[Asterisk-Users] Alarmreciver
Hi,Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate way.CheersAndrew ___ --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] iax2_poke_noanswer on IP change. Sometimes permanent.
I have 4 asterisk servers which are Friends and each one has an account for termination. A total of 5 peers each. Currently, the setup is as follows iax.conf= [FriendName] type=friend context=server_friend secret=donttell host=friend.dyndns.com qualify=750 = In the past i would also use a register = and set host=dynamic, but the registration would often timeout, sometimes never work between two particular servers, and didnt help maintain my peers after ip change. So for simplicity i follow the pattern above on all the servers, and i dont have to wonder why registration is timing out. When one of their IPs changed yesterday, that box became unreachable to the other 4. That's annoying but expected, and I guess I'll need to write a script to address that. Problem is that the servers also lost all its peers when its ip changed (at least it seemed to correspond with the change), including my provider. Reloading iax2 wouldn't fix it; restarting the server brought the provider and one friend back; stopping, waiting, and starting brought one more friend back; and one friend is still unreachable to it. Its important to note that the other 3 servers are doing fine with each other, and they can see the box who's ip changed after iax2 reload. I can't help but wonder: shouldn't the box who's ip changed still be able to find its friend? also, is there a better way of going about this than what I'm currently doing? i'm sure not everyone with a dynamic IP has this same problem. ___ --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] get no connection, very often, but not allways, why?
Hi, I have an ISDN phone connected to a hfc-s card. I use it to phone via an iax provider to foreign countries. Inside my country it works reliable, but to other country it happens very often that the other side hears ringing and before it can take the phone the line is dropped. What makes me wonder is that I hear no ringing at all. With asterisk -c I get this: Asterisk Ready. *CLI == Primary D-Channel on span 1 up for TEI 64 -- Accepting overlap voice call from '' to 'unspecified' on channel 0/2, span 1 -- Starting simple switch on 'Zap/2-1' -- Executing Dial(Zap/2-1, IAX2/user:password@sip.coco-connect.de/XXX) in new stack -- Called user:password@sip.coco-connect.de/XXX -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 -- Channel 0/2, span 1 got hangup -- Hungup 'IAX2/62.180.50.221:4569/1' == Spawn extension (extern, XXX, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' Does the line -- Channel 0/2, span 1 got hangup mean that the ISDN-Phone drops the line first? If yes, could it be, because the phone gets no ringtone a too long time? I use no timeout for this channel. Thanks for any help. Regards Gerald ___ --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] Any Polycom dealer willing to help?
Hi All, We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM 3.1.3 as per http://www.polycom.com/resource_center/1,,pw-492,00.html Can someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only old versions. Are there any kind Polycom authorized dealers who can help me? -- Eric ___ --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] Any Polycom dealer willing to help?
Eric, I have a copy of both. They are at my office. Send me an email directly and tomorrow I'll forward you a copy. - Gabe - Original Message - From: Eric Bishop To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, March 27, 2006 12:57 AM Subject: [Asterisk-Users] Any Polycom dealer willing to help? Hi All,We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM 3.1.3 as per http://www.polycom.com/resource_center/1,,pw-492,00.htmlCan someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only old versions.Are there any kind Polycom authorized dealers who can help me?-- Eric ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Polycom IP 301 is slow
The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. Denis. Hahaha, clearly this guy is on crack. (no offense) I have uploaded MP3s to my asterisk box and have it programmed to play them when I enter certain extensions. I use this to test the internet connection, but many times I'll just hit my speakerphone on my 501 and start bumpin' some Cry me a river from justin timberlake (no, im not gay...that is actually his one great song!) - Gabe ___ --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] Alarmreciver
At the risk of being redundant, VoIP and Alarm is known not to mix well. Some of the tones used by an alarm system do not behave in the same way as conventional DTMF. This will vary greatly based on the actual alarm format used (and there are at least thirty different formats.) I don't know the specifics, and there are few who do due to the proprietary nature of the industry. While I've never used the alarmreceiver application, I have used some of the best industry equipment available. Again, we just can't get VoIP to be reliable enough for alarm transmission. I really, really wish we could. I blame compression, as we have the exact same kinds of problems with digital cell service... Now, rather than just being a nay-sayer, let me refer you to the Bosch C900V2 device. It takes a signal from just about any panel and converts it into IP to be received by a Bosch receiver. It's a good fit for nearly everyone. Also, there's a good chance your alarm panel has an IP module. Call your installer and ask them what your options are. If they won't help you, go to www.ul.com (in the US at least) and start looking up listed Central Stations in your area. Call them and ask if they support your panel over IP. You can probably get monitoring from a UL-listed Central Station for a price you'll be willing to pay. There's a lot of competition out there. This money gets you 24-hour, redundant coverage, which is in my opinion worth it's weight in gold. If you need any alarm industry guidance, please feel free to contact me off-list. I could probably even refer you to a good CS if you'd like. Please don't misunderstand, I don't mean to detract from the alarmreciever application. There are some really interesting applications to be had, e.g.: 1) Your alarm panel calls asterisk, which notes the alarm zone and connects your cell phone to the nearest intercom device. 2) 'Who are you, and what do you want?' 3) You could then have the capability to cancel the alarm before the panel dials your central station. This would save on false alarm calls and could speed up response greatly. Also, if you failed to answer your cell, the CS would dispatch as normal. Bob McDowell From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Nowrot Sent: Monday, March 27, 2006 2:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Alarmreciver Hi, Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate way. Cheers Andrew ___ --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] Caller ID length
What is maximum length of name in caller ID? How much charters can I put and be sure it will work fine? -- Tomislav Parcina tparcina#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] AstCC
Hi,I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes.Any idea how to do?Thank you ___ --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] 7940 with Asterisk?
Skeeve Stevens wrote: I just picked up a Cisco 7940 from an Auction… and would like to use it on an Asterisk box. Can anyone give me a pointer where I should start so I can get it working? http://www.voip-info.org Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What codec extensions using now?
Yes. disallow=all allow=g723 Allow only g723 codec. roberto 2006/3/26, Mohammad Salaque [EMAIL PROTECTED]: Hello list,Another newbie question,.if I putdisallow=all andallow=g723my sip.cofdoes it mean thatextension could only communicate usingg723 ?bellow is one of my extension example [10112]username=10112type=friendsecret=xrecord_out=Adhocrecord_in=Adhocqualify=noport=5060nat=yeshost=dynamicdtmfmode=rfc2833disallow=allcontext=Office-lancanreinvite=no allow=g723thanksSalaque___--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 -- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED]For reliable and professional DNS, use DNS Made Easy! http://www.dnsmadeeasy.com/u/14989 ___ --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: Snom 360 problems
Detailed info about snom beta firmware can also be found at snom-wiki e.g. http://snom.com/wiki/index.php/Beta_Firmware#Release_Notes Regards, - Usman Tahir snom technology AG - Date: Sat, 25 Mar 2006 11:53:24 -0800 (PST) From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Snom 360 problems To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Fri, 24 Mar 2006, Usman Tahir wrote: For the conf on Xfer issue, use the latest beta http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin what's the changelog for 5.5.1b? -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] RE: Re: Best GUI for basic HostedPBX service
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Please stop send me email Best Regards, Mr.Peeramate Rochanasmita Project Manager/General Manager This message was sent to me? -- Tomislav Parcina tparcina#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] stop monitor on transfer
Anton Krall wrote:
Hi John, yes, Im using native transfer. What I do is use Monitor on the
dialplan of the extension that picks up the call coming from PSTN, so after
that, if the extension forward or transfers the call, monitor keeps
recording all thru the end of the call no matter where it is been
transferred to.
Hmmm. This is what I do:
XX,1,NoOp()
XX,2,MixMonitor(${UNIQUEID}.wav)
XX,3,Dial(SIP/201,15,jTt)
..
The call is then SIP transferred by the receptionist, and that's when
the recording ends.
