Re: [asterisk-users] illegal VoIP in India
Yeh problem is they are directly buying from providers in US/UK without paying 12 % tax on voip .. i guess people who buy itsp license can resell this minutes by paying tax to government in between . On 08/12/06, ram [EMAIL PROTECTED] wrote: I'm not sure, but does this only apply to VoIP service providers? What about self run asterisk servers? Tom Hi if the self running Asterisks people connected to Indian ISP not a problem i belive. if they are directly connecting to USA provider, Avoiding India ITSP that could be a problem i think. ram ___ --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] Codec Selection in asterisk
Vicky wrote:
I have around 20-30 softphones behind NAT .. My sip.conf has
nat=yes and
they all are able to register and make calls with no problem . My
voip
carrier supports gsm as well as ilbc .. Server takes calls from sip
phones ,
does call recording in between and forwards to voip carrier . My
problem is
that half of my softphones use ilbc and rest use gsm and my
provider
supports both gsm as well as ilbc . Now when i put
allow=gsmilbc in my
voip carrier's extension then it uses gsm ( first preference ) to
send
calls
but half of my softphones use ilbc so asterisk does codec
transcoding in
between using lot of cpu .. how ever my carrier does support ilbc
tooo but
when i put allow=ilbcgsm then it uses ilbc again and does codec
transcoding
from gsm to ilbc for rest of softphones . How can i make asterisk
to be
smart in choosing codec .. and use ilbc to voip carrier if
softphone is
using ilbc or use gsm when softphone is using gsm ( but still should
do call
recording in between ) .. I am using freepbx for most of
configuration
btw... Any suggestions ?
On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote:you can try this
patch,
0004825: [patch][post 1.4] New codec negotiation algorithm
http://bugs.digium.com/view.php?id=4825
I'm think, this is one of the most wanted feature,
but unfortunately will not be in asterisk 1.4 and we must wait for 1.6
to be officially supported feature :'(
PJ
On 7 Dec 2006, at 21:29, Vicky wrote:
I am still on asterisk 1.2 branch svn ( afraid of word beta on
server :( ) . I will try out that patch.
Alternatively try setting
${SIP_CODEC}
before you place the call to your provider.
I'd love to hear if it works.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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Re: [asterisk-users] Basic question regarding re-INVITE
canreinvite = yes in sip,conf ( trunk section ) ?? No t,t in dial command . No call recording in between , same codec should be supported by both trunk as well as extension . If trunk is iax2 and extension is sip then also asterisk will sit in media path . On 08/12/06, Alex Guan [EMAIL PROTECTED] wrote: All, This basic question might have been asked thousands of timesbut anyways: when can Asterisk send out an re-INVITE to the line/trunk side? It seems that the canreinvite does NOT matter for calls toward the trunk. E.g. When I put a phone on hold, the re-INVITE is sent from phone to the Asterisk, but then that's it. The Asterisk never sends it out. It seems to work for extension to extention, but not extension to line. What am I missing? Thanks! ___ --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] wierd callerid problem
Yeh asterisk seems to use extension number for calls between extensions on same server and sends callerid only for outside numbers ( via sip trunks ) . On 08/12/06, Greg Kennedy [EMAIL PROTECTED] wrote: I have a site running asterisk 1.2.8 with a hand full of polycoms and grandstream 2Kxp's. When a call is missed and you look at the missed call logs on either, its has the persons exten, not the incoming caller id. Any ideas? \\\|/// \\ ~ ~ // ( @ @ ) --oOOo-(_)-oOOo— ___ --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] Running Asterisk on a Home rotuer
Dovid B wrote: tacking pn = adding on - sorry for not being more specific. I have seen that people in the past have used a linksys router to run asterisk. It would be to expensive to bring in a PC for every location. So we want to import cheap home routers put asterisk on them as use them as the go in between the IP phones and the asterisk server. Check with Brian Capouch. He deployed Asterisk on Linksys WRT54G in some rural areas. Caveat here: Cheap = not enough horses :). Don't expect to pass many calls through one of those things. You might want to look at deploying a lightweight SIP proxy on the router instead of asterisk. Leo ___ --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] Server for 100 concurrent calls
Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Can anyone recommend the Server Specs that is ideal for this scenario. Im planning to lease a server. Calls are purely SIP or IAX2 only. Thanks in advance. Dan ___ --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] Management GUI
Hi All Can anyone suggest a comprehensive GUI manager for Asterisk. It doesn't matter if it is open source or commercial. We currently have 100's of users currently managed via the real time database. Groups of users belong to their own contexts. We would like a system that is able to integrate with our current real time setup and then allow us the ability to customise every feature of a user account from an interface as well as allowing other users to login an only manage people within their context. The GUI needs to have a distinction between configuring phones to act as terminals and then configuring agents who can roam around these phones. I look forward to hearing from anyone that can suggest a good GUI or maybe from someone who has a GUI they can customise for us. Many Thanks in Advance SP ___ --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 any Asterisk controllable thermostat?
Doug Crompton wrote: John, Two questions on your comments I have no seen an Insteon computer controller similiar to the old bottle rocket. Is there such a device? I am thinking of getting an Insteon starter kit bit I have so many X10 devices it will be awhie before, if ever, that I get it all changed over. Many items, like spotlights, are not available in Insteon. Similar, in terms of a wireless transmitter -- no. But they have both a serial and a USB computer controller that works over the power ($50-$70 for the controller). It works for both X10 and Insteon protocols. Why isn't that acceptable? And yes, some essential X10 replacements are not yet available. I have two of the X10 spotlights myself. But Insteon has a lot of interest from a lot of companies, so I expect to see a lot more variety in the next year or two. Note: There is opensource software available for the controller, so you don't have to pay the extra $70-$200 or more for the various non opensource software packages available. I would be interested in the Ethernet MWI. I am using many phones on an SPA3000 fxs and I can't seem to find an MWI on an analog phone that works with Asterisk and the SPA3000, although I have been told that there are some that do??? The quick answer would be to put a SIP phone with MWI where your wife wants to be able to see the light. I have a Budgtone 200 and MWI works fine on it. Of course then you have styling and color issues that might not past the muster. Well, the answering machine was a digital one that had multiple (3) VM boxes. It had a separate message waiting light for each box. That is the feature that my wife misses. I'm not sure what if any SIP phones provide multiple message waiting indicators. Besides, it is a moot point for me at this time, since I've already finished building the hardware, now it is just a simple matter of programming to get it to work. ___ --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] Management GUI
Hi Scott... http://www.bicomsystems.com/products/ Senad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Pinhorne Sent: 08 December 2006 11:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Management GUI Hi All Can anyone suggest a comprehensive GUI manager for Asterisk. It doesn't matter if it is open source or commercial. We currently have 100's of users currently managed via the real time database. Groups of users belong to their own contexts. We would like a system that is able to integrate with our current real time setup and then allow us the ability to customise every feature of a user account from an interface as well as allowing other users to login an only manage people within their context. The GUI needs to have a distinction between configuring phones to act as terminals and then configuring agents who can roam around these phones. I look forward to hearing from anyone that can suggest a good GUI or maybe from someone who has a GUI they can customise for us. Many Thanks in Advance SP ___ --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] Management GUI
Scott, What you write sounds standard to any Commerical Application. Our Call Center version has much more besides: CallCenter: http://87.238.74.83/admin/ [EMAIL PROTECTED] pbxware I will contact you directly if I might. Steve steve 'at} bicomsystems .dot} com - Original Message - From: Scott Pinhorne To: asterisk-users@lists.digium.com Sent: Friday, December 08, 2006 12:00 PM Subject: [asterisk-users] Management GUI Hi All Can anyone suggest a comprehensive GUI manager for Asterisk. It doesn't matter if it is open source or commercial. We currently have 100's of users currently managed via the real time database. Groups of users belong to their own contexts. We would like a system that is able to integrate with our current real time setup and then allow us the ability to customise every feature of a user account from an interface as well as allowing other users to login an only manage people within their context. The GUI needs to have a distinction between configuring phones to act as terminals and then configuring agents who can roam around these phones. I look forward to hearing from anyone that can suggest a good GUI or maybe from someone who has a GUI they can customise for us. Many Thanks in Advance SP -- ___ --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] CISCO 2600 - VWIC 1MFT-E1
As I understand your configuration , dial-peer voice 697617664 voip, only forward the pattern 697617664( destination-pattern 697617664) to XXX.XXX.XXX .115:5060 ( session target ipv4:XXX.XXX.XXX.115:5060) that I think is your Asterisk box. An incoming call in your E1 must much a destination pattern, your only destination pattern is 697617664. Usually an E1 has several DID associated it in a consecutive range, 91 5344XXX for example. otherwise, for outgoing calls you must configure a pots dial peer ,you can put a randon name to the dial peer. You can configure asterisk , without user registration with the sip.confinsecure option when I copied dial-peer voice 10 pots destination-pattern 0T should be .T it tells cisco 26xx router what patterns can be reached throught E1 I´ll take a look into the cisco web site for sip user authentication, I have a configuration done, but with FXS interfaces and worsk fine. best regards 2006/12/7, FaberK [EMAIL PROTECTED]: http://pastebin.ca/270840 This is the newone with some changements. Unfortunately, always the same problem. Fran, if I add the dial-peer voice 10 pots, I receive the advise that the number does not exist. Also, I do not find the way to add authentication username asterisk-uername password XX. The story continues... Thanks F. ___ --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] problem with asterisk 1.4
Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but i can't hear the other person voice. but my voice he can able to hear... some times i can't able to make (Between 2 sip comm.)call also... I'm using asterisk 1.4 versoin... could u tell me any suggestions.. Regards, nsthi ___ --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 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)
Update on this - I tried with the newest spandsp on the snapshots site still to no avail. I also ensured no other copies of spandsp exist, and adding SPANDSP_LIBS=-lspandsp to makeopts, but still getting the segfault when rxfax is called. On 07/12/06, Matt Gibson [EMAIL PROTECTED] wrote: Same thing occuring here, on gentoo as well :( On 07/12/06, Chris Glover [EMAIL PROTECTED] wrote: Hi, I have installed the latest version of asterisk(1.4.0-beta3), and built app_rxfax/txfax. I'm using spandsp from here, http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207.tar.gz Everything builds ok. I had to manually apply the patch from the site so configure would spot spandsp libraries. However, when I try dialing my virtual fax extension (either from a phone or fax machine) Asterisk bombs out with the following message... Executing [EMAIL PROTECTED]:1] RxFAX(SIP/101-081d63d0, /tmp/test.tif) in new stack asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: span_set_message_handler This was me dialing from a normal sip extension, hoping to hear fax tones. I did try the latest ebuild Gentoo have (1.2.13), this gave me perfect fax tones, but completely refused to include chan_zap, so I can't win :-) Please somebody tell me where I'm going wrong, been trying to get this to work for hours. I've got rid of all the old libraries, recompiled... my next step is to sacrifice a goat! Any help greatly appreciated. Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ --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] Is there any Asterisk controllable thermostat?
