lan A, the xlite client B, How do connect Lan A and Lan B to the internet? Do they both have a public static IP or are they dynamically assigned? Are they both the same routers? It might be far off but here's a couple of possible reasons: a) either one LAN keeps changing it's public IP (or just bad timing that the IP of Lan A changed when you tried to place your call to it) b) Either Router (a or b) might not allow the relevant packets through to xlite (or to the internet) Can you give more details on your configuration? Can you provide asterisk logs? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 160 analogue phones..
Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2 ISDNGuards in front of the machines. More to the point: The client has 160 existing analogue telephones which they don't really want to change right now, because a) they are very cheap b) the users don't need to re-train. I have thought of Rhino Channelbanks, but then realised I need to use 7 of them and connect each with a T1. I don't really want to run 7 T1 + the 2 PRIs into one asterisk box for performance reasons. Ideally, several 48-Port SIP-FXS channelbank woulds be ideal I guess ;-). Does such thing exist? Or how do others do this? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
On Fri, 2006-02-24 at 10:54 +1100, David Ankers wrote: Are you sure those switch figures are right? 16ms delay in the switch path sounds a bit long. Cisco's mid-range switches like the 2950 have switching times measured in micro seconds. Then again a 2626 procurve is only around $700. I meant micro-seconds, yes - my apologies. The 26xx series are ok, but I had specifically the 4108 in mind when I said 'good experience'. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mysql problems
On Fri, 2006-02-24 at 09:44 +0800, Ronald Wiplinger wrote: My database machine is broken and I have to use another one. I made somewhere mistake(s) and get now in the debug file: [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query: SELECT * FROM sip_buddies WHERE name = '886' [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query Failed because: Can't find file: './astconf/sip_buddies.frm' (errno: 13) first of, errno 13 is 'permission denied', so I guess your mysql database is running as a user who hasn't got permissions to the file. --- which makes it a question for the mysql mailing list. Anyways, on linux, you can use ps axu to find out as what user mysql runs as. Then change permissions/ownership on the files to match. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
On Sat, 2006-02-25 at 00:21 +1100, David Ankers wrote: Aha, micro seconds in networking terms is normally written usecs or us (actually it's the greek letter mu as in ulaw) rather than ms which are milliseconds seconds - what had me puzzled was that it was stated that this could harm the voice path! The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! There is then only 8 usecs between the two switches, how on earth would this make any difference to the voice path at all? Let alone induce any echo... Obviously the originally poster didn't understand the difference. And based on this, he's probably advising people not to use Netgear switches for voice, oh dear. Agree , previous statement was incorrect and I should probably not post late at night ;-) A few microseconds delay in the path obviously doesn't cause extra echo. Thank you for pointing that out. == Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TFTP server for GrandStream BT phones / need testing
This patch adds only GS BT phones recognition funcionality. tftpd-hpa does not correct handle the OPTION parameters in the TFTP packet ;( At first I tired to implement it into tftpd-hpa, but after debuging the code I give it up. The tftpd-hpa reads the parameters from the TFTP packet as they are in the packet and after each option it do the associated command. The packet from GS BT looks like --- cut --- Opcode: Read Request (1) Source File: boot.bin Type: octet Option: blksize = 1024 Option: tsize = 0 Option: timeout = 4 Option: grandstream_MODEL = BT-100 Option: grandstream_NAT = 1 Option: grandstream_ID = 000b8203e0e9 Option: grandstream_REV_BOOT = 001.000.001.000 Option: grandstream_REV_PHONE = 001.000.006.007 Option: grandstream_REV_VOC = 001.000.001.000 Option: grandstream_REV_HTML = 001.000.000.049 Option: grandstream_REV_RING1 = 001.000.000.000 Option: grandstream_REV_RING2 = 001.000.000.000 Option: grandstream_REV_RING3 = 000.000.000.000 --- cut --- So it reads ... Opcoce - do something Source File - send the file to the client ... Reads the Options from packet. So in the time or sending requested file, there I have no information about the phone. I was too lazy to correct this ;( oh I see, that would make the provisioning work through routers ;-) I use the MAC address to match different phones, which obviously breaks if I need a router between phone and tftpd-server. Thanks for clarifying and well done! Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