I'll have a look at native transfer and see if that changes things !
jd
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[Asterisk-Users] Polycom 501 Output volume
Hi Guys Is there anyway to adjust the output volume on the Polycom 501? ___ --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: Snom 360 problems
5.5.1b is neither listed on the snom-wiki nor is any changelog for 5.5.1b listed. -Dan On Mon, 27 Mar 2006, Usman Tahir wrote: Detailed info about snom beta firmware can also be found at snom-wiki e.g. http://snom.com/wiki/index.php/Beta_Firmware#Release_Notes Regards, - Usman Tahir snom technology AG - Date: Sat, 25 Mar 2006 11:53:24 -0800 (PST) From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Snom 360 problems To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Fri, 24 Mar 2006, Usman Tahir wrote: For the conf on Xfer issue, use the latest beta http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin what's the changelog for 5.5.1b? -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 ___ --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] registration with different username
Hello, I am trying to register to the asterisk with different phone number, login and password. This is my setting in the sip.conf: [246079011] type=friend context=cisco secret=XXX host=dynamic username=tomas allow=alaw nat=yes canreinvite=no mailbox=246079011 but I get this reply: Mar 27 13:17:00 NOTICE[5144]: chan_sip.c:10889 handle_request_register: Registration from '246079011sip:[EMAIL PROTECTED]' failed for '195.122.204.149' - Username/auth name mismatch Anybody can help me with it? Thanks a lot Regards Toams ___ --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] Call Simulator
Guyz, I wanna test my asterisk load capability before going to production, anyone know is there any call simulator to test this thing? Thanks in advance, Voipman ___ --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] Transfer after group pick-up
I can't transfer call which was picked up with feature - group pick up. I'm running * 1.2.5. The problem is that asterisk doesn't hear that I have pressed #1 and doesn't play transfer sound for me. Regular phone calls I can transfer without problem. Can anybody check is this a BUG? -- Tomislav Parcina tparcina#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: Re: Cisco 7960 - Have to press a menu button to dial
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Absolutely right :) \ escapes the next character, so if you wants *69 to go through immediately, you'd put \*69 so that the * gets recognized as a digit. , returns the dialtone sound. When my users hit 9, they like to hear the dialtone still so they know they're dialing outside. You got . and * right. Never put a 0 timeout on * or nothing else will work right. Hope that helps. Yes, you were weary helpful :)) Thank you. -- Tomislav Parcina tparcina#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: Free g729
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... There is no such thing as a 'free' G.729 - The DSP Group has claimed and defended the Patents they hold against the algorithm and process. Please do not use Asterisk/Digium related resources to exchange this information - They are the liable party as they provide a licensed version of G.729 from DSPg. I though so. But he mentioned something, and I just wanted to be sure. -- Tomislav Parcina tparcina#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] Voicemail to Email
Could anyone provide me some link in order tovoicemail to email working, I believe I have to give SMTP settings but do not know where. Thx Voipman ___ --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] Voicemail to Email
Voicemail uses sendmail on your system. If your machine can send mails using sendmail, so will asterisk. Rudolf On 3/27/06, voipman [EMAIL PROTECTED] wrote: Could anyone provide me some link in order to voicemail to email working, I believe I have to give SMTP settings but do not know where. Thx Voipman ___ --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] Voicemail to Email
Its in vociemail.conf. If you built asterisk with a basic running config there should be examples in there. Dovid --- voipman [EMAIL PROTECTED] wrote: Could anyone provide me some link in order to voicemail to email working, I believe I have to give SMTP settings but do not know where. Thx Voipman ___ --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!? Tired of spam? Yahoo! Mail has the best spam protection around http://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
Re: [Asterisk-Users] iax limit question
I found a solution... I just has to enter an Answer
line and now it behaves as I wanted. Here is the
working code:
[inbound]
exten = 1234567,1,Set(GROUP()=limit)
exten = 1234567,2,GotoIf($[${GROUP_COUNT()}2]?103)
exten = 1234567,3,Dial(Zap/5Zap/6,25,tT)
exten = 1234567,4,Voicemail,u110
exten = 1234567,5,hangup
exten = 1234567,103,Answer
exten = 1234567,104,Playtones(busy)
exten = 1234567,105,Wait(5)
exten = 1234567,106,Hangup
Check for OUTBOUND_GROUP variable in
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
It provides interesting capability to set the amount of calls on the called
channel but also on the calling channel.
In this case you should not need Answer.
Benchev
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Re: [Asterisk-Users] Polycom 501 Output volume
MBIT Technologies wrote: Hi Guys Is there anyway to adjust the output volume on the Polycom 501? Yes. I did this over the weekend. Look in your Polycom sip.cfg for a line tx.digital.handset. I had to set mine to -6 before the levels came down within tolerance. There is one for handset and base as well. 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
[Asterisk-Users] Re: compiling Zaptel-1.2.4 on CentOS 4.3
I got past this by changing spinlock.h in the /usr/src/kernels/2.6.9-34.EL-x86_64/include/linux/ folder. (I am using 64bit kernel) I changed: #define DEFINE_RWLOCK(x) rw_lock_t x = RW_LOCK_UNLOCKED #define DEFINE_RWLOCK(x) rwlock_t x = RW_LOCK_UNLOCKED to: #define DEFINE_SPINLOCK(x) spinlock_t x = SPIN_LOCK_UNLOCKED // guentis changes #define DEFINE_RWLOCK(x) rw_lock_t x = RW_LOCK_UNLOCKED #define DEFINE_RWLOCK(x) rwlock_t x = RW_LOCK_UNLOCKED -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Mark Quitoriano [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi Guys, Im having a problem compiling zaptel 1.2.4 on CentOS 4.3, anyone encountered this problem before? Here's the error i got: make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel-1.2.4/zaptel.o /usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before zone_lock /usr/src/zaptel-1.2.4 /zaptel.c:384: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel-1.2.4/zaptel.c:384: error: incompatible types in initialization /usr/src/zaptel-1.2.4/zaptel.c:384: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:384: warning: data definition has no type or storage class /usr/src/zaptel-1.2.4/zaptel.c:385: error: syntax error before chan_lock /usr/src/zaptel-1.2.4/zaptel.c:385: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel-1.2.4/zaptel.c:385: error: incompatible types in initialization /usr/src/zaptel-1.2.4/zaptel.c:385: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:385: warning: data definition has no type or storage class /usr/src/zaptel-1.2.4/zaptel.c:188: warning: 'fcstab' defined but not used make[2]: *** [/usr/src/zaptel-1.2.4/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel-1.2.4] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9- 34.EL-i686' make: *** [linux26] Error 2 -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 ___ CentOS mailing list [EMAIL PROTECTED] http://lists.centos.org/mailman/listinfo/centos ___ --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: Best GUI for basic HostedPBX service
You are signed up to the list. If you want out go to http://lists.digum.com --- Peeramate @ SIPPhone Thailand [EMAIL PROTECTED] wrote: Please stop send me email Best Regards, Mr.Peeramate Rochanasmita Project Manager/General Manager SIPphone (Thailand) Co., Ltd. 644/19 Moo 1 Klong Kum, Bung Kum Bangkok Thailand 10230 SIP No.100888 SIP Call Center No.888 Tel. 0 2690 3999 Fax. 0 2690 3535 Mobile. 0 1423 1423 Email : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Website : www.sipphone.co.th -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Par?ina Sent: Monday, March 27, 2006 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Best GUI for basic HostedPBX service In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I'm looking for a web GUI to offer my end-users (Hosted PBX), and I thought I'd pick a few brains first. I'm not looking to configure the Asterisk server itself, VI works adequately for that. But I want to give Web access to as many of the following features: This is something I'm will need in few months. If you find anything, please let the group know. -- Tomislav Parcina tparcina#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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://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] automatic callback when busy
I'm trying to set up the following application: When a SIP extensions calls another one which is busy, the caller would be able to ask for an automatic callback: when the callee becomes available again, asterisk would ring both the caller's and the callee's phones and connect them when both parties answer. Has anybody done this before? (I tried to search the archs but couldn't find this yet.) Any suggestions for the best solution? Thanks a lot in advance! Regards, -Tamás ___ --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] Call Simulator
SIPPS is one, I would like to hear of others. Of course you could create a dialplan that loops calls in and out. Thanks, Steve Totaro http://www.asteriskhelpdesk.com From: voipman [mailto:[EMAIL PROTECTED] Sent: Monday, March 27, 2006 6:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Call Simulator Guyz, I wanna test my asterisk load capability before going to production, anyone know is there any call simulator to test this thing? Thanks in advance, Voipman ___ --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] registration with different username
--- Tomas Komarek [EMAIL PROTECTED] wrote: Hello, I am trying to register to the asterisk with different phone number, login and password. This is my setting in the sip.conf: [246079011] type=friend context=cisco secret=XXX host=dynamic username=tomas allow=alaw nat=yes canreinvite=no mailbox=246079011 but I get this reply: Mar 27 13:17:00 NOTICE[5144]: chan_sip.c:10889 handle_request_register: Registration from '246079011sip:[EMAIL PROTECTED]' failed for '195.122.204.149' - Username/auth name mismatch Double check the user id and pass. Seems that asterisk is rejecting for that reason. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://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
Re: [Asterisk-Users] tsu-600