Doug, The Uniden CLX465 supports stutter dial tone (SDT) and provides a MWI. Might be overkill since it is an answering machine as well. There are a few others. Google for stutter dial tone or phone company compatible voice mail. The SPA3K can produce SDT. The Budgetone 102 also has an MWI. I never thought about painting the phone's case. The handset might be an issue. Sounds like an interesting marketing opportunity, like cell-phone covers. Bob... On Thu, 2006-12-07 at 10:14 -0500, Doug Crompton wrote: John, Two questions on your comments I have no seen an Insteon computer controller similiar to the old bottle rocket. Is there such a device? I am thinking of getting an Insteon starter kit bit I have so many X10 devices it will be awhie before, if ever, that I get it all changed over. Many items, like spotlights, are not available in Insteon. I would be interested in the Ethernet MWI. I am using many phones on an SPA3000 fxs and I can't seem to find an MWI on an analog phone that works with Asterisk and the SPA3000, although I have been told that there are some that do??? The quick answer would be to put a SIP phone with MWI where your wife wants to be able to see the light. I have a Budgtone 200 and MWI works fine on it. Of course then you have styling and color issues that might not past the muster. Doug On Thu, 7 Dec 2006, John Marvin wrote: I would suggest that people who don't already have an investment in home automation equipment should look at Insteon rather than X10. Insteon is a next generation version of X10 that provides backwards compatibility with X10. The devices are a little more expensive, but not as expensive as some of the other alternatives. Insteon provides 2 way communication and is a lot more reliable than X10. If you already have an investment in X10 devices you can slowly convert to Insteon, since Insteon provides backwards compatibility, i.e. X10 controllers can control Insteon devices and Insteon controllers can control X10 devices, however you won't get all the advantages of Insteon until you have Insteon controllers controlling Insteon devices. For people with some soldering and basic circuit design skills, you may want to consider using ethernet as a home automation bus for some things. I love the Olimex PIC WEB and PIC Mini Web development boards (they cost $49.95 and $39.95 respectively). They have an ethernet port and an expansion connector for the available PIC I/O pins. Microchip provides a free C compiler for Pic processors, and they also have an open source networking stack that works on the Olimex boards. So with a ribbon cable connector and a small breadboard with a few IC's and/or driver transistors you can build a device that responds to commands via the network (or via a built in web server) from your Asterisk server that does about any task you can think of. Lots of fun ... I'm currently building a voicemail indicator (my wife didn't like me taking her answering machine away with the blinking lights when we switched to Asterisk voicemail) using a PIC Web board. Next project will be a web based sprinkler controller. 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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --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] Asterisk 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)
The only time I have seen this problem myself is when Asterisk (and therefore rxfax) was built when the wrong spandsp header/library files were present on the system. The required order of events is: 1) Build spandsp 2) Install both spandsp binary libraries and includes, ensuring no old versions are present. 3) Rebuild asterisk from scratch 4) Profit... :) If that doesn't work, then there may be a more fundamental problem. Someone did comment a while back that the new code seemed to be freeing an already free chunk of memory - Perhaps Asterisk has changed behaviour and is freeing something that the app used to be required to clean up itself - AFAIK, the workaround was to comment out the free that caused the crash in app_rxfax.c. Cheers, Steve On 12/8/06, Matt Gibson [EMAIL PROTECTED] wrote: Update on this - I tried with the newest spandsp on the snapshots site still to no avail. I also ensured no other copies of spandsp exist, and adding SPANDSP_LIBS=-lspandsp to makeopts, but still getting the segfault when rxfax is called. On 07/12/06, Matt Gibson [EMAIL PROTECTED] wrote: Same thing occuring here, on gentoo as well :( On 07/12/06, Chris Glover [EMAIL PROTECTED] wrote: Hi, I have installed the latest version of asterisk(1.4.0-beta3), and built app_rxfax/txfax. I'm using spandsp from here, http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207.tar.gz Everything builds ok. I had to manually apply the patch from the site so configure would spot spandsp libraries. However, when I try dialing my virtual fax extension (either from a phone or fax machine) Asterisk bombs out with the following message... Executing [EMAIL PROTECTED]:1] RxFAX(SIP/101-081d63d0, /tmp/test.tif) in new stack asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: span_set_message_handler This was me dialing from a normal sip extension, hoping to hear fax tones. I did try the latest ebuild Gentoo have (1.2.13), this gave me perfect fax tones, but completely refused to include chan_zap, so I can't win :-) Please somebody tell me where I'm going wrong, been trying to get this to work for hours. I've got rid of all the old libraries, recompiled... my next step is to sacrifice a goat! Any help greatly appreciated. Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ --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
[asterisk-users] RE: Answer a call that is not ringing on yourextension
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Another solution is to use the Pickup() command. It will pick up a call on a specific extension that is in the ringing state: [Description] Pickup([EMAIL PROTECTED]): This application can pickup any ringing channel that is calling the specified extension. If no context is specified, the current context will be used. For example, my co-workers extension is 203. I hear his phone ringing, and I dial my pre-defined pickup extension (**203) to pickup his call. Dialplan example: Exten = **203,1,Pickup(203) Exten = **203,2,Hangup() Note: the read I use ** is GXP-2000 phones will dial **exten while the BLF light is in ringing state. You can use whatever you want. Wes Baehr Wes, thank you for this information! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] CISCO 2600 - VWIC 1MFT-E1
http://pastebin.ca/271763 Hi to all, To Fran: As I understand your configuration , dial-peer voice 697617664 voip, only forward the pattern 697617664( destination-pattern 697617664) to XXX.XXX.XXX. 115:5060 ( session target ipv4:XXX.XXX.XXX.115:5060) that I think is your Asterisk box. you are right, XXX.XXX.XXX.115:5060 is my * box where I've created a friend called 697617664 An incoming call in your E1 must much a destination pattern, your only destination pattern is 697617664. Usually an E1 has several DID associated it in a consecutive range, 91 5344XXX for example. here too, you are right, but I'm trying to receive at leat 1 call to 697617664, then for all the others will be not a problem. But first i need to let it works...!!! otherwise, for outgoing calls you must configure a pots dial peer ,you can put a randon name to the dial peer. You can configure asterisk , without user registration with the sip.confinsecure option when I copied dial-peer voice 10 pots destination-pattern 0T should be .T it tells cisco 26xx router what patterns can be reached throught E1 I´ll take a look into the cisco web site for sip user authentication, I have a configuration done, but with FXS interfaces and worsk fine. For outgoing calls, at this moment I'm not interested. On the new configuration, I've also changed the codecs, leaving the g711 only. Unfortunately always the same: calling my number, the call reach the 2600(infact I hear the tone), but is not forwarded to the sip-server. To Pavel: thanks for your suggestion regarding MGCP, but the fact is that I got all sip, and never worked with mgcp. Thanks to all Best Regards F. ___ --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] Re: centos 4.4 + asterisk
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... That kernel-devel fix is just for ZAPTEL. The bug has been solved in 4.4 To make it more understandable - Cent OS 4.4 doesn't have problems with Zaptel installation. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Re: SetCallingPres propagation
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello, We have several regional asterisk's connected to a central one making the the PRI calls through a TE410P card. When using SetCallingPres(prohibited) on a call at the regional level, that setting it not forwarded to the central asterisk and the call is made as if no callerid had been sent: the telco substitutes the network number. Using SetCallingPres(prohibited) on the central asterisk works though: the call is received with no callerid at all. How can I suppress callerid presentation at the regional level and keep that setting when trunking the call from regional to central asterisk's? Hi Louis! You shouldn't prohibit CallerID on regional level. Instead you should send *79 (or something else) to central Asterisk. When central Asterisk receives *79SOME_EXTEN he should cut of *79 and execute SetCallingPres(prohibited). Hope this helps. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Re: [asterisk-biz] Server for 100 concurrent calls
[EMAIL PROTECTED] a écrit : Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Since Digium doesn't provide g723 codecs (as far as I'm aware), and there's yet no transcoding card for Asterisk (one is supposed to be out at some point, but when... god knows), for the moment you should look into something else than Asterisk. Cheers, Jean-Michel. ___ --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] Re: Switching from FreeBSD to Linux - which distro?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs great on it. Debian is good too. They have Asterisk packages, but they're generally a little bit old. Source installations work fine. Both have large, active developer and user communities. Hi Carla! Can you tell me from where do you download rpm's for Cent OS 4? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Re: Switching from FreeBSD to Linux - which distro?
yum can be used... direct download from http://isoredirect.centos.org/centos/4/os/i386/CentOS/RPMS/ Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs great on it. Debian is good too. They have Asterisk packages, but they're generally a little bit old. Source installations work fine. Both have large, active developer and user communities. Hi Carla! Can you tell me from where do you download rpm's for Cent OS 4? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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
[asterisk-users] AGI interaction with php
Hi i am planning to develop a php script that will be called from AGI for the management of an IVR application. I'd like to be able to do the following things from php: - retrive callerid - play some audio files to the caller - wait for some DTMF digits - retrive the DTMF - stop the call the php have to collect some information from the user and after some check on a database inster some records into it. Can i do that directly from php or i must do something else? Maybe do you suggest other languages to do that? Thanks nik ___ --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] How to communicated Both SIP and IAX2 each other ?
