[asterisk-users] Call was hangup when LIMIT_WARNING_FILE was playing
Dear All,
I tried to use 'L' option on my dial command.
I set the x to 65000(65 seconds), y to 6(60 seconds), z to
3(30 seconds).
The max calltime should be 65 seconds, and it will play beep.gsm at
60 seconds left. And repeat the beep at 30 seconds left.
But the call will be hangup by system at 60 seconds left.
In another word, when it plays warning file, the call will be hangup.
The answeredtime is only 5 seconds.
Can anybody give me an idea for it?
*** extensions.conf ***
[default]
exten= _+[1-9].,1,SetCallerID()
exten= _+[1-9].,2,Set(LIMIT_WARNING_FILE=beep)
exten= _+[1-9].,3,Set(LIMIT_TIMEOUT_FILE=beep)
exten= _+[1-9].,4,Dial(zap/g1/002${EXTEN:1}|60|L(65000:6:3))
exten= _+[1-9].,105,Hangup
Log from CLI
***
-- Seeding '24012100' at 61.217.XXX.XXX:8625 for 60
-- Accepting AUTHENTICATED call from 61.217.XXX.XXX:
requested format = ilbc,
requested prefs = (),
actual format = ilbc,
host prefs = (ilbc),
priority = mine
-- Executing SetCallerID(IAX2/24012100-2, ) in new stack
-- Executing Set(IAX2/24012100-2, LIMIT_WARNING_FILE=beep) in new stack
-- Executing Set(IAX2/24012100-2, LIMIT_TIMEOUT_FILE=beep) in new stack
-- Executing Dial(IAX2/24012100-2,
zap/g1/0028621|60|L(65000:6:3)) in new stack
-- Limit Data for this call:
-- - timelimit = 65000
-- - play_warning = 6
-- - play_to_caller= yes
-- - play_to_callee= no
-- - warning_freq = 3
-- - start_sound = UNDEF
-- - warning_sound = beep
-- - end_sound = beep
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/0028621
-- Zap/29-1 is proceeding passing it to IAX2/24012100-2
-- Zap/29-1 is ringing
-- Zap/29-1 answered IAX2/24012100-2
-- Hungup 'Zap/29-1'
== Spawn extension (default, +8621, 4) exited non-zero on
'IAX2/24012100-2'
-- Hungup 'IAX2/24012100-2'--
Best Regards
Charles
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Re: [asterisk-users] Asterisk and Cisco PRI gateway config
quite OT: do you have some info, about what platforms supports qsig decode? I found, that first supported in 12.4.(9)T, but I don't know, if only on 28xx or also in older routers (like 3660) with NM-HDV-E1... and what is the name for qsig decode feature? I compare features in two ci$co firmwares (12.4.9T vs. 12.4.4T) using feature navigator, but cannot find anything about new feature like qsig decode support PJ Yehavi Bourvine +972-8-9489444 wrote: In my case the Cisco did all the Q.sig work so Asterisk's Q.sig functionality was not used. __Yehavi: ___ --Bandwidth 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] peer-to-peer RTP trouble in SIP
23 feb 2007 kl. 09.52 skrev Michiel van Baak: Hey, We have asterisk 1.2.4 (old I know) with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to eachother. Our supplier told us they dont send SDP with the INVITE. Can this be the problem causing dect to dect calls to always use asterisk in the RTP path ? If they do not send SDP with the INVITE there will be no media at all in the call. Very simple. Can be that they do not support re-invites. If so, you should see an error message in the SIP communication. /O ___ --Bandwidth 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] CWI, call-limit and incominglimit
23 feb 2007 kl. 12.42 skrev Steve Davies: Hi, In older versions of asterisk I used to be able to use incominglimit=1 to effectively disable call waiting on a specific SIP channel (Where broken phones do not allow this on the handset itself) In 1.2.x this became call-limit=1, but this prevents the phone from opening a 2nd line in order to transfer a call using attended transfer. The WiKi suggests using SetGroup() etc, but this does not cater for the case where you are Dialling several different phones simultaneously. You can still set one call-limit for the user and another for the peer. The peer call-limit would be used to prevent call waiting and the user limit could be set to a reasonable level so the phone can do transfers. I _could_ dial a whole bunch of Local channels, each of which checked for an extension usage count, but the additional load and complexity in the dialplan seems a bit over-the-top to me, especially when there used to be a one-line solution to this. I also considered separate user and peer sections in sip.conf, but the hosts are dynamic, and there is no way to link the IP address of the peer to the user. Why is that an issue? The user authenticates on the incoming call, no IP address is needed since the auth is done on the From: header. /O ___ --Bandwidth 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] Dial() command h and H options for SIP channel