we are thinking about replacing a median 1 pbx system, we have about 40 phone. i got 4 incoming pot lines (all the same number), i don't know if i can use one tsu600 port as a fxo (for the pots) and all the rest as fxs, or should i use a tdm400p with 4 fxo's (for the pots,inside the asterisk box) and two t100p (also inside the asterisk box) for each tsu600. also i just need to know that it can be do and someone else has already done it so if i get into trouble i'll have someone who might know the answer. tell me more about your setup. i would really like to just copy someone elses setup :) Chris Mason (Lists) wrote: mike webb wrote: i wrote previous about a setup i thought might work with asterisk and the tsu-600. no one replied, so i thought i would ask if anyone is using a tsu-600 with asterisk and if so how do you have it setup ?? ___ I have three working. The work fine except there is no callerid on the units I got. What else do you need? ___ --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] Alarmreciver
Hi,Thanks for so fast reply.Now, rather than just being a nay-sayer, let me refer you to the BoschC900V2 device. It takes a signal from just about any panel and converts it into IP to be received by a Bosch receiver.Is it possible to connect C900V2 with Asterisk, (did you do such a thing, did you try to do it), or you need this special Bosch receiver?Cheers Andrew ___ --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] Alarmreciver
The C900V2 only connects with Bosch receivers. In fact, all of the IP communicators in the industry are proprietary. There is a committee working towards a standard, but my understanding is that we still have a decent wait ahead of us. Bob McDowell From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Nowrot Sent: Monday, March 27, 2006 7:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Alarmreciver Hi, Thanks for so fast reply. Now, rather than just being a nay-sayer, let me refer you to the Bosch C900V2 device. It takes a signal from just about any panel and converts it into IP to be received by a Bosch receiver. Is it possible to connect C900V2 with Asterisk, (did you do such a thing, did you try to do it), or you need this special Bosch receiver? Cheers Andrew ___ --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: Cisco 7970
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Best bet is to get Asterisk Chan_Sccp http://chan-sccp.berlios.de/ 1.) setup your /etc/asterisk/sccp.conf with something like: 2.) setup lines 30/31 as a custom extension in astersik (i used amp) and had it dial SCCP/30 and SCCP/31 as needed 3.) setup /tftpboot config for SEPMAC.xml First thank you for your mail and instructions that you have provide to me. Now, I have done everything you said except and 3.) file name isn't SEPMAC.xml but SEPMAC.cnf.xml When Cisco 7970 boots up it looks for this on tftp. 27.3.2006 15:24 :TFTP Error from 10.0.0.175 requesting CTLSEP0016C87754CE.tlv : File does not exist 27.3.2006 15:24 :Sending SEP0016C87754CE.cnf.xml to (10.0.0.175) 27.3.2006 15:24 :Sent SEP0016C87754CE.cnf.xml to (10.0.0.175), 2312 bytes 27.3.2006 15:24 :TFTP Error from 10.0.0.175 requesting loads : File does not exist 27.3.2006 15:24 :TFTP Error from 10.0.0.175 requesting td-sccp.jar : File does not exist 27.3.2006 15:24 :TFTP Error from 10.0.0.175 requesting g3-tones.xml : File does not exist 7970 had this firmware version: Load File: TERM70.DEFAULT App Load ID: Jar70.2-9-1-45.sbn JVM Load ID: CVM70.2-0-1-45.sbn OS Load ID: cnu70.2-7-5-50.sbn Boot Load ID: 7970_64060118.bin The problem is that 7970 never registers to *. I have entered asterisk IP address at 2 places in SEPMAC.cnf.xml. *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- MGCPMedia Gateway Control Protocol no yes no SIP Session Initiation Protocol (S yes yes yes Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Phone Standard Linux Telephony API D no no no Zap Zapata Telephony Driverno yes no Local Local Proxy Channel Driver no yes no IAX2Inter Asterisk eXchange Driver yes yes yes SCCPSkinny Client Control Protocol yes yes no Seams that SCCP is installed correctly. But phone never registers (on lover left corner of 7970 I have circle/clock and word Registering). *CLI sccp show devicestypes NAME ADDRESS MAC Reg. State === == *CLI Have I done something wrong? -- Tomislav Parcina tparcina#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] registration with different username
Well, I did, but the reason is still the same, if the username is different from the phone number, asterisk rejects the registration :-( Dovid Bender napsal(a): --- Tomas Komarek [EMAIL PROTECTED] wrote: Hello, I am trying to register to the asterisk with different phone number, login and password. This is my setting in the sip.conf: [246079011] type=friend context=cisco secret=XXX host=dynamic username=tomas allow=alaw nat=yes canreinvite=no mailbox=246079011 but I get this reply: Mar 27 13:17:00 NOTICE[5144]: chan_sip.c:10889 handle_request_register: Registration from '246079011sip:[EMAIL PROTECTED]' failed for '195.122.204.149' - Username/auth name mismatch Double check the user id and pass. Seems that asterisk is rejecting for that reason. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://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 ___ --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] Timeout waiting for response to Originate
Hello,I am using Asterisk-java, the Manager. And I have a problem I don't know howto sort it out!:Sometimes, when I send an OriginateAction my code receives an exception withthis message: Timeout waiting for response to OriginateI don't know what it means as Asterisk receives the action and then dials tothe telephone, might anybody show me what is the problem???Thanks in advance, -- María ___ --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] automatic callback when busy
I'm postponing this activity indefinitely but I collected some ideas.
Try something similar to this recipe:
First of all store dialed extension number as
exten = _[2-8]XX,102,SetVar(${UNIQUEID}=${EXTEN})
exten = _[2-8]XX,103,Goto(busyphone,s,1)
then you can use 3 options as press 3 for voice mail 6 for loop until free
and 9 for registering for automatic call back:
[busyphone]
;busy message voicemail and queue
exten = s,1,Answer()
exten = s,2,Wait(2)
exten = s,3,DigitTimeout(2)
exten = s,4,ResponseTimeout(2)
exten = s,5,Background(/etc/asterisk/voice/pabx/mtl-busy)
exten = 3,1,VoiceMail(b${${UNIQUEID}})
exten = 6,1,Dial(SIP/${${UNIQUEID}},20,trS(1080))
exten = 6,2,Playback(/etc/asterisk/voice/pabx/mtl-unavailable)
exten = 6,3,Goto(outside,s,1)
exten = 6,102,Wait(5)
exten = 6,103,Goto(6,1)
exten = 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM})
exten = 9,2,Hangup
exten = i,1,Goto(outside,s,1)
exten = t,1,Goto(outside,s,1)
exten = T,1,Goto(outside,s,1)
Don't blam eme if there is some error.
--
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tamás Bondár
Sent: Monday, March 27, 2006 2:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] automatic callback when busy
I'm trying to set up the following application:
When a SIP extensions calls another one which is busy, the
caller would be able to ask for an automatic callback: when
the callee becomes available again, asterisk would ring both
the caller's and the callee's phones and connect them when
both parties answer.
Has anybody done this before? (I tried to search the archs
but couldn't find this yet.) Any suggestions for the best solution?
Thanks a lot in advance!
Regards,
-Tamás
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RE: [Asterisk-Users] stop monitor on transfer
Really?
Mmhh seems you got working what I want and I what you want.. Hehehe try
using monitor instead of mixmonitor.. Maybe there is a difference in apps.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|John Daragon
|Sent: Monday, March 27, 2006 4:56 AM
|Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: Re: [Asterisk-Users] stop monitor on transfer
|
|Anton Krall wrote:
| Hi John, yes, Im using native transfer. What I do is use Monitor on
| the dialplan of the extension that picks up the call coming
|from PSTN,
| so after that, if the extension forward or transfers the
|call, monitor
| keeps recording all thru the end of the call no matter where it is
| been transferred to.
|
|
|Hmmm. This is what I do:
|
|XX,1,NoOp()
|XX,2,MixMonitor(${UNIQUEID}.wav)
|XX,3,Dial(SIP/201,15,jTt)
|..
|
|The call is then SIP transferred by the receptionist, and
|that's when the recording ends.
|
|I'll have a look at native transfer and see if that changes things !
|
|jd
|
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|
|
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RE: [Asterisk-Users] Polycom 501 Output volume
what do you men adjust? (I guess you already tried the keys on the pad right)? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MBIT TechnologiesSent: Monday, March 27, 2006 4:57 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Polycom 501 Output volume Hi Guys Is there anyway to adjust the output volume on the Polycom 501? ___ --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] Any Polycom dealer willing to help?
If you purchased your phones from an authorizedreseller they shouId be able to provide this. Ican help you. Please contact me off list. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Eric Bishop [mailto:[EMAIL PROTECTED] Sent: Monday, March 27, 2006 3:58 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Any Polycom dealer willing to help? Hi All,We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM 3.1.3 as per http://www.polycom.com/resource_center/1,,pw-492,00.htmlCan someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only old versions.Are there any kind Polycom authorized dealers who can help me?-- Eric ___ --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] Alarmreciver
Quoting Andrew Nowrot [EMAIL PROTECTED]: Hi, Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate way. I've been using it for a couple of years and it works great. I've found that some SIP devices need to be set to In-Band for DTMF signalling. This will also force you to use G.711. It seems the contactID format is really picky about timing and some SIP devices seem to fiddle around with the timing when doing out of band DTMF. This message was sent using IMP, the Internet Messaging Program. ___ --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: Re: OT: Unblocking bloced CID
I have a regular PRI from our CLEC and I *do* get blocked numebrs.. the bit is set to tell me to hide the number. I definately (as the 'phone company') want to be getting all call data for tracing purposes, should we ever need it, but we can certainly honor that bit and not display the number. Further, and here is where the legal question comes in. Is it legal to 'unblock' to the end user, that blocked number?Personally I feel it SHOULD be.. after all it is my time I'm about to spend picking up a phone and talking to someone, I want to know who is about to come blasting out of my phone On 3/27/06, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... It's a toll free number. You can call it from anywhere and the costs of the call go on the callee not the caller. Thank you. -- Tomislav Parcina tparcina#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
FW: [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?