Yes, as long as Asterisk is in between the two, it can perform the protocol translation. regards David On 12/8/06, raviprakash sunkara [EMAIL PROTECTED] wrote: Hello Users.. Is it possible to do. one UA is SIP and other UA is IAX2, UA(sip)---OpenSER-- Asterisk-- UA(IAX2) . UA(IAX2) --- Asterisk --- OpenSER -- UA (SIP ). other wise we can like that.. UA(SIP ) --- Asterisk-UA(IAX2) But SIP message and IAX messages are different , Then How can we communicate the both SIP and IAX2 -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --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] RE: Polycom buddies question
Use an empty line key to monitor the other phone On Dec 7, 2006, at 1:44 PM, Bill Gibbs wrote: Figures I email this and realized I can hit Menu 1 (Features) 4 (Presence) 2 (Buddy Status) Wow that’s a lot of key strokes. Anyway to reduce that to a one button touch? I don’t mind doing that but I guess I should think of the users J Bill From: Bill Gibbs Sent: Thursday, December 07, 2006 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Polycom buddies question I know this is not asterisk specific but we all know this group is used for Polycom issues as well… I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this… Bill ___ --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] AGI interaction with php
This should get you started: http://www.voip-info.org/wiki/view/Asterisk+AGI+php http://phpagi.sourceforge.net/ Regards, Ove nik600 wrote: Hi i am planning to develop a php script that will be called from AGI for the management of an IVR application. I'd like to be able to do the following things from php: - retrive callerid - play some audio files to the caller - wait for some DTMF digits - retrive the DTMF - stop the call the php have to collect some information from the user and after some check on a database inster some records into it. Can i do that directly from php or i must do something else? Maybe do you suggest other languages to do that? Thanks nik ___ --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
[asterisk-users] Vonage SIP access via asterisk?
Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. Thank You, Steven BerkHolz Soon to be known as HIROTEC AMERICA www.hirotecamerica.com ___ --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 y AGC
Gracias por el translate! 2006/12/8, Angelito Manansala [EMAIL PROTECTED]: IN ENGLISH VERSION: Good night I have mounted the system of predictive marker ASTGUICLIENT in 2 Servants, in one of them who are a Server HP Proliant G3 350 3GB ram with 11 Slackware and Asterisk 1.2.12.1, this single server is in charge of the voice. Soon another server a little but modest (HP ML110 3.2GB), that has Apache and the BD MySQL with ASTGUICLIENT 2.0.1, my agents is connected to this I complete q already the system draws out there, softphone Eyebeam of my agents estan connected to server VOIP, I have in addition in the server voip a card digium to 2 installed ports 2 lines ISDN, In the ASTGUICLIENT I have several campaigns or working, which remove to calls through lines ISDN, single one of these campaigns Extraction the calls through a main IAX or SIP that I have with my branch of another country or my supplier of voice. I have tip maximum of 25 agents connected, the problem this in which arrives a little while in which the marker no longer passes calls to the agents, rather is delayed too much in marking and to pass the call., If one marks from softphone (eyebeam) in pantallla of same I obtain message TRYNG. And after several seconds it manages to remove the call, in the case of my agents q work with the marker, hope by a long time q pass the calls to him. To that it must east problem, the load of my servants is not much not to pass of 5% in his load average? I have updated the version of asterisk that tapeworm and of UNDER thinking q podria to solve but with the new versions the problem even appears? Sera q with 2 servants even sharing the load cannot put but of 20 agents to the system? Previously I had 20 agents everything in a servant, east tapeworm problem but the system was not very slow and towards very difficult the work thus decidi to divide to him to the load to the servant clearing to him the BD and the WEB to happen to him to another servant. Some of you has had east problem, of q forms have solved it Thanks beforehand for its answers Greetings. On 12/8/06, Aldo Alexander Leyva Alvarado [EMAIL PROTECTED] wrote: Buenas noches Tengo montado el sistema de marcador predictivo ASTGUICLIENT en 2 Servidores, en uno de ellos que es un Server HP Proliant G3 350 3GB RAM con Slackware 11 y Asterisk 1.2.12.1 , dicho server solo se encarga de la voz. Luego otro server un poco mas modesto (HP ML110 3.2GB), que tiene Apache y la BD MySQL con el ASTGUICLIENT 2.0.1, mis agentes se conectan a este ultimo ya q el sistema corree alli, los softphone Eyebeam de mis agentes estan conectados al server VOIP, Tengo ademas en el server voip una tarjeta digium de 2 puertos instalados 2 lineas ISDN, En el ASTGUICLIENT tengo varias campañas ya trabajando, las cuales sacan llamadas a traves de las lineas ISDN, solo una de estas campañas Saca las llamadas a traves de una troncal IAX o SIP que tengo con mi filial de otro pais o mi proveedor de voz. Tengo un pico maximo de 25 agentes conectados, el problema esta en que llega un momento en que el marcador ya no pasa llamadas a los agentes, mejor dicho se demora demasiado en marcar y pasar la llamada.,Si uno marca desde un softphone (eyebeam) en la pantallla de mismo obtengo el mensaje TRYNG ..Y despues de varios segundos logra sacar la llamada, en el caso de mi agentes q trabajan con el marcador, esperan por un largo tiempo q le pasen las llamadas. A que se debe este problema, la carga de mis servidores no es mucha no pasar de 5% en su load average? He actualizado la version del asterisk que tenia y del SO pensando q lo podria solucionar pero aun con las nuevas versiones el problema se presenta? Sera q aun con 2 servidores compartiendo la carga no pueda meter mas de 20 agentes al sistema? Anteriormente tuve 20 agentes todo en un servidor, no tenia este problema pero el sistema estaba muy lento y hacia muy dificil el trabajo por lo cual decidi dividirle la carga al servidor quitandole la BD y el WEB para pasarle a otro servidor. Alguno de ustedes ha tenido este problema, de q forma lo han solucionado Gracias de antemano por sus respuestas Saludos Aldo Leyva ___ --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 -- Lito Manansala www.voicefidelity.net Mobile: +63.906.437.0459 PSTN: +63.44.790.6292 sip:[EMAIL PROTECTED] msn: [EMAIL PROTECTED] skype: bulcrack ym: onchang_2000 ___ --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
Re: [asterisk-users] Re: [asterisk-biz] Server for 100 concurrent calls
g723 codec isn't problem, you can obtain for all asterisk versions from: http://kvin.lv/pub/Linux/Asterisk/ PJ Jean-Michel Hiver wrote: [EMAIL PROTECTED] a écrit : Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Since Digium doesn't provide g723 codecs (as far as I'm aware), and there's yet no transcoding card for Asterisk (one is supposed to be out at some point, but when... god knows), for the moment you should look into something else than Asterisk. Cheers, Jean-Michel. ___ --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] How to communicated Both SIP and IAX2 each other ?
how can protocol translation affect jitter propagation to both voip ends (UAs) for dejjiterring? because iax doesn't use RTP for voice stream, it can be issue (?) PJ David Thomas wrote: Yes, as long as Asterisk is in between the two, it can perform the protocol translation. regards David On 12/8/06, raviprakash sunkara [EMAIL PROTECTED] wrote: Hello Users.. Is it possible to do. one UA is SIP and other UA is IAX2, UA(sip)---OpenSER-- Asterisk-- UA(IAX2) . UA(IAX2) --- Asterisk --- OpenSER -- UA (SIP ). other wise we can like that.. UA(SIP ) --- Asterisk-UA(IAX2) But SIP message and IAX messages are different , Then How can we communicate the both SIP and IAX2 -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --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] Switching from FreeBSD to Linux - which distro?
John Novack wrote: Carla Schroder wrote: On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote: On 12/6/06, John Novack [EMAIL PROTECTED] wrote: Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run into some gotcha down the road where there is some missing file that needs to be put who knows where. Wow! Are you sure about that? Doesn't seem like an issue to me. yum install foo is easy, and I've always preferred servers that are as lean as possible, rather than all porky with unnecessary packages and services. Someone else mentioned AstLinux, and it is very nice. About 40 megabytes. No lard at all. That may be true for you and those that know Linux and how to respond to a missing file because it wasn't initially installed. For those who don't practice Linux as a religion but simply want to use a telephony application, it works to install everything, and move on to learning Asterisk and all IT'S warts and gotchas, John Novack ___ --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 That sounds like a microsoft way of doing things.. install 25X more crap than you will ever use. What ever happened to planning and RTFM? signature.asc Description: OpenPGP digital signature ___ --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] Re: Running Asterisk on a Home rotuer
David Cook (Canada) wrote: On 12/7/06, Dovid B [EMAIL PROTECTED] wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid Sure. I have 5 units out there on Linksys WRT54GS v1.1 through v4 units. The software is OpenWRT.org. Asterisk is simply an available package to load once you have replace the original firmware with OpenWRT. Last I checked the Asterisk package version is quite old. Of course you could download the development environment and upgrade that. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --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] Re: Switching from FreeBSD to Linux - which distro?