Simple formula: 1. Total Revenue 2. % of revenue derived from phone usage 3. =Cost of downtime by using SoHo or consumer gear. It's not a question of if a SoHo or low cost device will screw up, it is a question of when. This is 23 years of experience talking. Where I work, the value of #3 above is $16 Cdn a *second*. We are below 500 employees, so we fall into the SMB segment. Sometimes I'm appalled by statements that a $700 switch or a $400 phone isn't worth it. Huh?? Maybe in Absolutely right! for something as critical as switches cabling I always recommend to spend real money. Don't ever try to save money any equipment that is required to operate the business. (Had very good experience with HP procurves over the last 10 years or so). There is no point buying netgear or other low-cost switches for a business ever. The cost saving of being able to pin-point a cabling/NIC/bandwidth problem down to the port on the switch easily and quickly is wonderful. Combined with SNMP and all the other goodies good switches come with, our clients save a lot of money by paying me less for my time ( d'oh ;-) ). The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! I see no reason however to spend $400 on a single phone though, because if a single phone breaks, it's not going to bring your business to a standstill, is it? (I guess unless you only have one in the first place ;-) ) conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup call on Hold
On Thu, 2006-02-23 at 11:08 -0500, Sean Cook wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Is it possible to pickup a call that is on hold on another extension? Does anyone have any magic they can share on this topic? I am struggling to teach call parking at a local shop where we installed *. It would simplify my life so much if they could just put the call on hold and pick it up on another line. Have you tried the parking application? Depending on what phone you use you might be able to reprogram the hold button. http://www.voip-info.org/wiki-Asterisk+call+parking Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
On Thu, 2006-02-23 at 15:48 -0700, Colin Anderson wrote: The cost saving of being able to pin-point a cabling/NIC/bandwidth problem down to the port on the switch easily and quickly is wonderful We also use 3com NJ-200's which is a 4 port switch in a wall plate that has SNMP and other goodies. I can troubleshoot down to the wall plate, anywhere in the world. Last year I was on holidays in Vancouver (1000K away from the office) and I got the call that an exec couldn't plug his laptop into the wall, no signal, and he was pissed. I whip out my laptop, walk across the street to Starbucks, got a wifi signal, VPN in, I check it out - nope, it's your stupid laptop, PHB-boy. Turns out he disabled the onboard NIC. That single incident, to me justifies the whole expense of a good infrastructure (and to the PHB too - he was spooked that I could do that) ___ I used to have VNC and TopgunSSH on my Palm and an Infred connection to my mobile and from there on to the internet ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming/Outgoing call question
On Thu, 2006-02-23 at 17:53 -0500, Kevin Smith wrote:
Hey everyone,
I have a more of an opinion question then a technical question. The
asterisk server I am setting up is going to host 3 different businesses.
Each business is in the same building, and on the same network. My
question is regarding calls coming in and going out. We are a small ISP
and have a lot of numbers that are forwarded to our phone system. The
other companies have about 3 to 5 numbers going into their offices. My
question is if there is a good way to test for which number and where to
send it to.
Right now my though process was something like this (keep in mind I
haven't wrote it):
[default]
include = Our-Numbers
include = Business1
include = Business2
[Out-Numbers]
exten = s,1,gotoif,$[${EXTEN}=Number1 |
${EXTEN}=Number2..${EXTEN}=NumberN]?Match:1|:
Is that the best way to test for the number that is being dialed? Or can
you recommend a better way. If anyone has done something similar could
you share how you did this type of a setup? I know I could manually put
in each one, but I think there probably is a better way. If I have to go
that route, then I probably will write a script to generate the file.
Thanks,
Kevin
For incoming calls I'd do something like (simplified):
[incoming-calls]
exten = BUSINESS1,1,Dial(SIP/business1)
exten = BUSINESS2,1,Dial(SIP/business2)
...
assuming you got channels that give you that information, e.g. ISDN or
so. Analogue pots won't provide the information which number was
dialled.