Actually, I have tested this here with an Aastra 9133i and an [EMAIL PROTECTED] server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. In another thread, I've seen a response that the GXP2000 does the same. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |mustardman29 |Sent: 24 March 2006 01:10 |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP realtime |working ? |Importance: High | |So your Polycom 501's will eventually re-subscribe and BLF |will eventually |start working again after a reboot using your patch? How long |will that |take? Is the time to re-subscribe something you can set on the phone? | |That would be quite acceptable to me if the phone eventually |re-subscribed |on it's own without requiring a reboot. What I am saying is |that my Aastra |9133i and Grandstream GXP2000 NEVER re-subscribe after a reboot with or |without the patch. I tried lot's of different settings to try make it |happen unless I am doing something wrong or not waiting long |enough for the |phones to re-subscribe. I must have tested it for at least 3 |hours and BLF |never came back. I confirmed it with the Asterisk CLI as well. | | -Original Message- | From: BJ Weschke [mailto:[EMAIL PROTECTED] | Sent: Thursday, March 23, 2006 5:34 PM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: Re: [Asterisk-Users] Re: Fw: anybody has SIP | realtime working ? | | On 3/23/06, mustardman29 [EMAIL PROTECTED] wrote: | Thanks BJ, | | I tried your patch and it worked fine for me so thank you | so much for | the effort. It is very much appreciated. Especially since | I am not | capable of coding myself. | | Unless I can get a total solution so that it just works no | matter if I | reload or reboot then it's not really a solution for me. I have to | either not implement BLF for install something other than Asterisk. | | Telling the client that all they have to do is reboot their phone | everytime BLF stops working is not the sort of impression |I want to | make. Yes, it will probably be rare if the system is rock | solid with | no nightly/weekly cron jobs to reboot at night and UPS'ed | etc. but a | phone system feature has to either just work always or not | be used at all IMHO. | | As far as I'm concerned, BLF simply does not work because | of this :(. | | | Well - here's the thing. Using the code/approach in 6047, | the Polycom and other devices never get the message that | appears to be the kiss of death for the subscription to go | away and not come back. Using this approach, it's certainly | true that immediately after a restart of Asterisk (not a | reload - reloads are fine now with this code) the | subscription will not work, but like registrations, | subscriptions expire and the phone will sign up again and the | subscription will get renewed and become active again after | restart. Unfortunately, unlike registrations, there's no | guick/tactical way for us to keep track and reseed a | subscription immediately after a restart as we do with registrations. | | -- | Bird's The Word Technologies, Inc. | http://www.btwtech.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] Bluetooth headset in handsfree mode with SJPhone or X-lite
Hi, After much searching I have found that it might be possible to get a bluetooth headset to answer/hangup with SJPhone or Xlite if the headset supports handsfree mode. My Toshiba bluetooth stack supports this but I have not been able to figure out how to enable it. Also Windows XP desktop bluetooth stack does not support handsfree but Windows CE does (go figure). Has anyone got handsfree mode working with a bluetooth headset? How about working with SJPhone or Xlite or some other SIP phone? For some reason the SJPhone when used with a bluetooth headset disconnects/reconnects bluetooth when the answer/hangup button is used on the headset (how the hell did that come about). Using a bluetooth headset with a SIP phone and asterisk would really help me by removing those pesky wires 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
Re: [Asterisk-Users] Polycom 501 Output volume
Anton Krall wrote: what do you men adjust? (I guess you already tried the keys on the pad right)? On my system, when you watch ztmonitor on a channel, it is maxing out the output volume, causing local side echo. Reducing the tx.digital.handset gain bring the graph down to an acceptable range and hopefully eliminate the echo (Waiting for reports). 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] RE: Re: OT: Unblocking bloced CID
On 3/27/06, Matt [EMAIL PROTECTED] wrote: I have a regular PRI from our CLEC and I *do* get blocked numebrs.. the bit is set to tell me to hide the number. I definately (as the 'phone company') want to be getting all call data for tracing purposes, should we ever need it, but we can certainly honor that bit and not display the number. Further, and here is where the legal question comes in. Is it legal to 'unblock' to the end user, that blocked number?Personally I feel it SHOULD be.. after all it is my time I'm about to spend picking up a phone and talking to someone, I want to know who is about to come blasting out of my phone I would say that if it's legal for you to get it, then it's *not* legal for you to display it, and if it is legal for you to display it, then it is not legal for your provider to send it to you. Read this: http://www.epic.org/privacy/caller_id/fcc_final.html On 3/27/06, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... It's a toll free number. You can call it from anywhere and the costs of the call go on the callee not the caller. Thank you. -- Tomislav Parcina tparcina#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 ___ --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: IAX Incoming/Outgoing
I could ask why it can't authenticate against the key, but we've already been there. So, if I have 5 asterisk systems, and I want to have a different key on each, and each system has a user and a peer section, and I have to use different usernames... oh boy... this sounds like a horrible mess. I've been using a setup of one user for incoming and many outgoing peers. I'm not sure what the other poster meant that you can't do this. It works just fine. One thing I'll mention, and maybe if the developers are reading they can comment if this has changed, but in 1.0.x, and versions of CVS up to at least 05/2005, changes to the users and peers in iax.conf would often require a full restart to take effect. I don't use RSA since my IAX links all go over IPSec tunnels, but here's what my users and peers look like: [iax-in] type=user secret= context=extensions trunk=no tos=0x04 disallow=all allow=gsm [ast551-out] type=peer secret= username=ast551 host=XX.XX.XX.XX qualify=1000 disallow=all allow=gsm trunk=no tos=0x04 [ast129-out] type=peer secret= username=ast129 host=YY.YY.YY.YY qualify=1000 disallow=all allow=gsm trunk=no tos=0x04 etc - Noah -Original Message- From: Joshua Colp [mailto:joshnet at nbnet.nb.ca] Sent: Saturday, March 25, 2006 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing It still needs to know the username so it knows what entry in iax.conf to use for that information, such as the key to use. Joshua Colp - Original Message - From: Douglas Garstang [mailto:dgarstang at oneeighty.com] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-users at lists.digium.com] Sent: Sat, 25 Mar 2006 15:15:24 -0400 Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing Why do I need a username at all if I am doing rsa authentication? Why doesn't it match against the key? -Original Message- From: Joshua Colp [mailto:joshnet at nbnet.nb.ca] Sent: Saturday, March 25, 2006 12:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing You do realize you're not sending along a username so it's using another method to try to discover the username you're trying to authenticate as on the server side? Apparently not. IAX2/username_to_use at peer_entry_to_use/extension at context Joshua Colp - Original Message - From: Douglas Garstang [mailto:dgarstang at oneeighty.com] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-users at lists.digium.com] Sent: Sat, 25 Mar 2006 14:55:28 -0400 Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing Well, I just tried your approach. I broke them all up into users/peers. Now it makes even LESS sense. The pbx1 system is connecting to the pbx2 system, and according to the iax debug, is sending a username of 'pbx3_in'. *lol* [pbx1_in] type=user auth=rsa inkeys=pbx1 context=global_pbx_transfer deny=0.0.0.0 permit=xxx.187.142.203 [pbx1_out] type=peer auth=rsa outkey=pbx1 host=pbx1.ipt.yyy.com [pbx2_in] type=user auth=rsa inkeys=pbx2 context=global_pbx_transfer deny=0.0.0.0 permit=xxx.187.142.204 [pbx2_out] type=peer auth=rsa outkey=pbx1 host=pbx2.ipt.yyy.com [pbx3_in] type=user auth=rsa inkeys=pbx3 context=global_pbx_transfer deny=0.0.0.0 permit=xxx.187.142.234 [pbx3_out] type=peer auth=rsa outkey=pbx1 host=pbx3.ipt.yyy.com Here's how I connect: exten = s-CHANUNAVAIL,1,Dial(IAX2/pbx2_out/[EMAIL PROTECTED],25,g) and here's the IAX debug: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 1 DCall: 0 [xxx.187.142.204:4569] VERSION : 2 CALLED NUMBER : 2944099 CODEC_PREFS : (ulaw|g729) CALLING NUMBER : 2944093 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Foo LANGUAGE: en CALLED CONTEXT : global_pbx_transfer FORMAT : 4 CAPABILITY : 65535 ADSICPE : 2 DATE TIME : 2006-03-25 11:54:36 hestia*CLI -- Called pbx2_out/2944099 at global_pbx_transfer Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 2 DCall: 1 [xxx.187.142.204:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 5ms SCall: 2 DCall: 1 [xxx.187.142.204:4569] AUTHMETHODS : 4 CHALLENGE : 129428696 USERNAME: pbx3_in WHAT THE HELL IS THIS DOING HERE? -Original Message- From: Brian Capouch [mailto:brianc at palaver.net] Sent: Saturday, March 25, 2006 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]
RE: [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite
Try replacing the XP Bluetooth stack with the widcomm drivers...google is your friend! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 27, 2006 6:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite Hi, After much searching I have found that it might be possible to get a bluetooth headset to answer/hangup with SJPhone or Xlite if the headset supports handsfree mode. My Toshiba bluetooth stack supports this but I have not been able to figure out how to enable it. Also Windows XP desktop bluetooth stack does not support handsfree but Windows CE does (go figure). Has anyone got handsfree mode working with a bluetooth headset? How about working with SJPhone or Xlite or some other SIP phone? For some reason the SJPhone when used with a bluetooth headset disconnects/reconnects bluetooth when the answer/hangup button is used on the headset (how the hell did that come about). Using a bluetooth headset with a SIP phone and asterisk would really help me by removing those pesky wires 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 This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Re: OT: Unblocking bloced CID
On Monday 27 March 2006 08:58, Matt wrote: Further, and here is where the legal question comes in. Is it legal to 'unblock' to the end user, that blocked number?Personally I feel it SHOULD be.. after all it is my time I'm about to spend picking up a phone and talking to someone, I want to know who is about to come blasting out of my phone That's what your own dialplan causing any blocked number to go directly to voicemail is for. If I'm not mistaken it *is* illegal for you as a service provider to NOT honour those presentation bits. -A. ___ --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] Small - Medium Billing Software needed
Hi, Our asterisk installation will be a man-in-the-middle providing local,long,international VOIP services to our customers and our asterisk will be connect via VOIP to international carriers. We use asterisk 1.2.5 with mysql in centos 4.2 Kernel 2.6 I have looked at astbill and it sounds interesting, but their forum seems dead (lack of activity or a lot of unanswered questions). Any other suggestions for some open source or commercial billing systems for small installations like mine? We are looking for (this list is not complete) 1.2.5 Asterisk support in realtime h323, gsm and g729 web configuration if possible Dynamic International Rate Table (Each customer can have his own price list using Brands) LCR web based if possible MySQL based (because we use realtime asterisk) prepaid / postpaid (but *not* interested in online credit card processing at this moment) Switchboard (Displays live status of users phones and ongoing calls) this message will be posted to the bussiness list. Thanks to all. -- ---Erick ___ --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: Re: OT: Unblocking bloced CID
Who it is legal for or not to display those numbers is not realy the point here, as in a law suit you will both (you and your provider) be held liable. But the law clearly states that the end user should NOT see that number if the number is blocked. On 3/27/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Monday 27 March 2006 08:58, Matt wrote: Further, and here is where the legal question comes in. Is it legal to 'unblock' to the end user, that blocked number?Personally I feel it SHOULD be.. after all it is my time I'm about to spend picking up a phone and talking to someone, I want to know who is about to come blasting out of my phone That's what your own dialplan causing any blocked number to go directly to voicemail is for. If I'm not mistaken it *is* illegal for you as a service provider to NOT honour those presentation bits. -A. ___ --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] Who hangup.