If you are new to CentOS or redhat based OS's, I would recommend using yum, as it will resolve any dependencies automatically. If you wish to install RPMS directly, you can download them from any CentOS mirror. See the CentOS website. Note: a default install of CentOS installs a bunch of unnecessary services that you will want to turn off using chkconfig service_name off. David On 12/8/06, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs great on it. Debian is good too. They have Asterisk packages, but they're generally a little bit old. Source installations work fine. Both have large, active developer and user communities. Hi Carla! Can you tell me from where do you download rpm's for Cent OS 4? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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
[asterisk-users] TDM400 and analog phone - can't dial
Hi all, I have a problem with dialing digits from my analog phone connected to TDM400 with one FXS card. I can call the phone from SIP, but when I try to dial digits from it, after first digit I receive a busy tone. I thouht that it is the problem with DTFM frequencies, so I changed zone to my country setting, but with no luck. The next thing I did was to make change zaptel.h in zaptel source so the zaptel should recognize pulse dialing and recompiled zaptel again, but only remotely by ssh - so, now, Im not at the office and I cannot get there until Monday, I cannot try if now at least pulse dialing works now. : - ( I'm newbie to Asterisk so, please, can someone check my configuration and tell me I have everything alright (I think its ok, I did it the same way as in asterisk TFOT book) and I can focus to grab theory for DTFM problem through weekend? Thanks a lot. Petosh my configuration files: *extensions.conf:* [internal] exten = 101,1,Dial(SIP/petosh,20) exten = 101,2,Playback(my/notavailable) exten = 101,3,Hangup( ) exten = 200,1,Dial(Zap/1,20) exten = 200,2,Playback(my/notavailable) exten = 200,3,Hangup( ) *zapata.conf: *[trunkgroups] [channels] usercallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes context=incoming signalling=fxo_ks channel = 1 *zaptel.conf:* fxoks=1 loadzone=cz defaultzone=cz ___ --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] MWI across multiple servers
Jon, I would be as well very interested in your Voicemail Solution : AGI + Web Interface to retrieve voice messages. By the way, you sotre to MySQL, do you use ODBC for that ? or something else, in that case, what ;o) ? Thanks in advance ! Jean-Marc On 12/7/06, Jon Farmer [EMAIL PROTECTED] wrote: I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Porier, Jeremy M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 6 December, 2006 4:20:04 PM Subject: [asterisk-users] MWI across multiple servers We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --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 Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --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
[asterisk-users] Re: How to communicated Both SIP and IAX2 each other?
Nothing is end to end in this case. It is two separate sessions, one SIP and one iax. -- -- Steven http://www.glimasoutheast.org Pavel Jezek [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] how can protocol translation affect jitter propagation to both voip ends (UAs) for dejjiterring? because iax doesn't use RTP for voice stream, it can be issue (?) PJ David Thomas wrote: Yes, as long as Asterisk is in between the two, it can perform the protocol translation. regards David On 12/8/06, raviprakash sunkara [EMAIL PROTECTED] wrote: Hello Users.. Is it possible to do. one UA is SIP and other UA is IAX2, UA(sip)---OpenSER-- Asterisk-- UA(IAX2) . UA(IAX2) --- Asterisk --- OpenSER -- UA (SIP ). other wise we can like that.. UA(SIP ) --- Asterisk-UA(IAX2) But SIP message and IAX messages are different , Then How can we communicate the both SIP and IAX2 -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --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 ___ --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] AGI interaction with php
nik600 wrote: Hi i am planning to develop a php script that will be called from AGI for the management of an IVR application. I'd like to be able to do the following things from php: - retrive callerid - play some audio files to the caller - wait for some DTMF digits - retrive the DTMF - stop the call the php have to collect some information from the user and after some check on a database inster some records into it. Can i do that directly from php or i must do something else? Maybe do you suggest other languages to do that? Hi, We have done all the above with PHP from AGI, and it seems to work fine. So go for it! -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] cal recording with email
I'm trying to set on-demand call recording. Here's a snippet of the
pertinent dialplan. The purpose of this is to allow one user in
particular to be able to receive an email recording of the call
everytime he dials *91 + number. The problem is that the email is not
going out or being generated when I use the ${CALLFILENAME} variable.
When I use the actual file name of the gsm recording, the emails go out
without a problem.
[rec-tt-trunkdial]
exten=_*91NXX.,1,SetVar(CALLFILENAME=${TIMESTAMP}:${CALLERIDNUM})
exten=_*91NXX.,n,Monitor(gsm,/var/spool/asterisk/monitor/${CALLFILENAME
},m)
exten=_*91NXX.,n,Set(CALLERID(num)=7188233325)
exten=_*91NXX.,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN:2},,gtTr
)
exten=_*91NXX.,n,Wait(5)
exten=_*91NXX.,n,System(cat /etc/macro-text | mailx -a
/var/spool/asterisk/monitor/ ${CALLFILENAME}.gsm -s Recorded
[EMAIL PROTECTED])
exten=_*91NXX.,n,Hangup()
This is my asterisk console output:
Connected to Asterisk 1.2.12.1 currently running on pbx (pid = 1999)
Verbosity is at least 3
-- Hungup 'IAX2/voicepulse02-8'
-- Executing Wait(SIP/1001-081d9b80, 2) in new stack
-- Executing System(SIP/1001-081d9b80, cat /etc/macro-text |
mailx -a /var/spool/asterisk/monitor/20061208-103611:1001.gsm -s
hello [EMAIL PROTECTED]) in new stack
-- Executing Hangup(SIP/1001-081d9b80, ) in new stack
== Spawn extension (rec-tt-trunkdial, *912126245943, 7) exited
non-zero on 'SIP/1001-081d9b80'
Nothing actually happens. For testing I replaced the ${CALLFILENAME}
variable in the System() command with the actual recording name:
Like this in extensions.conf:
exten=_*91NXX.,n,System(cat /etc/macro-text | mailx -a
/var/spool/asterisk/monitor/20061208-103611:1001.gsm -s Recorded
[EMAIL PROTECTED])
This worked fine so I'm guessing that there's something wrong I'm doing
when passing the ${CALLFILENAME} variable to the linux shell in
System().
Any help would be appreciated. Thanks in advance.
___
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[asterisk-users] CTI: put on hold a call
Hi list, I need no control a call via AMI or AGI or whatever. I don't know how to put a call on hold. Example: an external call ring, in the dial plan I call Dial application to an internal SIP phone. But my SIP phone does not have the on hold feature, so how to put the callee on hold ? Thanks Greg ___ --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] CTI: put on hold a call
One suggestion is to transfer the call to an on-hold extension that plays music, then go pick up the call later... or get a new SIP phone. : ) ~Joel From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Duchatelet Sent: Friday, December 08, 2006 9:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CTI: put on hold a call Hi list, I need no control a call via AMI or AGI or whatever. I don't know how to put a call on hold. Example: an external call ring, in the dial plan I call Dial application to an internal SIP phone. But my SIP phone does not have the on hold feature, so how to put the callee on hold ? Thanks Greg ___ --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] Re: How to communicated Both SIP and IAX2 each other?
so that, jitterbuffer should be enabled forced on sip and iax channel on asterisk (because UAs have no knowledge about jitter on opposite link), from first example? UA(sip)---OpenSER-- Asterisk-- UA(IAX2) Steven wrote: Nothing is end to end in this case. It is two separate sessions, one SIP and one iax. ___ --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] cal recording with email
I think you need quotes around the file-name, but I could be wrong.
It's what I would try, though.
Good luck!
Joe
On Fri, 2006-12-08 at 09:46 -0600, Jeronimo Romero wrote:
I’m trying to set on-demand call recording. Here’s a snippet of the
pertinent dialplan. The purpose of this is to allow one user in
particular to be able to receive an email recording of the call
everytime he dials *91 + number. The problem is that the email is
not going out or being generated when I use the ${CALLFILENAME}
variable. When I use the actual file name of the gsm recording, the
emails go out without a problem.
[rec-tt-trunkdial]
exten=_*91NXX.,1,SetVar(CALLFILENAME=${TIMESTAMP}:${CALLERIDNUM})
exten=_*91NXX.,n,Monitor(gsm,/var/spool/asterisk/monitor/${CALLFILENAME},m)
exten=_*91NXX.,n,Set(CALLERID(num)=7188233325)
exten=_*91NXX.,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN:2},,gtTr)
exten=_*91NXX.,n,Wait(5)
exten=_*91NXX.,n,System(cat /etc/macro-text | mailx
-a /var/spool/asterisk/monitor/ ${CALLFILENAME}.gsm -s Recorded
[EMAIL PROTECTED])
exten=_*91NXX.,n,Hangup()
This is my asterisk console output:
Connected to Asterisk 1.2.12.1 currently running on pbx (pid = 1999)
Verbosity is at least 3
-- Hungup 'IAX2/voicepulse02-8'
-- Executing Wait(SIP/1001-081d9b80, 2) in new stack
-- Executing System(SIP/1001-081d9b80, cat /etc/macro-text |
mailx -a /var/spool/asterisk/monitor/20061208-103611:1001.gsm -s
hello [EMAIL PROTECTED]) in new stack
-- Executing Hangup(SIP/1001-081d9b80, ) in new stack
== Spawn extension (rec-tt-trunkdial, *912126245943, 7) exited
non-zero on 'SIP/1001-081d9b80'
Nothing actually happens. For testing I replaced the ${CALLFILENAME}
variable in the System() command with the actual recording name:
Like this in extensions.conf:
exten=_*91NXX.,n,System(cat /etc/macro-text | mailx
-a /var/spool/asterisk/monitor/20061208-103611:1001.gsm -s
Recorded [EMAIL PROTECTED])
This worked fine so I’m guessing that there’s something wrong I’m
doing when passing the ${CALLFILENAME} variable to the linux shell in
System().
Any help would be appreciated. Thanks in advance.