Asterisk will match the number that was dialled in the dialplan as the
extension. You might want to put a pattern in in case the number dialled
isn't specifically listed. (e.g. your numbers?)
Conrad
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Re: [Asterisk-Users] What business IP phone to use
1) Budgetones: Don't bother for a business setting. The speaker phone is basically useless (echo problems) and the handset is horrible. If you follow the suggestion on the Wiki to drill out the handset, it improves things marginally, but not much. Users talking to you will constantly complain about you sound muffled. It's think it's a frequency response thing and not a volume thing, I think it's just getting lower than a standard 8 khz sample out of the microphone, because it's so cheap. I wouldn't dismiss the budgetones so easily. We use about 20 budgetones 100/101. We exclusively use firmware 1.0.6.7. All the phones are provisioned centrally via tftp which works really well (almost plug and play, except I got to type in the MAC of a new phone into my script and run the script). I cannot recall a single time a phone 'crashed','froze' or didn't register properly. We tried snom 320 and a telappliant phone[1]. We sent the telappliant phone straight back to the supplier because it was so horrible. We still use the snom for our receptionist, but our users actually prefer the budgetones. Personally I really like the snom 320, but not all users find phones as exciting as me ;-) We have no issues with echo nor complaints about the voice quality. When we introduced the phones (coming from BT analogue phones) users actually commented on the improved clarity of speech. None of the phones broke (and they do get mistreated ;) ) since we started using them, which is about 2 years ago. Here in UK the phone cost less than £50, considerably less if you buy 30+. Of course it's not the right phone for receptionists or phone-power-users or people who rely on a speakerphone, but it's simplicity seems to appeal to some users. Heck, for that price it's worth buying one as a demonstration unit. The headset jack on the back is also a nice feature: If you don't like the headset you can simply plug your earphones in. It might be worth mentioning that we disabled most of the 'features' on the phone itself, like call waiting, transfer etc and instead are handled by asterisk which might explain why our phones don't crash ;-) conrad [1] http://www.voiptalk.org/products/Telappliant+IP2006+SIP+Phone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TFTP server for GrandStream BT phones / need testing
On Wed, 2006-02-22 at 11:37 +0100, Peter Hudec wrote: hi, I known, that this is not * related, but a lot of members of this ML uses GS BT phones. I have patched the atftp serber to recognize the TFTP OPTION, whis these phone send during boot. Patch includes - another locations for configs, firmware and ring tones - different FW versions for phones - custom ring tones for the phones You can find patch, source/unpatched/ and DEB for debian/sarge at http://projects.hudecof.net/linux/atftp/ Does the patch add any functionality to atftp that tftpd-hpa[1] doesn't have? [1] http://packages.debian.org/stable/net/tftpd-hpa ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH from RCA jack?
On Fri, 2006-02-17 at 09:10 -0500, Alexander Lopez wrote: I have not done this but I could probably send you in the right direction. * MOH uses a he standard out of an audio program (ie mpg123) you should be able to add a custom mohtype in the musiconhold.conf file. All you need is to 'play' the audio from the line in on your MB and put it on STDOUT. Otherwise you can 'record' the message via line-in, edit it for length, and convert to MP3. I'm not sure whether asterisk is going to fire up multiple instances or just a single one of the player in your case, but if not you might be able to use sox to take dsp as input and output ulaw on stdout (e.g. define this in musiconhold.conf: sox -t ossdsp /dev/dsp -t ul -). I kind of suspect that you will get locking problems between multiple sox'es too, but that depends Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap BRI card
On Fri, 2006-02-17 at 20:08 +0100, Michiel van Baak wrote: On 16:14, Fri 17 Feb 06, Mimmus wrote: Hi, I'm asking to myself what's the main problem in using cheap BRI cards (30-60Euro, as these HFC-based) vs. great active cards as Eicon DIVA. I run an asterisk box with 2HFC cheap (billion) cards (7.99 GBP ~ 1 Euro?). It's our own phone system (not a clients' site). It works well now, but occassionally it's a bit fiddly. It took me a while to install and get rid of all the echo, kernel panics and segfaults ( I went through isdn4linux, capi, misdn and finally ended up with bristuff). Unless you got a house full of technicians you probably won't save much by buying cheaper ISDN BRI cards in the long term ;) conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?