Hello people. I`m running asterisk 1.0.9. In a phone call, I want to know who hangup, the caller or the callee. It this posible? Thanks in advance. José Luis ___ --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] Caller ID length
For the US PSTN network the limit seems to be 15 characters. For Asterisk you can safely use 20 characters with most VOIP phones. MATT--- On 3/27/06, Tomislav Parčina [EMAIL PROTECTED] wrote: What is maximum length of name in caller ID? How much charters can I put and be sure it will work fine? -- Tomislav Parcina tparcina#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] after-queues
Hi, I have the following requirement.. after a customer is answered bya Queue, I want him to be redirected to another extensions, where an IVR would answer and ask for his opinion about the analyst who just solved his issue. Is there a way to redirect him automatically, or do I have to ask my agents to manually transfer the users to this IVR extension? Thank you Dov ___ --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] after-queues
On 3/27/06, Dov Bigio [EMAIL PROTECTED] wrote: Hi, I have the following requirement.. after a customer is answered by a Queue, I want him to be redirected to another extensions, where an IVR would answer and ask for his opinion about the analyst who just solved his issue. Is there a way to redirect him automatically, or do I have to ask my agents to manually transfer the users to this IVR extension? If the agent hangs up it's going to signal termination of the call, so they would currently need to transfer out to an extension prior to moving on. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Config File Management
I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is extensions.conf. All the other files, even sip.conf have subtle differences which preclude them from being the same file (binaddr for example). Thanks, 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
[Asterisk-Users] Inaudible voice and sleepy voice
What could have caused a system(on the same side of NAT on our LAN ) that have been working perfectly ie you can call and both parties can hear themselves very well to start having the problem described below (1) the caller can hear the other party very well ,but the other party hears cracked and sleepy voice .(2)the caller can hear the other party very well ,but the other party hears cracked and inaudible voice I will be expecting your reply Thanks Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --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] Background() App From AGI
I have the following python AGI script. I know it's been abstracted, but it's still pretty easy to see what's happening. self.agi.channelAnswer() self.agi.wait(1) self.agi.execCmd(background,enter-conf-call-number,) self.agi.execCmd(Read,confNum|||,) confNum = self.agi.getVar(confNum) I enter DTMF digits, and read the result with Read() while the sound file is still playing. I always lose the first digit. The docs aren't clear but it appears that Background() is designed to grab the first DTMF digit it sees. I don't want Background() to chomp my first DTMF digit! I want to read them all with Read(). How can I play a sound file, while still waiting for DTMF input and get all the DTMF digits entered? Thanks, 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] Config File Management
I'm using CVS. I only have one server right now. I use it on other clusters to sync files and it works for me.. On 3/27/06, Douglas Garstang [EMAIL PROTECTED] wrote: I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is extensions.conf. All the other files, even sip.conf have subtle differences which preclude them from being the same file (binaddr for example). Thanks, 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 ___ --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] Testing asterisk faxing functionality
I have asterisk with rxfax txfax modules.I want to test fax sendig and reciving in one asterisk instance, in extensions.conf I have : exten = 1234567,1,rxfax(/home/patryk/fax-new.tif|debug) exten = s,1,Dial(1234567) exten = s,2,txfax(/home/patryk/fax.tif|caller|debug) but I doesn't seem to work.But when I'm calling on this number I can hear fax tones. I can't find sip client with fax fuctionality for linux I think it would help with testing. ___ --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: Best GUI for basic HostedPBX service
FreePBX allows you to set up multiple companies as well as determine what level of access each user has. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina Sent: Sunday, March 26, 2006 10:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Best GUI for basic HostedPBX service In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I'm looking for a web GUI to offer my end-users (Hosted PBX), and I thought I'd pick a few brains first. I'm not looking to configure the Asterisk server itself, VI works adequately for that. But I want to give Web access to as many of the following features: This is something I'm will need in few months. If you find anything, please let the group know. -- Tomislav Parcina tparcina#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
RE: [Asterisk-Users] Voicemail to Email
Just remember that a lot of email systems don't accept email from unverifiable domains. If your using a domain for your Linux/Asterisk server that does not resolve to a public IP then you may not be able to receive voicemail to email. I know that Hotmail WILL work no matter what so try that first. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Monday, March 27, 2006 4:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail to Email Its in vociemail.conf. If you built asterisk with a basic running config there should be examples in there. Dovid --- voipman [EMAIL PROTECTED] wrote: Could anyone provide me some link in order to voicemail to email working, I believe I have to give SMTP settings but do not know where. Thx Voipman ___ --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!? Tired of spam? Yahoo! Mail has the best spam protection around http://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
Re: [Asterisk-Users] Testing asterisk faxing functionality
I was playing with the fax stuff over IP on Friday. Unless you're receiving faxes from a PSTN circuit, it doesn't work so well. Also, I don't think you can chain txfax and rxfax like that. When you hit the s,2 part, it's going to play the fax out to the handset you dialed from. You'll need something like hylafax to send the fax. And you probably want to call Dial(Local/[EMAIL PROTECTED]) to call a local extension.. On 3/27/06, patryk [EMAIL PROTECTED] wrote: I have asterisk with rxfax txfax modules.I want to test fax sendig and reciving in one asterisk instance, in extensions.conf I have : exten = 1234567,1,rxfax(/home/patryk/fax-new.tif|debug) exten = s,1,Dial(1234567) exten = s,2,txfax(/home/patryk/fax.tif|caller|debug) but I doesn't seem to work.But when I'm calling on this number I can hear fax tones. I can't find sip client with fax fuctionality for linux I think it would help with testing. ___ --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: Fw: anybody has SIP realtime working ?