___
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[asterisk-users] 5.8gig phone MWI
Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug ___ --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] 5.8gig phone MWI
Doug Crompton wrote: Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug I've tested the MWI with the Uniden TRU-8866 phone and it works for me. I've tested it with the Digium TDM400P FXS. Steve ___ --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] Vonage SIP access via asterisk?
BerkHolz, Steven wrote: Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. It would help if you told us exactly which vonage product/service the number, proxy, port and password are for. ___ --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] Vonage SIP access via asterisk?
http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#621VonageBusinessPlusandVonageSoftphoneb Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security BerkHolz, Steven wrote: Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. Thank You, Steven BerkHolz Soon to be known as HIROTEC AMERICA www.hirotecamerica.com ___ --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 Inbound (clean). Database: 0654-1, 12/07/2006 - 12/8/2006 11:10:08 AM ___ --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] CTI: put on hold a call
Another way would be to control the channel from asterisk. It is a SIP feature, not an asterisk feature. I have a SIP phone (not a softphone) and want to control it from the computer. Greg One suggestion is to transfer the call to an on-hold extension that plays music, then go pick up the call later. or get a new SIP phone. : ) ~Joel _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Duchatelet Sent: Friday, December 08, 2006 9:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CTI: put on hold a call Hi list, I need no control a call via AMI or AGI or whatever. I don't know how to put a call on hold. Example: an external call ring, in the dial plan I call Dial application to an internal SIP phone. But my SIP phone does not have the on hold feature, so how to put the callee on hold ? Thanks Greg ___ --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] Re: Vonage SIP access via asterisk?
The service is Business Plus. It is a BYOD SIP service. -- -- Steven http://www.glimasoutheast.org Paul [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] BerkHolz, Steven wrote: Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. It would help if you told us exactly which vonage product/service the number, proxy, port and password are for. ___ --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
[asterisk-users] codec_speex.c: Out of buffer space
Hi all I have installed asterisk 1.2.13 on my P4 Pc with 512MB Ram , FC5 Trunk with my sip provider, on the provider side i have purchaged g729 installed on the client X-lite using speex when i try to make call, i in the log below message Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out of buffer space could some one tell me what is this caused, how can i fix this ? Ram ___ --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] TDM400 and analog phone - can't dial
Hi Jerry, THANKS A LOT. I viewed configuration files so many times, but I had to be blind so I didn't noticed that mistake. I was solving this problem for almost two days with no success... thanks a lot again. :) It can sound weird, but I cannot wait for Monday when I go to work... :D Petosh - Original Message - From: Jerry [EMAIL PROTECTED] To: Petr Kovar [EMAIL PROTECTED] Sent: Friday, December 08, 2006 4:44 PM Subject: Re: [asterisk-users] TDM400 and analog phone - can't dial Hi Petosh, Hi all, I have a problem with dialing digits from my analog phone connected to TDM400 with one FXS card. I can call the phone from SIP, but when I try [...] I'm newbie to Asterisk so, please, can someone check my configuration and tell me I have everything alright (I think its ok, I did it the same way as in asterisk TFOT book) and I can focus to grab theory for DTFM problem through weekend? It's not DTMF. my configuration files: *extensions.conf:* [internal] exten = 101,1,Dial(SIP/petosh,20) exten = 101,2,Playback(my/notavailable) exten = 101,3,Hangup( ) exten = 200,1,Dial(Zap/1,20) exten = 200,2,Playback(my/notavailable) exten = 200,3,Hangup( ) [plus stuff omitted ... important later] *zapata.conf: *[trunkgroups] [channels] usercallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes ; *** Added by me - note the next line (which is in your original) context=incoming signalling=fxo_ks channel = 1 BANG! and there it is. Either you are missing a context of incoming in your extensions.conf, or you didn't list it. Whatever the case, Asterisk is trying to dial based on this incoming context. You can change it to internal, but realize how it works ... it is the part of the dialplan where the digits you are dialing are searched for. If you just cut and pasted from the FOT, it could be they were using an FXO (connected to a phone line), and thus the incoming context was for answering the phone. Hope that makes sense. Thanks, J. ___ --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] Verizon VoiceWing support
Has anyone been able to get Asterisk to work with Verizon's VoiceWing service? I'm in the process of testing Asterisk to see if it will fit the needs of my company. Since I already have Verizon's VoiceWing VoIP service, I figured if I can tie into it, that would let me evaluate service going to a VoIP provider. I've done a bunch of searching, but didn't turn up anything about how to get Asterisk to talk to VoiceWing. Verizon does not seem to officially offer anything except use of their supplied ATA (a LinkSys PAP2 that is locked down just like Vonage does... and none of the Vonage hacks seem to work on the VoiceWing one, so I can't get in and see how it is configured). I know Verizon Business offers VoIP services, including IP Trunking with the expectation that you will supply your own interface hardware. So I figured VoiceWing may be going off the same or similar systems and thus be able to support Asterisk if only the connection info was known. So, has anyone already figured this out and can point me in the right direction? -chris www.mythtech.net ___ --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] How to communicated Both SIP and IAX2 each other ?
Hello Users.. Is it possible to do. one UA is SIP and other UA is IAX2, UA(sip)---OpenSER-- Asterisk-- UA(IAX2) . UA(IAX2) --- Asterisk --- OpenSER -- UA (SIP ). other wise we can like that.. UA(SIP ) --- Asterisk-UA(IAX2) But SIP message and IAX messages are different , Then How can we communicate the both SIP and IAX2 -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --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] Re: Vonage SIP access via asterisk?
That and any other ref.s I have found give me a 404 error when dialing out. My Sip show registry is also empty. ref: We're at 64.x.x.x port 12146 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 21 lines Reliably Transmitting (NAT) to 216.115.20.41:5061: INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport From: SteveB TEST sip:[EMAIL PROTECTED];tag=as35e23a92 To: sip:[EMAIL PROTECTED]:5061 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 08 Dec 2006 17:15:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 494 v=0 o=root 9983 9983 IN IP4 64.118.155.160 s=session c=IN IP4 64.118.155.160 t=0 0 m=audio 12146 RTP/AVP 0 8 4 3 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called [EMAIL PROTECTED] tg05*CLI -- SIP read from 216.115.20.41:5061: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport From: SteveB TEST sip:[EMAIL PROTECTED];tag=as35e23a92 To: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 15 Content-Length: 0 --- (8 headers 0 lines) --- Transmitting (NAT) to 216.115.20.41:5061: ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport From: SteveB TEST sip:[EMAIL PROTECTED];tag=as35e23a92 To: sip:[EMAIL PROTECTED]:5061 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- -- Steven http://www.glimasoutheast.org Al Bochter [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#621VonageBusinessPlusandVonageSoftphoneb Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security BerkHolz, Steven wrote: Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. Thank You, Steven BerkHolz Soon to be known as HIROTEC AMERICA www.hirotecamerica.com ___ --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 Inbound (clean). Database: 0654-1, 12/07/2006 - 12/8/2006 11:10:08 AM ___ --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] Asterisk 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)
In this case, the machine was a spandsp virgin, it had never been installed before. I made sure I ran ldconfig before and after building, and still no joy. I have managed to get iaxmodem and hylafax to work quite well though :-) Chris On Fri, 2006-12-08 at 12:43 +, Steve Davies wrote: The only time I have seen this problem myself is when Asterisk (and therefore rxfax) was built when the wrong spandsp header/library files were present on the system. The required order of events is: 1) Build spandsp 2) Install both spandsp binary libraries and includes, ensuring no old versions are present. 3) Rebuild asterisk from scratch 4) Profit... :) If that doesn't work, then there may be a more fundamental problem. Someone did comment a while back that the new code seemed to be freeing an already free chunk of memory - Perhaps Asterisk has changed behaviour and is freeing something that the app used to be required to clean up itself - AFAIK, the workaround was to comment out the free that caused the crash in app_rxfax.c. Cheers, Steve On 12/8/06, Matt Gibson [EMAIL PROTECTED] wrote: Update on this - I tried with the newest spandsp on the snapshots site still to no avail. I also ensured no other copies of spandsp exist, and adding SPANDSP_LIBS=-lspandsp to makeopts, but still getting the segfault when rxfax is called. On 07/12/06, Matt Gibson [EMAIL PROTECTED] wrote: Same thing occuring here, on gentoo as well :( On 07/12/06, Chris Glover [EMAIL PROTECTED] wrote: Hi, I have installed the latest version of asterisk(1.4.0-beta3), and built app_rxfax/txfax. I'm using spandsp from here, http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207.tar.gz Everything builds ok. I had to manually apply the patch from the site so configure would spot spandsp libraries. However, when I try dialing my virtual fax extension (either from a phone or fax machine) Asterisk bombs out with the following message... Executing [EMAIL PROTECTED]:1] RxFAX(SIP/101-081d63d0, /tmp/test.tif) in new stack asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: span_set_message_handler This was me dialing from a normal sip extension, hoping to hear fax tones. I did try the latest ebuild Gentoo have (1.2.13), this gave me perfect fax tones, but completely refused to include chan_zap, so I can't win :-) Please somebody tell me where I'm going wrong, been trying to get this to work for hours. I've got rid of all the old libraries, recompiled... my next step is to sacrifice a goat! Any help greatly appreciated. Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ --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 -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ --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] Question on retrieve_file() function in app_voicemail.c
I understand this function (line 832 in app_voicemail.c) is used to retrieve a voice message. What I don't understand however is why .txt is appended to the end of the filename. Could someone shed some light on this for me? Thanks, Jez if (msgnum -1) make_file(fn, sizeof(fn), dir, msgnum); else ast_copy_string(fn, dir, sizeof(fn)); snprintf(full_fn, sizeof(full_fn), %s.txt, fn); f = fopen(full_fn, w+); Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --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] Switching from FreeBSD to Linux - which distro?