On Thu, 2006-02-16 at 23:39 +0100, adibar wrote: Hi List Is someone out there using one or more GSMgateway(s) from CyberTelecom ? Me and some friends are interested in buying some of them, but before we would like to ask, how the experiences are others have made. e.g. How easy to setup ? How reliable ? How's the voice quality ? etc. I use the analogue version. it was very easy to setup, essentially plug sim in and go. voice quality is good. Delivery was prompt. one caveat I found though: It doesn't seem to work with T-Mobile in UK. Linus Surgus on asterisk-biz suggested it might only be working on 900Mhz instead of - as advertised - 1800 Mhz and 900 Mhz. I am going to try an orange simcard next, because orange also uses 1800 (like t-mobile). Because of this I cannot comment on the reliabilty yet. -- conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?
On Wed, 2006-02-15 at 08:47 +, Chris Bagnall wrote: I do not even know which brands/models to consider that are out there. Given that we are in the US, and want to use BRI to improve sound quality (no echo, no static), what would be some good cards to look at? I hear a lot about BRIStuff, which I think is used on the Junghanns cards (like the quadBRI PCI ISDN), using the CAPI channel. Are those the Cadillac of ISDN cards? Consensus certainly seems to be the Junghanns cards are amongst the best, but not exactly cheap. If you only need to service 2 BRIs, you might want to look at some of the passive options. We have a number of sites here in the UK running 2 HFC-S based cards in a box, all of which seem quite satisfactory (no echo, etc.). Over here you can pick up HFC-S based cards (the ones we use are these: http://www.solwise.co.uk/isdn.htm) for under £20, so they're probably even cheaper on your side of the pond. Same here, we use 2 MRI HFC-S cards in one box. We use bristuff. We had terrible issues with isdn4linux and capi (admittedly that was a year ago, it might be better now). With bristuff it all works very well. conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] attended call transfer
On Mon, 2006-02-13 at 21:20 -0800, Michael Collins wrote: JCC, So let's consider an operator, takes a call and decides to attended transfer it to Bob because it's slow and she want's to ask something, but the instant she picks that option another call comes in. If hanging up converted it to blind transfer she could get on with her work and answer the next call, as it is she needs to wait till something happens and possibly lose the next call. OK, it's a stretch but it does seem like hanging up the call is just wrong! Absolutely right. I looked at res_features.c and thought maybe I can do a quick fix and invoke app_dial instead of hanging up the channel ;) But that fails because the channel remains locked. I know absolutely nothing about the locks of a channel in asterisk but I'll dig around and in a few years I might be able to fix it ;) Of course if someone has got good recommendations how to properly do it I'm all ears. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug in bristuff?
On Mon, 2006-02-06 at 22:58 +, Conrad Wood wrote: Unqiueid: asterisk-1713-1139266402.909 ^ Please note the spelling of uniqueid. I find the spelling in res_features.c - but only once I patched it with bristuff patches. Does anyone know whether that is a known problem with bristuff? If so is it fixed in a later version? Where do I report a bug in bristuff? ;) For what it's worth: this seems to be fixed in newer version of bristuff... thanks everybody... ;-) Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf number exists in file. How can i do this?
On Wed, 2006-02-08 at 14:37 +0100, Arne Morten Johansen wrote:
Hi there.
I currently have a GotoIf statement that goes to a special
extension priority if the CID match with one of the numbers in
my “list” of CIDs. The way I’ve done it now is by multiple OR
operators. There must be a better way. Anyone got some
suggestions?
This is basicly what I want. “If CID Exists in $File, goto
s,10”. So when I want to add a new CID I just add a new line
in a txt file.
Or, maybe you can use the existence of a file rather then the content of
it?
exten = s,1,System(test -e /var/lib/asterisk/callerids/${CALLERID})
exten = s,2,NoOp(Normal caller)
exten = s,102,NoOp(special caller)
this way you can add callerids by simply
touch /var/lib/asterisk/callerids/phonenumber
does that help?
Conrad
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Re: [Asterisk-Users] bug in bristuff?