I tried again and you are correct. It does work on the Aastra 9133i but takes about an hour with no way to change that that I can find. The GXP2000 happens a lot sooner. I think it can be configured on the GXP2000. Turns out the problem I had is that the Aastra 9133i does not resubscribe to an Xlite softphone extension for some reason. I turned off the software firewall on the PC but that didn't help. The extension only shows up on the 9133i after I reboot it. The GXP2000 does not have this problem. -Original Message- From: William Harrison [mailto:[EMAIL PROTECTED] Sent: Monday, March 27, 2006 6:05 AM To: asterisk-users@lists.digium.com Subject: FW: [Asterisk-Users] Re: Fw: anybody has SIP realtime working ? Actually, I have tested this here with an Aastra 9133i and an [EMAIL PROTECTED] server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. In another thread, I've seen a response that the GXP2000 does the same. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |mustardman29 |Sent: 24 March 2006 01:10 |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP realtime working |? |Importance: High | |So your Polycom 501's will eventually re-subscribe and BLF will |eventually start working again after a reboot using your patch? How |long will that take? Is the time to re-subscribe something you can set |on the phone? | |That would be quite acceptable to me if the phone eventually |re-subscribed on it's own without requiring a reboot. What I am saying |is that my Aastra 9133i and Grandstream GXP2000 NEVER re-subscribe |after a reboot with or without the patch. I tried lot's of different |settings to try make it happen unless I am doing something wrong or not |waiting long enough for the phones to re-subscribe. I must have tested |it for at least 3 hours and BLF never came back. I confirmed it with |the Asterisk CLI as well. | | -Original Message- | From: BJ Weschke [mailto:[EMAIL PROTECTED] | Sent: Thursday, March 23, 2006 5:34 PM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: Re: [Asterisk-Users] Re: Fw: anybody has SIP realtime | working ? | | On 3/23/06, mustardman29 [EMAIL PROTECTED] wrote: | Thanks BJ, | | I tried your patch and it worked fine for me so thank you | so much for | the effort. It is very much appreciated. Especially since | I am not | capable of coding myself. | | Unless I can get a total solution so that it just works no | matter if I | reload or reboot then it's not really a solution for me. I have to | either not implement BLF for install something other than Asterisk. | | Telling the client that all they have to do is reboot their phone | everytime BLF stops working is not the sort of impression |I want to | make. Yes, it will probably be rare if the system is rock | solid with | no nightly/weekly cron jobs to reboot at night and UPS'ed | etc. but a | phone system feature has to either just work always or not | be used at all IMHO. | | As far as I'm concerned, BLF simply does not work because | of this :(. | | | Well - here's the thing. Using the code/approach in 6047, the | Polycom and other devices never get the message that appears to be | the kiss of death for the subscription to go away and not come back. | Using this approach, it's certainly true that immediately after a | restart of Asterisk (not a reload - reloads are fine now with this | code) the subscription will not work, but like registrations, | subscriptions expire and the phone will sign up again and the | subscription will get renewed and become active again after restart. | Unfortunately, unlike registrations, there's no guick/tactical way | for us to keep track and reseed a subscription immediately after a | restart as we do with registrations. | | -- | Bird's The Word Technologies, Inc. | http://www.btwtech.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] Testing asterisk faxing functionality
You could always use System() to copy a call spool file to launch the outbound fax call. I don't really think a 3rd party app is necessary. -Corey On Mon, 27 Mar 2006, Gary Richardson wrote: I was playing with the fax stuff over IP on Friday. Unless you're receiving faxes from a PSTN circuit, it doesn't work so well. Also, I don't think you can chain txfax and rxfax like that. When you hit the s,2 part, it's going to play the fax out to the handset you dialed from. You'll need something like hylafax to send the fax. And you probably want to call Dial(Local/[EMAIL PROTECTED]) to call a local extension.. On 3/27/06, patryk [EMAIL PROTECTED] wrote: I have asterisk with rxfax txfax modules.I want to test fax sendig and reciving in one asterisk instance, in extensions.conf I have : exten = 1234567,1,rxfax(/home/patryk/fax-new.tif|debug) exten = s,1,Dial(1234567) exten = s,2,txfax(/home/patryk/fax.tif|caller|debug) but I doesn't seem to work.But when I'm calling on this number I can hear fax tones. I can't find sip client with fax fuctionality for linux I think it would help with testing. ___ --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 * This message has been scanned for viruses and dangerous content, and is believed to be clean. * This message has been scanned for viruses and dangerous content, 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] CLI Echo
Hello All: I used the Authenticate command against a list of 4 passwords, however is there anyway I can get these to echo in CLI for debugging purposes? My auth line looks like this: exten = s,2,Authenticate(/home/listofnumbers|[|a]) ___ --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] CLI Echo
Jeremy wrote: Hello All: I used the Authenticate command against a list of 4 passwords, however is there anyway I can get these to echo in CLI for debugging purposes? My auth line looks like this: exten = s,2,Authenticate(/home/listofnumbers|[|a]) show application NoOp ___ --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] Polycoms and hints
How does the hinting work on the polycoms? I've got a polycom set up with hinting, I can see when the shared line rings, but I can't tell if someone's on the line. Any suggestions? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Authorization by ip
Can somebody send me a config of how to authorize SIP client by IP? Sam ___ --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] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?
Can Somebody send a working instruction to me on how to install g729 and 9723.1? I could not open the http://aussievoip.com.au/tiki-index.php?page=G729-Install Thank you, Goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters (Asterisk) Sent: Saturday, January 21, 2006 11:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk Discussion Subject: Re: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? On Sat, January 21, 2006 22:10, MapsAir said: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? I tried to follow the instruction from http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ and http://aussievoip.com.au/tiki-index.php?page=G729-Install but I can't. I did it with [EMAIL PROTECTED] 1.5, but not 2.2 Working on it now... Will let you know how, if I succeed! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] SIP caller id
I am using some Cisco 7940s with the 8.0 CM SIP image on them, and was wondering if there is a way to have the caller ID display as just NAME number as opposed to NAME [EMAIL PROTECTED]. The way it currently is, the missed calls directory cant be dialed, and my users really want this feature. Is there any good way to strip the @ off of the caller ID? I dont see it being sent by asterisk is the reason Any help is appreciated! ___ --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] Polycoms and hints
On 3/27/06, Aaron Daniel [EMAIL PROTECTED] wrote: How does the hinting work on the polycoms? I've got a polycom set up with hinting, I can see when the shared line rings, but I can't tell if someone's on the line. Any suggestions? Shared lines still don't work with Asterisk on the polycoms. To get hinting working, you need to enable the presence feature in sip.cfg if it's not enabled already and then put in a buddy in your mac address-directory.xml file and then enable the buddy watch feature with bw1/bw in the contact for the one you want to watch. This should then cause the phone to subscribe to the Asterisk server for the hint. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] FreePBX AAH
Does anyone know if FreePBX can be installed on a Linux box that was built using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only due to my lack of Linux experience... Thanks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] Installing the none commercial intel g729 codecsinto [EMAIL PROTECTED] 2.2?
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ I think this give a pretty good how to on installing the g729 and 723. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ADEGOKE ARUNA Sent: Tuesday, March 28, 2006 12:55 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Installing the none commercial intel g729 codecsinto [EMAIL PROTECTED] 2.2? Can Somebody send a working instruction to me on how to install g729 and 9723.1? I could not open the http://aussievoip.com.au/tiki-index.php?page=G729-Install Thank you, Goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters (Asterisk) Sent: Saturday, January 21, 2006 11:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk Discussion Subject: Re: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? On Sat, January 21, 2006 22:10, MapsAir said: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? I tried to follow the instruction from http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ and http://aussievoip.com.au/tiki-index.php?page=G729-Install but I can't. I did it with [EMAIL PROTECTED] 1.5, but not 2.2 Working on it now... Will let you know how, if I succeed! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Re: Cisco 7970