Other than for Zap cards, why would you want to switch from *BSD to linux? I don't run * on *BSD, but I've heard it runs very smoothly and stable (probably more than several linux distros). Just curious. Thanks, Daniel -Original Message- From: John Novack [EMAIL PROTECTED] Sent: Thu, December 7, 2006 8:42 pm To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Switching from FreeBSD to Linux - which distro? Carla Schroder wrote: On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote: On 12/6/06, John Novack [EMAIL PROTECTED] wrote: Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run into some gotcha down the road where there is some missing file that needs to be put who knows where. Wow! Are you sure about that? Doesn't seem like an issue to me. yum install foo is easy, and I've always preferred servers that are as lean as possible, rather than all porky with unnecessary packages and services. Someone else mentioned AstLinux, and it is very nice. About 40 megabytes. No lard at all. That may be true for you and those that know Linux and how to respond to a missing file because it wasn't initially installed. For those who don't practice Linux as a religion but simply want to use a telephony application, it works to install everything, and move on to learning Asterisk and all IT'S warts and gotchas, John Novack ___ --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
[asterisk-users] downloading asterisk GUI
This may be a Linux newby question, but here it goes. I was reading the instructions on downloading and installing Asterisk GUI, but I can't get this to work. svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui What would be the equivalent command in CentOS 4? http://astrecipes.net/?n=217 ___ --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] queue agent Monitor
One thing you could do is use a third-party product like our QueueMetrics
(available free for smaller systems/SOHOs) and use its own internal logic
to link a callerid to all other information (call status, agent, time,
etc), search by different criteria and remote call listening.
Hope this helps,
l.
On Thu, 07 Dec 2006 23:34:53 +0100, Ed Nuñez [EMAIL PROTECTED] wrote:
I just tried that and it doesn't work. This may be perhaps because the
file name needs to be defined before the call is sent to the queue.
When I saw you answer I thought it would work because it sounded very
logical. :-)
This is the macro I use to send the call to the extension
Just in case I put the line before and after the extension.
[macro-extensions]
exten = s,1,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})
exten = s,2,Dial(${ARG1}|30|t,,wW)
exten = s,3,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})
exten = s,4,Voicemail(u${ARG2})
exten = s,104,Voicemail(b${ARG2})
Ed Nuñez
--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
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[asterisk-users] Asterisk eating the Asterisk key!
Hi all, I'm using Asterisk 1.4.0-beta2 and lately I've noticed that I'm having trouble accessing my voicemail at work using phones on my Asterisk system. I have to press the * key during the voicemail login process. When I do, it seems that Asterisk eats it and doesn't send it along. I suspect it has something to do with the features.conf file, which you can look at at: http://diehlnet.com/features.conf Otherwise, any advise would be most welcome. Mike Diehl. ___ --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] MWI across multiple servers
On 8 Dec 2006, at 15:02, Jean-Marc Salsa wrote: Jon, I would be as well very interested in your Voicemail Solution : AGI + Web Interface to retrieve voice messages. By the way, you sotre to MySQL, do you use ODBC for that ? or something else, in that case, what ;o) ? Thanks in advance ! Jean-Marc On 12/7/06, Jon Farmer [EMAIL PROTECTED] wrote: I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards You might want to look at integrating my (free opensource) gsmPlay applet into the web front end of that, it would let your users play their gsm voicemails without installing quicktime... Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] downloading asterisk GUI
svn is application called subversion, you should download and install it first. - Original Message - From: Ed Nuñez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 08, 2006 7:18 PM Subject: [asterisk-users] downloading asterisk GUI This may be a Linux newby question, but here it goes. I was reading the instructions on downloading and installing Asterisk GUI, but I can't get this to work. svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui What would be the equivalent command in CentOS 4? http://astrecipes.net/?n=217 -- ___ --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
[asterisk-users] Dial groups, groups of phones, multiple line keys
I have 4 Polycom phones with multiple line keys so multiple incoming calls work fine The way I would like the incoming call flow to work is as follows: 1) 2 groups consisting of 2 phones each 2) Incoming call rings the first group, if no answer, the 2nd group is rung 3) However if the first 2 are on a call or busy, it will immediately ring the 2nd group 4) If one of the first group is in use, the available phone is rung, if no answer, roll over to group 2 5) If group 2 one phone is busy, ring the other one only 6) Finally drop into voicemail if no answer at all Suggestions on how to do that yet still keep the multiple line keys? Would this be a good use of CheckGroup and Set(GROUP())? I could use astdb but I wanted to stay away from persistent variables. I looked into ChanIsAvail but I don't think that is what I want. So I guess what I am looking for is there a way to find out if a device is using ANY channel, because I can check that (say CheckIfPhoneInUse(SIP/phone1)) and set dynamic variable values based on that, then decide what phones to ring based on that. Bill ___ --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] Backgroung usage
I try to use the background cmd for send incomings call on dial plan. I try in an internal number for resting: exten = 405,1,DigitTimeout,5 exten = 405,2,ResponseTimeout,10 exten = 405,3,Background(vm-accueilcreat) exten = 1,1,Goto(creat-in,s,1) exten = 2,1,Dial(IAX2/301,15,tr) exten = 3,1,Hangup But nothing happen when i hit 1, 2, or 3. Wher is the mistake?? I have something similar but I use the 's' extension. This works for me. exten = 405,1,GoTo(incomingco,s,1) [incomingco] exten = s,2,Wait,1 exten = s,3,Set(TIMEOUT(digit)=22) exten = s,4,Set(TIMEOUT(response)=40) exten = s,5,BackGround(HU-welcome) exten = s,6,BackGround(HU-welcome) exten = t,1,Goto(Queue_main,4024631371,1) include = desks Don Pobanz ___ --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] Question on retrieve_file() function in app_voicemail.c
On Fri, Dec 08, 2006 at 08:23:36AM -0800, je . wrote: I understand this function (line 832 in app_voicemail.c) is used to retrieve a voice message. What I don't understand however is why .txt is appended to the end of the filename. Could someone shed some light on this for me? This is the small .txt file that contains some metadata and is in the voicemail box. Do you use the simple file-based mailbox? If so, simply have a look in the mailbox directory. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Re: [asterisk-biz] Server for 100 concurrent calls
that site also has g729 codecs for asterisk but is it legal to use them ?? ( digium charges $10 each g729 channel ) On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: g723 codec isn't problem, you can obtain for all asterisk versions from: http://kvin.lv/pub/Linux/Asterisk/ PJ Jean-Michel Hiver wrote: [EMAIL PROTECTED] a écrit : Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Since Digium doesn't provide g723 codecs (as far as I'm aware), and there's yet no transcoding card for Asterisk (one is supposed to be out at some point, but when... god knows), for the moment you should look into something else than Asterisk. Cheers, Jean-Michel. ___ --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
[asterisk-users] Douglas Garstang [EMAIL PROTECTED]
On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote: Hi Steve. Thanks, but unfortunately, I can't be involved in that. We are running Asterisk in a production environment and we're using 1.2, not 1.4. I don't have the resources to work with 1.4. Last time I filed a bug against 1.2 I got told off. Here's an example of that cruddy output. hestia*CLI dundi show peer 00:0e:0c:a1:92:4d Peer:00:0e:0c:a1:92:4d Model: Symmetric Host:xxx.187.142.203 Dynamic: no KeyPend: no Reg: No In Key: dundikey Out Key: dundikey Include logic: -- include all Query logic: -- permit all hestia*CLI The delimiter should not be the colon, as the data may also contain a colon (in this case the MAC address). That makes it really difficult to split the data into fields. Also, the apparent key:value rule gets broken when you get down to the Include Logic line. The '--include all' should be on the same line! Just about every single Asterisk command has screwed up output like this. Fixing all this would be a LOT of work. Doug. Doug-- I'm confused now. You had some references to manager related output, and now, you are complaining that CLI output isn't easily machine readable. CLI responses were always intended to be read by humans, and the format above is tailored to be read by humans. I have no intention in modifying CLI responses. They look fine. If you are going to analyze human-readable output, you are getting into the realm of data mining, and yes, it'll be hard to parse. Add to that, the fact that there is no guarantee that the responses won't change from release to release to fit with changing conditions, times, and needs, and you have a real challenge ahead of you. Am I to assume that you playing with CLI output, because you need info you can't get via the manager interface? We would need to (in the case of your example) extend the dundi module to provide manager actions that would provide the info you need. If this is the case, then it would help to know which juicy tidbits you need, so we can do it. If this isn't the case, then it might be better to interact with asterisk at a different level... reading the config files, writing an app, something...! Funcs are another way to access otherwise embedded information, as your example indicates. Maybe we need to create a function to access the fields you need? Tell us what and why, and we may have either condolences or advice for you. What about the \r\n stuff? Where, what? Let me know... I hope you haven't decided, now that you've programmed around it, you don't want it to change! Because Murphy's Law says we'll find the probs with or without your help, and you'll have to deal with the changes at some point in time. I'm willing to fix problems in 1.2; I'll make sure they carry forward into 1.4 and above. murf -Original Message- From: Steve Murphy [mailto:[EMAIL PROTECTED] Sent: Thursday, December 07, 2006 4:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: What's up with the Manager Interface?!?! Doug, Everyone: I'll make you an offer you (hopefully) can't refuse: I've been fixing manager bugs here and there, and am willing to take on any manager issues out there, for 1.4, and trunk, especially, so as to have things nice and solid for 1.4 before it gets out of beta. So, give me some details. I will file the bug, if you don't. I will reproduce(if I can), and debug, and fix 'em. Just tell me (as explicitly as possible, please!) what the problems are-- especially you, Doug-- where are those inconsistencies, exactly? Richard-- I'll lab up 1.4 and see if I can get the hiccups you mention. murf smime.p7s Description: S/MIME cryptographic signature ___ --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] Re: [asterisk-biz] Server for 100 concurrent calls
that site also has g729 codecs for asterisk but is it legal to use them ?? ( digium charges $10 each g729 channel ) On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: g723 codec isn't problem, you can obtain for all asterisk versions from: http://kvin.lv/pub/Linux/Asterisk/ PJ Jean-Michel Hiver wrote: [EMAIL PROTECTED] a écrit : Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Since Digium doesn't provide g723 codecs (as far as I'm aware), and there's yet no transcoding card for Asterisk (one is supposed to be out at some point, but when... god knows), for the moment you should look into something else than Asterisk. Cheers, Jean-Michel. ___ --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
[asterisk-users] Repeated Digits
Hi, Have anyone experience repeated digits when connecting a call from SIP and terminating it to a PRI Channel? On the other side of the PRI Channel is an IVR that expect a pin but the digits come repeated. For example, you dial 12345 but it is received as 12224445 -- Gustavo Flores IT Manager IAS FILM Corp. ___ --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 forgetting about client registration or Polycom phone forgetting to register?