On Wed, 2006-02-08 at 08:35 +0100, stoffell wrote: On 2/6/06, Conrad Wood [EMAIL PROTECTED] wrote: Please note the spelling of uniqueid. I find the spelling in res_features.c - but only once I patched it with bristuff patches. Does anyone know whether that is a known problem with bristuff? If so is it fixed in a later version? What version of bristuff are you using? Then I can have a look in my bristuff to see if I have the same problem.. Where do I report a bug in bristuff? ;) Check this website to contact the author of bristuff, http://www.junghanns.net I'm using bristuff-0.3.0-PRE-1.tar.gz. I thought there might be some bugtracker for that too, otherwise I'll email junghanns - thanks.. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bug in bristuff?
Hi everyone, I get these events sent like this: Event: ParkedCall Privilege: call,all Exten: 701 Channel: Zap/4-1 From: IAX2/cnw-4 Timeout: 120 CallerID: X CallerIDName: Conrad Wood Unqiueid: asterisk-1713-1139266402.909 ^ Please note the spelling of uniqueid. I find the spelling in res_features.c - but only once I patched it with bristuff patches. Does anyone know whether that is a known problem with bristuff? If so is it fixed in a later version? Where do I report a bug in bristuff? ;) Thanks, Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] feedback on grandstream budgetone
Hi everyone. I see many posts complaining about Grandstream equipment, so I thought I tell the other side of the story as well. We use Grandstream Budgetone 101 phones in our office and they work extremely well. We have no echo, no crashes, no sudden resets, they just work. We provision them using TFTP. Our asterisk box has 2 HFCPCI ISDN BRI cards. It's running asterisk 1.2.0 on a cheap and very basic fujitsu-siemens workstation. It was working well with asterisk 1.0.1-9 as well. Best of all the budgetone-101 seem to be quite resistant against the usual mistreatment of slamming down the phone, throwing it against the wall, pulling and stretching of cords etc. Kind Regards, Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipura ata 3000 UK (BT) CAllerid
Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] early dial (grandstream bt100)
Hi everyone,
I'm trying to get early dial to work with our grandstream 100 phones.
The phones use SIP, asterisk is 1-0-5 on debian GNU/Linux (sarge).
Outside connections are via 2 ISDN BRI (British Telecom) lines using 2
billion isdn cards
and bristuff.
The phones are set up to be in context [internalphone].
I numbered all the internal extension with _6XX
That works well with early dial. As soon as the 3rd digit is dialed, the
phone connects to the internal extension.
I also have an extension like so:
exten = _9X.,1,Goto(dialout,${EXTEN},1)
[dialout]
exten = _X.,1, magic agi scripts to resolve callerid for billing
purposes
...
... takes about 1 sec to complete
...
exten = _X,7,Dial(Zap/g1/${EXTEN}
I have overlapdial = yes in zapata.conf.
When I dial the 1st digit AFTER 9 it starts to dial and fails because
the number is incomplete.
How do others do early dial? I'm sure I'm missing something bluntly
obvious, but I can't figure out what.
I thought of setting the phones to DTMF inband, so that the digits are
passed to the exchange, but that seems awfully messy.
(btw. I did read the wiki and the SIP-Invite stuff is actually working
on my grandstream firmware)
Any clues, anyone?
Thank you,
Conrad
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Re: [Asterisk-Users] Console ALSA Sound
On Fri, 2005-06-17 at 22:34 +0200, Conrad Beckert wrote: Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference FM is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad Try adding noload = chan_alsa.so to your /etc/asterisk/asterisk.conf. (if required also chan_oss.so) Conrad ___ Asterisk-Users mailing list Asterisk-Users@lists.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] a simple call to my girlfriend
On Thu, 2005-06-02 at 15:32 -0800, Mojo with Horan Company, LLC wrote: While I agree with Firefly being a top-notch IAX client, Hendrik was hoping for a linux client. I'm also curious which one people recommend. actually I still use gnophone because it's got some cool features I haven't found on others, but it doesn't do iax2 kiax works well for me, does iax2, and is being trialled in our company to replace gnophone pdq. Conrad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