Yes, my mistake in /tftpboot/SEPMAC.cnf.xml. Having said that, Please double check that you have set the line: permit=192.168.1.90/255.255.255.255 ; This device can register only using this ip address or in your case: permit=10.0.0.175 /255.255.255.255 ; This device can register only using this ip address in the /etc/asterisk/sccp.conf. The TFTP errors are related to automatically updating firmware, custom ringtones, and what I believe is SCCP encryption support which is not necessary. J Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Best bet is to get Asterisk Chan_Sccp http://chan-sccp.berlios.de/ 1.) setup your /etc/asterisk/sccp.conf with something like: 2.) setup lines 30/31 as a custom extension in astersik (i used amp) and had it dial SCCP/30 and SCCP/31 as needed 3.) setup /tftpboot config for SEPMAC.xml First thank you for your mail and instructions that you have provide to me. Now, I have done everything you said except and 3.) file name isn't SEPMAC.xml but SEPMAC.cnf.xml When Cisco 7970 boots up it looks for this on tftp. 27.3.2006 15:24 :TFTP Error from 10.0.0.175 requesting CTLSEP0016C87754CE.tlv : File does not exist 27.3.2006 15:24 :Sending SEP0016C87754CE.cnf.xml to (10.0.0.175) 27.3.2006 15:24 :Sent SEP0016C87754CE.cnf.xml to (10.0.0.175), 2312 bytes 27.3.2006 15:24 :TFTP Error from 10.0.0.175 requesting loads : File does not exist 27.3.2006 15:24 :TFTP Error from 10.0.0.175 requesting td-sccp.jar : File does not exist 27.3.2006 15:24 :TFTP Error from 10.0.0.175 requesting g3-tones.xml : File does not exist 7970 had this firmware version: Load File: TERM70.DEFAULT App Load ID: Jar70.2-9-1-45.sbn JVM Load ID: CVM70.2-0-1-45.sbn OS Load ID: cnu70.2-7-5-50.sbn Boot Load ID: 7970_64060118.bin The problem is that 7970 never registers to *. I have entered asterisk IP address at 2 places in SEPMAC.cnf.xml. *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- MGCPMedia Gateway Control Protocol no yes no SIP Session Initiation Protocol (S yes yes yes Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Phone Standard Linux Telephony API D no no no Zap Zapata Telephony Driverno yes no Local Local Proxy Channel Driver no yes no IAX2Inter Asterisk eXchange Driver yes yes yes SCCPSkinny Client Control Protocol yes yes no Seams that SCCP is installed correctly. But phone never registers (on lover left corner of 7970 I have circle/clock and word Registering). *CLI sccp show devicestypes NAME ADDRESS MAC Reg. State === == *CLI Have I done something wrong? -- Tomislav Parcina tparcina#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
Re: [Asterisk-Users] FreePBX AAH
Yes, you can. On 3/27/06, Jim Houser [EMAIL PROTECTED] wrote: Does anyone know if FreePBX can be installed on a Linux box that was built using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only due to my lack of Linux experience... Thanks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] FreePBX AAH
Worked fine for me. I did lose my MAINT link off the Portal, but I simply added it back. Thank you, Jyran Glucky Advisory Programmer BlueWare, Inc. Strategic HealthWare Solutions 3060 W. 13th Street Cadillac, MI 49601 Phone: (231) 779-0224 ext. 111 Fax: 231-779-1002 Skype: Jyran Glucky AIM: JyranGlucky mailto:[EMAIL PROTECTED] http://www.blueware.net DID YOU KNOW? BlueWare is the Grand Prize Winner of the 2005 IBM Beacon Award BEST DB2 (Document Management) Application Worldwide. BlueWare Market Share for Hospital Document Management Systems is in 25 states in the US. Does anyone know if FreePBX can be installed on a Linux box that was built using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only due to my lack of Linux experience... Thanks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] Bluetooth headset in handsfree mode with SJPhoneor X-lite
Hi, I am not having trouble with the bluetooth stack since the Toshiba stack has the headset profile which supports a subset of AT commands http://en.wikipedia.org/wiki/AT_command from GSM 07.07 for minimal controls including the ability to ring, answer a call, hang up and adjust the volume. The problem is getting the softphone to work with these AT commands so that the answer/hangup function will work from the bluetooth headset. Thanks wendell hamilton wrote: Try replacing the XP Bluetooth stack with the widcomm drivers...google is your friend! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 27, 2006 6:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite Hi, After much searching I have found that it might be possible to get a bluetooth headset to answer/hangup with SJPhone or Xlite if the headset supports handsfree mode. My Toshiba bluetooth stack supports this but I have not been able to figure out how to enable it. Also Windows XP desktop bluetooth stack does not support handsfree but Windows CE does (go figure). Has anyone got handsfree mode working with a bluetooth headset? How about working with SJPhone or Xlite or some other SIP phone? For some reason the SJPhone when used with a bluetooth headset disconnects/reconnects bluetooth when the answer/hangup button is used on the headset (how the hell did that come about). Using a bluetooth headset with a SIP phone and asterisk would really help me by removing those pesky wires 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 This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Unicall Question
Hi I have a litle question, what is then version stable, in the web server i can see unicall version x.2.x and version x.3.x, and the time is same unicall-0.0.2e/ 11-Nov-2005 18:33 unicall-0.0.3pre8/ 11-Nov-2005 18:37Where i can find the change log or the diference from this versionsthanks ___ --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] FreePBX AAH
Pardon the question, but what I understand of FreePBX is that it's basically Asterisk with a web interface and some additional modules. Is that correct? Can you install FreePBX on a system which ALREADY has asterisk up and running or does it require ITS version of asterisk? Thanks, Waldo On Mar 27, 2006, at 12:29 PM, Tom Vile wrote: Yes, you can. On 3/27/06, Jim Houser [EMAIL PROTECTED] wrote: Does anyone know if FreePBX can be installed on a Linux box that was built using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only due to my lack of Linux experience... Thanks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] automatic callback when busy
OK, if I see well, this is the key idea here:
exten = 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM})
that is, putting the caller and callee number into AstDB under the CallBack
family.
Can you confirm that Asterisk takes care of the rest? If there is a record
like this in the database will it dial both extensions and connect them?
(Sorry, I've never heard such a feature.)
Thanks,
-Tamás
On Monday 27 March 2006 15.51, Mimmus wrote:
I'm postponing this activity indefinitely but I collected some ideas.
Try something similar to this recipe:
First of all store dialed extension number as
exten = _[2-8]XX,102,SetVar(${UNIQUEID}=${EXTEN})
exten = _[2-8]XX,103,Goto(busyphone,s,1)
then you can use 3 options as press 3 for voice mail 6 for loop until free
and 9 for registering for automatic call back:
[busyphone]
;busy message voicemail and queue
exten = s,1,Answer()
exten = s,2,Wait(2)
exten = s,3,DigitTimeout(2)
exten = s,4,ResponseTimeout(2)
exten = s,5,Background(/etc/asterisk/voice/pabx/mtl-busy)
exten = 3,1,VoiceMail(b${${UNIQUEID}})
exten = 6,1,Dial(SIP/${${UNIQUEID}},20,trS(1080))
exten = 6,2,Playback(/etc/asterisk/voice/pabx/mtl-unavailable)
exten = 6,3,Goto(outside,s,1)
exten = 6,102,Wait(5)
exten = 6,103,Goto(6,1)
exten = 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM})
exten = 9,2,Hangup
exten = i,1,Goto(outside,s,1)
exten = t,1,Goto(outside,s,1)
exten = T,1,Goto(outside,s,1)
Don't blam eme if there is some error.
--
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tamás Bondár
Sent: Monday, March 27, 2006 2:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] automatic callback when busy
I'm trying to set up the following application:
When a SIP extensions calls another one which is busy, the
caller would be able to ask for an automatic callback: when
the callee becomes available again, asterisk would ring both
the caller's and the callee's phones and connect them when
both parties answer.
Has anybody done this before? (I tried to search the archs
but couldn't find this yet.) Any suggestions for the best solution?
Thanks a lot in advance!
Regards,
-Tamás
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RE: [Asterisk-Users] FreePBX AAH
My understanding is you can install it on any Linux server running Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Monday, March 27, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FreePBX AAH Pardon the question, but what I understand of FreePBX is that it's basically Asterisk with a web interface and some additional modules. Is that correct? Can you install FreePBX on a system which ALREADY has asterisk up and running or does it require ITS version of asterisk? Thanks, Waldo On Mar 27, 2006, at 12:29 PM, Tom Vile wrote: Yes, you can. On 3/27/06, Jim Houser [EMAIL PROTECTED] wrote: Does anyone know if FreePBX can be installed on a Linux box that was built using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only due to my lack of Linux experience... Thanks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] automatic callback when busy
How can I edit the DB?
Tamás Bondár wrote:
OK, if I see well, this is the key idea here:
exten = 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM})
that is, putting the caller and callee number into AstDB under the CallBack
family.
Can you confirm that Asterisk takes care of the rest? If there is a record
like this in the database will it dial both extensions and connect them?
(Sorry, I've never heard such a feature.)
Thanks,
-Tamás
On Monday 27 March 2006 15.51, Mimmus wrote:
I'm postponing this activity indefinitely but I collected some ideas.
Try something similar to this recipe:
First of all store dialed extension number as
exten = _[2-8]XX,102,SetVar(${UNIQUEID}=${EXTEN})
exten = _[2-8]XX,103,Goto(busyphone,s,1)
then you can use 3 options as press 3 for voice mail 6 for loop until free
and 9 for registering for automatic call back:
[busyphone]
;busy message voicemail and queue
exten = s,1,Answer()
exten = s,2,Wait(2)
exten = s,3,DigitTimeout(2)
exten = s,4,ResponseTimeout(2)
exten = s,5,Background(/etc/asterisk/voice/pabx/mtl-busy)
exten = 3,1,VoiceMail(b${${UNIQUEID}})
exten = 6,1,Dial(SIP/${${UNIQUEID}},20,trS(1080))
exten = 6,2,Playback(/etc/asterisk/voice/pabx/mtl-unavailable)
exten = 6,3,Goto(outside,s,1)
exten = 6,102,Wait(5)
exten = 6,103,Goto(6,1)
exten = 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM})
exten = 9,2,Hangup
exten = i,1,Goto(outside,s,1)
exten = t,1,Goto(outside,s,1)
exten = T,1,Goto(outside,s,1)
Don't blam eme if there is some error.
--
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tamás Bondár
Sent: Monday, March 27, 2006 2:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] automatic callback when busy
I'm trying to set up the following application:
When a SIP extensions calls another one which is busy, the
caller would be able to ask for an automatic callback: when
the callee becomes available again, asterisk would ring both
the caller's and the callee's phones and connect them when
both parties answer.
Has anybody done this before? (I tried to search the archs
but couldn't find this yet.) Any suggestions for the best solution?
Thanks a lot in advance!