I'm having trouble with Polycom 501 phones that asterisk forgets how to reach them. /etc/asterisk/sip.conf: [general] context=default MusicOnHold=default port=5060 bindaddr=0.0.0.0 srvlookup=no;yes language=en dtmfmode=rfc2833 maxexpiry=600 defaultexpiry=120 [502] type=friend username=502 secret=pass host=dynamic [EMAIL PROTECTED] callerid= Operator 502 context=rm dtmfmode=rfc2833 accountcode= setvar=DINTERNAL=1 In extensions.conf I have hints setup that is monitored from a 601 with the expansion module. I also have around 7 sessions connecting to the manager API over the network using http://www.snapanumber.com/ . Versions: Asterisk 1.2.13 built by root @ pbx on a x86_64 running Linux on 2006-11-13 16:44:01 UTC [EMAIL PROTECTED]:~# cat /home/Polycom/sip.ver 1.6.7.0094 for 11402_001 1.6.7.0098 for all other platforms bootrom is 3.2.1 Please hep. TIA ___ --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] Backgroung usage
How long in seconds is the vm-accueilcreat recording? Have you tried pressing 1,2, or 3 while it's played? On 12/7/06, Olivier Saulnier [EMAIL PROTECTED] wrote: Hello, I try to use the background cmd for send incomings call on dial plan. I try in an internal number for resting: exten = 405,1,DigitTimeout,5 exten = 405,2,ResponseTimeout,10 exten = 405,3,Background(vm-accueilcreat) exten = 1,1,Goto(creat-in,s,1) exten = 2,1,Dial(IAX2/301,15,tr) exten = 3,1,Hangup But nothing happen when i hit 1, 2, or 3. Wher is the mistake?? Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] Asterisk forgetting about client registration or Polycom phone forgetting to register?
I'm having trouble with Polycom 501 phones that asterisk forgets how to reach them. ... host=dynamic We've found much better results with the static IP here. Can you try this? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --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] Douglas Garstang [EMAIL PROTECTED]
-Original Message- From: Steve Murphy [mailto:[EMAIL PROTECTED] Sent: Friday, December 08, 2006 12:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Douglas Garstang [EMAIL PROTECTED] On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote: Hi Steve. Thanks, but unfortunately, I can't be involved in that. We are running Asterisk in a production environment and we're using 1.2, not 1.4. I don't have the resources to work with 1.4. Last time I filed a bug against 1.2 I got told off. Here's an example of that cruddy output. hestia*CLI dundi show peer 00:0e:0c:a1:92:4d Peer:00:0e:0c:a1:92:4d Model: Symmetric Host:xxx.187.142.203 Dynamic: no KeyPend: no Reg: No In Key: dundikey Out Key: dundikey Include logic: -- include all Query logic: -- permit all hestia*CLI The delimiter should not be the colon, as the data may also contain a colon (in this case the MAC address). That makes it really difficult to split the data into fields. Also, the apparent key:value rule gets broken when you get down to the Include Logic line. The '--include all' should be on the same line! Just about every single Asterisk command has screwed up output like this. Fixing all this would be a LOT of work. Doug. Doug-- I'm confused now. You had some references to manager related output, and now, you are complaining that CLI output isn't easily machine readable. CLI responses were always intended to be read by humans, and the format above is tailored to be read by humans. I have no intention in modifying CLI responses. They look fine. If you are going to analyze human-readable output, you are getting into the realm of data mining, and yes, it'll be hard to parse. Add to that, the fact that there is no guarantee that the responses won't change from release to release to fit with changing conditions, times, and needs, and you have a real challenge ahead of you. Steve, the cli output I showed IS from the Manager Interface, accessed via syntax 'Action: Command:\nCommand foo\n\n' syntax. Given that the number of native commands available to the Manager is small, this means that MOST commands must be accessed via the 'Action: Command\nCommand: command\n\n' syntax, and it therefore means that CLI inconsistencies plague most operations. Why haven't ALL the CLI commands been added to the Asterisk manager interface? Am I to assume that you playing with CLI output, because you need info you can't get via the manager interface? We would need to (in the case of your example) extend the dundi module to provide manager actions that would provide the info you need. If this is the case, then it would help to know which juicy tidbits you need, so we can do it. How about extending all commands so that everything available via the CLI is available via the AMI? Otherwise, what's the point in having a manager interface? If this isn't the case, then it might be better to interact with asterisk at a different level... reading the config files, writing an app, something...! Funcs are another way to access otherwise embedded information, as your example indicates. Maybe we need to create a function to access the fields you need? Oh right... C programming. Tell us what and why, and we may have either condolences or advice for you. We need to access the complete range of functions available via the CLI, via the manager interface. ___ --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] Re: What's up with the Manager Interface?!?!
Steve Murphy wrote: *snipped I've been fixing manager bugs here and there, and am willing to take on any manager issues out there, for 1.4, and trunk, especially, so as to have things nice and solid for 1.4 before it gets out of beta. *snipped Richard-- I'll lab up 1.4 and see if I can get the hiccups you mention. *snipped given the differences between the 1.2 and 1.4 manager interfaces, i am not sure how much this will help... i just modified my application to tail the last 20 lines of asterisk debug (verbose 3), asterisk messages, and my applications debug log to another log. (at one site) this *should* make it a little easier for me to track down the 'hiccup' issue. ___ --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] downloading asterisk GUI
yum install subversion On 09/12/06, Kovar Petr [EMAIL PROTECTED] wrote: svn is application called subversion, you should download and install it first. - Original Message - *From:* Ed Nuñez [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Friday, December 08, 2006 7:18 PM *Subject:* [asterisk-users] downloading asterisk GUI This may be a Linux newby question, but here it goes. I was reading the instructions on downloading and installing Asterisk GUI, but I can't get this to work. svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui What would be the equivalent command in CentOS 4? http://astrecipes.net/?n=217 -- ___ --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] No ID from the calling party in SIP Header
callerid=John Doe 1234 On 05/12/06, Sven Beisiegel [EMAIL PROTECTED] wrote: Hi... I just started working with Asterisk and found something that looks like an error, but i want to be sure, so that's why I'm asking you. When i make a call from A to B (both SIP clients), I don't see the name of the called party in the phone that initiated the call, just the dialed number. I made an ethereal trace and found out, that there is no name during the initiation in the SIP Header? But there is a Remote-Party-ID in the SIP Packet that goes from the Server to the called party...There is nothing like P-Asserted-Id in the SIP Packet that goes to the calling party. My question... Is this an error or did i forget to activate something? The configuration of the sip.conf is: [general] language=de port=5060 disallow=all allow=alaw allow=ulaw allow=GSM nat=no canreinvite=no tos=lowdelay context=default [9001] type=friend username=9001 secret=password host=dynamic callerid=Beckenbauer, Franz 9001 context=default mailbox=9001 callgroup=1 pickupgroup=1 sendrpid=yes [9002] type=friend username=9002 secret=password host=dynamic callerid=Walter, Fritz 9002 context=default mailbox=9002 callgroup=1 pickupgroup=1 sendrpid=yes cheers, Sven ___ --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] Question on retrieve_file() function in app_voicemail.c
Great, exactly what I was looking for. Thanks so much! Shabbat shalom Jez --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Dec 08, 2006 at 08:23:36AM -0800, je . wrote: I understand this function (line 832 in app_voicemail.c) is used to retrieve a voice message. What I don't understand however is why .txt is appended to the end of the filename. Could someone shed some light on this for me? This is the small .txt file that contains some metadata and is in the voicemail box. Do you use the simple file-based mailbox? If so, simply have a look in the mailbox directory. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --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] Best book to learn SIP details ?
Hi, Which is the best book to self-learn SIP ? Regards ___ --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] SIP/IAX Fax Detect on Asterisk 1.4
Hello, Has anyone managed to compile app_nvfaxdetect on asterisk 1.4? Is there any other way of detecting incoming fax calls on non-Zap channels? Julian. ___ --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] Switching from FreeBSD to Linux - which distro?
Derek Whitten wrote: John Novack wrote: snip That sounds like a microsoft way of doing things.. install 25X more crap than you will ever use. What ever happened to planning and RTFM? I guess it all depends on what the objective is. One can sit around and RTFM and play with oneself or one can get on with the task at hand, in this case learning a telephony application and ITs warts and shortcomings. There is no harm in installing an extra thousand or two files that are never needed and services that aren't run or used Disk space is REALLY cheap, in fact most of the hardware is these days. Time is valuable. everyone has a limited amount of it, so why waste hours or days of it because one file wasn't installed. If you want to discourage potential users of Asterisk, not installing the complete distro is a wonderful way to do it John Novack ___ --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] Re: Switching from FreeBSD to Linux - which distro?