Regards,
-Tamás
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RE: [Asterisk-Users] FreePBX AAH
FreePBX is a configuration manager for Asterisk. It is NOT its own version of Asterisk, it is simply a GUI to manage the config files. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Monday, March 27, 2006 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FreePBX AAH Pardon the question, but what I understand of FreePBX is that it's basically Asterisk with a web interface and some additional modules. Is that correct? Can you install FreePBX on a system which ALREADY has asterisk up and running or does it require ITS version of asterisk? Thanks, Waldo On Mar 27, 2006, at 12:29 PM, Tom Vile wrote: Yes, you can. On 3/27/06, Jim Houser [EMAIL PROTECTED] wrote: Does anyone know if FreePBX can be installed on a Linux box that was built using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only due to my lack of Linux experience... Thanks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] * Meetme Freeze patch found
On Friday 24 March 2006 16:05, Benoit Panizzon wrote: Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. I can now confirm: No freezes/crashes anymore since I applied the patch. -Benoit- ___ --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] Searchable forums
Where can I do a keyword search of the posting in biz and users forums? asterisk.org just links to http://lists.digium.com/pipermail/and that doesn't let me do a string search across all postings. thanks, -- ---Erick PerezLinux User 376588http://counter.li.org/(Get counted!!!)Panama, Republic of Panama ___ --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] * Meetme Freeze patch found
I'm a bit newbie, could you tell me how to i apply the patch? Thanks in advance Marco Mouta On 3/27/06, Benoit Panizzon [EMAIL PROTECTED] wrote: On Friday 24 March 2006 16:05, Benoit Panizzon wrote: Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. I can now confirm: No freezes/crashes anymore since I applied the patch. -Benoit- ___ --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] Config File Management
Sorry for thread breaking... I'm on digest. I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? I'm using CVS. I only have one server right now. I use it on other clusters to sync files and it works for me.. Instead of doing this, I ended up creating a MySQL database and a few scripts to generate the config files for each of my servers. All I have to do is update the database, and the correct server pulls the information from the DB, generates the file, reloads, and sends reboot messages to the proper phones. Very specific to my needs, but extremely fast and effective. And all it requires on each Asterisk server is cron, PHP, and php-mysql. I had to customize a few of the variables inside the PHP scripts for each server, but by putting them close to the top, it's not a real big deal when I update the scripts to customize them for my servers. Mind you, I only have 4 servers on this system, but we don't anticipate growing beyond one more server for a while. One thing to mention that I have found: use lots of macros. Some of my macros require 6 or 7 arguments, but they are extremely flexible and trivial to generate on the fly through these tools. Each extension fits in only one line in the dialplan (calls a macro). Entries in the DB turn on and off features, sets the timeout, forwards to another extension or sends to voicemail, etc. Just what I'm doing. Hope it helps. 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] Receptionist Phones (was 3Com Phones)
Thanks for all the comments on the 3Com phones. Thankfully, there is a large number of phones out there to dig through looking for the right solution. What I have not been able to find, after spending all weekend looking, is a good solution for an attendant console. We have 2 receptionists that need to be able to view all 60+ phones (we could probably weed it down a bit if we had to, but would like to be able to cover all the phones) and see who is on the phone already. I would like to avoid a software solution as those tend to be confusing and hard for non-computer savvy people to deal with. I have seen that the polycom setup (601+sidecar) works but only for up to 7 phones. Does anybody have a recommendation for a solution for this? I find it hard to believe that nobody makes a compatible phone (or add-on) that is compatible with Asterisk. It seems like such a common thing. Daniel Hazelbaker ___ --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] * Meetme Freeze patch found
http://www.google.com/search?sourceid=navclientie=UTF-8rls=GGLD,GGLD:2 004-48,GGLD:enq=apply+patch+linux patch -p0 patch-file-name-here Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Marco Mouta [mailto:[EMAIL PROTECTED] Sent: Monday, March 27, 2006 1:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * Meetme Freeze patch found I'm a bit newbie, could you tell me how to i apply the patch? Thanks in advance Marco Mouta On 3/27/06, Benoit Panizzon [EMAIL PROTECTED] wrote: On Friday 24 March 2006 16:05, Benoit Panizzon wrote: Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. I can now confirm: No freezes/crashes anymore since I applied the patch. -Benoit- ___ --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] automatic callback when busy
On Monday 27 March 2006 20.07, Daniel wrote: How can I edit the DB? This may be a starting point for you: http://www.voip-info.org/wiki/view/Asterisk+database Or the related section of the book Asterisk: TFOT http://safari.oreilly.com/JVXSL.asp?x=1mode=sectionsortKey=ranksortOrder=descview=sectionxmlid=0596009623k=20g=srchText=asteriskcode=h=0m=l=1j=listcatid=s=1b=1f=1t=1c=1u=1r=o=1n=1d=1p=1a=0page=0 ___ --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] Searchable forums
Hi Erick - Where can I do a keyword search of the posting in biz and users forums? asterisk.org just links to http://lists.digium.com/pipermail/ and that doesn't let me do a string search across all postings. I'm guessing you mean the mailing lists rather than the forums. If so, you can use google. Just use a search string like this: search terms site:lists.digium.com This will search all the digium mailing lists. If you want to search just one list, you can use this: search terms site:lists.digium.com/pipermail/asterisk-biz/ or search terms site:lists.digium.com/pipermail/asterisk-users/ - Noah ___ --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] Receptionist Phones (was 3Com Phones)
We would be interested in the same. We have had only limited success getting Snom's phones to do this. And, you're right, this is such a common thing, there MUST be something out there that can do the job. Darrell S. Long BestWeb Corporation Daniel Hazelbaker wrote: Thanks for all the comments on the 3Com phones. Thankfully, there is a large number of phones out there to dig through looking for the right solution. What I have not been able to find, after spending all weekend looking, is a good solution for an attendant console. We have 2 receptionists that need to be able to view all 60+ phones (we could probably weed it down a bit if we had to, but would like to be able to cover all the phones) and see who is on the phone already. I would like to avoid a software solution as those tend to be confusing and hard for non-computer savvy people to deal with. I have seen that the polycom setup (601+sidecar) works but only for up to 7 phones. Does anybody have a recommendation for a solution for this? I find it hard to believe that nobody makes a compatible phone (or add-on) that is compatible with Asterisk. It seems like such a common thing. Daniel Hazelbaker ___ --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] Testing asterisk faxing functionality
You could always use System() to copy a call spool file to launch the outbound fax call. I don't really think a 3rd party app is necessary. Could You explain this please? Or maybe some links to documentation and examples ? Thanks Patryk. ___ --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] Receptionist Phones (was 3Com Phones)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Hazelbaker Sent: Monday, March 27, 2006 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Receptionist Phones (was 3Com Phones) Thanks for all the comments on the 3Com phones. Thankfully, there is a large number of phones out there to dig through looking for the right solution. What I have not been able to find, after spending all weekend looking, is a good solution for an attendant console. We have 2 receptionists that need to be able to view all 60+ phones (we could probably weed it down a bit if we had to, but would like to be able to cover all the phones) and see who is on the phone already. I would like to avoid a software solution as those tend to be confusing and hard for non-computer savvy people to deal with. I have seen that the polycom setup (601+sidecar) works but only for up to 7 phones. Does anybody have a recommendation for a solution for this? I find it hard to believe that nobody makes a compatible phone (or add-on) that is compatible with Asterisk. It seems like such a common thing. Daniel Hazelbaker ___ --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 Have you looked that the flash operator panel? http://www.asternic.org/demo.html I know you mentioned not wanting a software solution because of confusion but I think that would be pretty easy to understand. Curt ___ --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] Polycom IP 301 is slow
On Sun, Mar 26, 2006 at 08:03:55PM -0600, Darrick Hartman said: Denis Galv?o - iSolve wrote: The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. WHAT! The Polycom phones that have speaker phone features (the 50x/60x) are great speaker phones. The 301 is not an speaker phone. It only has a listen-only hands free setup. In fact, the speaker phone is so good, most people can't tell that I'm on a speakerphone and are surprised when I tell them. I regularly use the phone both as a local conference phone and as a member in a conference with other people on speakerphones too. No issues at all, and great sound quality. The only conference phone I found that is better is one of the dedicated conference phones such as the 4000 or the old analog versions of it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2.6 and Zaptel 1.2.5 Released
The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.6 and Zaptel 1.2.5. Both of these releases include a number of important bug fixes, and users are encouraged to upgrade their systems when possible. See the included ChangeLog files for more details on what has been fixed. The releases are available on the Digium FTP servers as PGP signed tarballs and also as PGP signed patch files, to ease upgrading from the previous versions. The keys used to sign these files can be verified by using the keyserver at pgp.mit.edu. Thanks for your support of Asterisk and Zaptel! ___ --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 1.2.6 and Zaptel 1.2.5 Released
The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.6 and Zaptel 1.2.5. Both of these releases include a number of important bug fixes, and users are encouraged to upgrade their systems when possible. See the included ChangeLog files for more details on what has been fixed. The releases are available on the Digium FTP servers as PGP signed tarballs and also as PGP signed patch files, to ease upgrading from the previous versions. The keys used to sign these files can be verified by using the keyserver at pgp.mit.edu. Thanks for your support of Asterisk and Zaptel! ___ --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] Searchable forums
Superb replies. Thanks to Jon and Noah On 3/27/06, Noah Miller [EMAIL PROTECTED] wrote: Hi Erick - Where can I do a keyword search of the posting in biz and users forums? asterisk.org just links to http://lists.digium.com/pipermail/ and that doesn't let me do a string search across all postings.I'm guessing you mean the mailing lists rather than the forums.If so, you can use google.Just use a search string like this:search terms site:lists.digium.comThis will search all the digium mailing lists.If you want to search just one list, you can use this:search terms site:lists.digium.com/pipermail/asterisk-biz/orsearch terms site: lists.digium.com/pipermail/asterisk-users/- Noah___--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 -- ---Erick PerezLinux User 376588http://counter.li.org/(Get counted!!!)Panama, Republic of Panama ___ --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