David Thomas wrote: If you are new to CentOS or redhat based OS's, I would recommend using yum, as it will resolve any dependencies automatically. If you wish to install RPMS directly, you can download them from any CentOS mirror. See the CentOS website. Note: a default install of CentOS installs a bunch of unnecessary services that you will want to turn off using chkconfig service_name off. David It MIGHT be useful for SOMEONE to specify what those unnecessary services are John Novack ___ --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] Switching from FreeBSD to Linux - which distro?
Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. CentOS works well for me : http://www.centos.org/ ___ --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] CTI: put on hold a call
I´m looking for the same feature performed with the manager, but I think should be the same problem you are experiencing I need to place music on hold (park) an specific call, while the agent performs a process/question/inquiry, and then retakes the call. Is there not a way to park the call? Gregory Duchatelet escribió: Another way would be to control the channel from asterisk. It is a SIP feature, not an asterisk feature… I have a SIP phone (not a softphone) and want to control it from the computer. Greg One suggestion is to transfer the call to an “on-hold” extension that plays music, then go pick up the call later… or get a new SIP phone. : ) ~Joel *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Gregory Duchatelet *Sent:* Friday, December 08, 2006 9:51 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] CTI: put on hold a call Hi list, I need no control a call via AMI or AGI or whatever. I don’t know how to put a call on hold. Example: an external call ring, in the dial plan I call “Dial” application to an internal SIP phone. But my SIP phone does not have the “on hold” feature, so how to put the callee on hold ? Thanks Greg ___ --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] downloading asterisk GUI
Thanks Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mail list Sent: Friday, December 08, 2006 4:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] downloading asterisk GUI yum install subversion On 09/12/06, Kovar Petr [EMAIL PROTECTED] wrote: svn is application called subversion, you should download and install it first. - Original Message - From: Ed Nuñez mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com Sent: Friday, December 08, 2006 7:18 PM Subject: [asterisk-users] downloading asterisk GUI This may be a Linux newby question, but here it goes. I was reading the instructions on downloading and installing Asterisk GUI, but I can't get this to work. svn checkout http://svn.digium.com/svn/asterisk-gui/trunk http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui What would be the equivalent command in CentOS 4? http://astrecipes.net/?n=217 _ ___ --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 image001.gif Description: image001.gif ___ --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] Monitor Zap Status - Full E-mail...
So there are 0 watchers while the GXP is configured to that hint? are you sure you set the phone to Asterisk BLF? On 11/15/06, Ken Williams [EMAIL PROTECTED] wrote: Upon further investigation I must be doing something wrong. It was my understanding that a hint extension could be anything, it wasn't the same as a real extension, though you could make it the same to make it easier. That being said *exten = 702,hint,SIP/702 *works, while *exten = 102,hint,SIP/702* doesn't. I've got a GXP-2000 with the first button set to AsteriskBLF username 102 and the second button set to AsteriskBLF username 702, only the second button actively monitors 702. I've read, reread, rereread and so on a ot of examples of hint files and I can't figure out why the GXP-2000 doesn't like them. When I do *show hints* in CLI it is registering both 102 702 properly. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Ken Williams *Sent:* Wednesday, November 15, 2006 1:54 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Monitor Zap Status - Full E-mail... I've installed Grandstream GPX-2000 phones and have successfully enabled one of my buttons to use Asterisk BLF for an extension. I can tell when this extension is available, is being rung, or is on the line. I'd like to do the same for my Zaptel channels, to be able to see when a line is onhook, ringing or offhook. I tried the following but alas, it doesn't seem to be working: *exten = 102,hint,ZAP/2* I based that on: *exten = 732,hint,SIP/732 * which does work for the SIP phones. If I do show hints in the CLI, I get 102 : ZAP/2 State:Idle Watchers 0 732 : ZAP/1 State:Idle Watchers 0 When a call is made on the ZAP/2 line the State changes to InUse, so I know it's working on that side. Any thoughts or suggestions as to how I can monitor a ZAP line on my GPX-2000? The problem is we have 6 lines, so my plan was to use the 7 buttons down the side, the top 6 for lines and the 7th for paging. Thanks for the help, Ken *Sorry about duplicate e-mail, accidentally hit ENTER when holding CTRL instead of V to paste ___ --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] Cisco 7970 SIP upgrade issues
Hi Ok I have the right version many thanks However I am still a tad stuck (Sorry) I have all the configs to upgrade from SCCP to SIP but what config files do I need just to upgrade the sccp to the 7.0-3 version. I am assuming I need to have a file in the tftp dir that tells the phone to load a specific image. Thanks - Original Message - From: Lacy Moore - Aspendora To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 01, 2006 10:29 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues On 11/29/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( You apparently downloaded the wrong version. I don't know what version you downloaded. You need the zip version of cmterm-7970-7971-sccp-7.0-3. Unzip it to your tftp directory. There is no setup file. -- ___ --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
[asterisk-users] Polycom soft buttons not working
Anyone else have problems with soft buttons not being responsive at all? 2 of the 4 soft buttons do not respond, no matter how hard you push. It is an IP500. Well over 1 year old. ___ --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] Plantronics and Snom RF feedback
the autolifter is for phones without a headset jack. On 12/7/06, J. Oquendo [EMAIL PROTECTED] wrote: Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's inner oddities this is how I got it to work. Plantronic -- RJ11 -- SnomHandset Port (on Snom Base) Handset -- Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) -- Snom's Headset port, the auto lift on the Plantronic wouldn't work until the person pressed the headset key. Even by leaving the headset key on by default, Snom would revert to normal (non headset) mode whenever the headset piece was used. (Sort of defeats the purpose of walking away from your phone only to walk back to re-press the headset key)... How are others setting up these Plantronics... -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --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] Re: Switching from FreeBSD to Linux - which distro?
On redhat based OS's I would do this... You can run the following command to see what services are enabled: chkconfig --list | grep 3:on Then disable whichever ones you dont need... The services may vary a bit depending on hardware or what packages you have installed. I often disable everything except network, iptables sshd; like this... chkconfig acpid off chkconfig atd off chkconfig autofs off chkconfig cpuspeed off chkconfig cups off chkconfig gpm off chkconfig haldaemon off chkconfig isdn off chkconfig mdmonitor off chkconfig messagebus off chkconfig netfs off chkconfig nfslock off chkconfig pcmcia off chkconfig portmap off chkconfig rawdevices off chkconfig rpcgssd off chkconfig rpcidmapd off chkconfig anacron off chkconfig crond off chkconfig kudzu off chkconfig sendmail off chkconfig smartd off chkconfig syslog off chkconfig xinetd off chkconfig irqbalance off chkconfig microcode_ctl off chkconfig sshd on chkconfig iptables on chkconfig network on then reboot. Regards, David ___ --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] SIP Quality Metrics
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[asterisk-users] trixbox
Hi Does trixbox comes with a predictive dialer, i want to use a predictive dialer with trix box or asterisk, please let me know what is the best tot use. Regards Kanishka ___ --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] using a mobile phone as a handset via bluetooth
Normally when you think of using Bluetooth with mobile phones you think of using it to attach a headset wirelessly to a mobile phone... can it work the other way? Can I have a Bluetooth card on my laptop/desktop such that my mobile phone can be a handset to a softphone on the laptop/desktop? Thanks James ___ --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 voice recording through TE110p
Hi all, We are in the process of setting up a E1 (TE110p)connection based asterisk server in which we want to record all the voice conversations.Is this facility supported on asterisk if so how to configure.What are hardware dependencies invloued in setting up this facility. Thanks in advance. with regards raja - Everyone is raving about the all-new Yahoo! Mail beta.___ --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 voice recording through TE110p
Asterisk can record all outgoing calls ( see voip-info.org for asterisk cmd monitor and mixmonitor ) hardware requirements depends on volume of calls to be recorded . Faster sata raid or scsi drives recommended for high number of alternate calls . On 09/12/06, Raja Chidambaram [EMAIL PROTECTED] wrote: Hi all, We are in the process of setting up a E1 (TE110p)connection based asterisk server in which we want to record all the voice conversations.Isthis facility supported on asterisk if so how to configure.What are hardware dependencies invloued in setting up this facility. Thanks in advance. with regards raja -- Everyone is raving about the all-new Yahoo! Mail beta.http://us.rd.yahoo.com/evt=45083/*http://advision.webevents.yahoo.com/mailbeta ___ --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] 5.8gig phone MWI
Thanks, but unfortunately that is an expensive 2 line phone compared to others in their line that have a base and two or three remotes for the same price. Seems a lot to pay for a MWI. I wonder if anyone has had experience with panasonic wireless 5.8gig and MWI?? They advertise compatibility on some models but I also saw a review comment that it did not work. Doug On Fri, 8 Dec 2006, Steve Prior wrote: Doug Crompton wrote: Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug I've tested the MWI with the Uniden TRU-8866 phone and it works for me. I've tested it with the Digium TDM400P FXS. Steve ___ --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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --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] 5.8gig phone MWI
You're trying to teach a pig to sing. The uniden items you refer to probably have their own internal answering machine, mine does. It's designed to light the lamp only when it's own machine has a message. On 12/8/06, Doug Crompton [EMAIL PROTECTED] wrote: Thanks, but unfortunately that is an expensive 2 line phone compared to others in their line that have a base and two or three remotes for the same price. Seems a lot to pay for a MWI. I wonder if anyone has had experience with panasonic wireless 5.8gig and MWI?? They advertise compatibility on some models but I also saw a review comment that it did not work. Doug On Fri, 8 Dec 2006, Steve Prior wrote: Doug Crompton wrote: Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug I've tested the MWI with the Uniden TRU-8866 phone and it works for me. I've tested it with the Digium TDM400P FXS. Steve ___ --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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --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