23 feb 2007 kl. 14.06 skrev ast guy: Hi, Just need to confirm whether dial() command provided options h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * work for SIP channels as well ? Yes, Asterisk is a multiprotocol Open Source PBX. Those functions work in all channels that support DTMF. I've seen phones where using the * in a dial string doesn't work, but during the call as DTMF it should work properly. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Asterisk use SIP info command
23 feb 2007 kl. 19.55 skrev Philipp Kempgen: Olle E Johansson wrote: 22 feb 2007 kl. 23.40 skrev Philipp Kempgen: Olle E Johansson wrote: 22 feb 2007 kl. 19.34 skrev Philipp Kempgen: I thought it might be useful to be able to ask Asterisk for the current SIP CSeq through the Manager API in order to send your own SIP messages during a call outside of Asterisk (for AOC, whatever). Each time you ask for the CSeq Asterisk should increment the value so it does not get out of sync. Anyone sharing my opinion? We might open a feature request. We're trying to keep the Asterisk architecture multiprotocol and do things in a uniform way from the dialplan. Things like this would certainly break that, since it is very SIP- specific. Better to implement needed functionality in Asterisk. Thanks for you reply. That's basically what you have said more than once on the bug tracker. :) Thanks. Then I know that at least one person has read and understood :-) (Sorry, but sometimes it feels like being alone out there on the tracker...) :) On the other hand people are waiting for quick solutions to blinking Snom lights and AOC without really caring for the whole picture. We do have a lot of support for blinking lamps - for devices, conferences, parking lots and now in trunk for anything. People refrain from using the trunk in a production environment. And as I can remember even the trunk does not address the Snom pickup problem. That's on of the things bristuff is popular for. ...and they should NOT use trunk in production. I was just giving some hints on where we are going :-) Marketing, you know. The call pickup problem is something I've been working on, but it hasn't been prioritized by any customer, so I haven't been able to focus on it for much time. AOC is a very european thing and I keep shouting about it when I'm in Huntsville, so they're aware of the problem. Great. :) For a european company it's like this: We have AOC now, can we have that with Asterisk? No. (Or at least not very easily, eg. without a patch) But you probably know that. Make sure Digium sales are aware of the need of AOC support in Europe. There are a few patches for AOC support in the bug tracker, please review them. I know SNOM has some proprietary extensions for AOC, but what's the state on other devices? Snom has this page in their Wiki: http://www.snom.com/wiki/index.php/Advice_of_charge_(AOC)_in_SIP But they don't really say whether they actually use this in their phones or if it's more like a working draft. Apart from http://tools.ietf.org/html/draft-garcia-sipping-etsi-ngn-p- headers-00#section-4.1 is there any other standard that I should be aware of? I have no idea. THanks for the pointer. Anyone that knows about the state of AOC in non-SNOM SIP phones? Other implementations of AOC in SIP? What about SIP/IMS ? /O ___ --Bandwidth 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] ReceiveText()?
24 feb 2007 kl. 03.15 skrev Yuan LIU: How do I receive text sent from SendText() application? Asterisk lists text capability, so SendText() is successful. But I don't see an application to actually use it. EyeBeam and several SIP phones does receive those messages. We need to make sure that the application and the parser supports UTF8 messages, as both SIP and IAX2 is standardized on UTF8 text messaging. /O ___ --Bandwidth 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] CWI, call-limit and incominglimit
Olle E Johansson wrote: 23 feb 2007 kl. 12.42 skrev Steve Davies: Hi, In older versions of asterisk I used to be able to use incominglimit=1 to effectively disable call waiting on a specific SIP channel (Where broken phones do not allow this on the handset itself) In 1.2.x this became call-limit=1, but this prevents the phone from opening a 2nd line in order to transfer a call using attended transfer. The WiKi suggests using SetGroup() etc, but this does not cater for the case where you are Dialling several different phones simultaneously. You can still set one call-limit for the user and another for the peer. The peer call-limit would be used to prevent call waiting and the user limit could be set to a reasonable level so the phone can do transfers. it can be also usefull to use 'limitonpeer' option in sip.conf with this, it would not be needed to define separate type=user and type=peer for each phone, instead define one type=friend and apply limitonpeers=yes ;limitonpeers=no; Apply all call limits (limit=) only to peers, never ; to users. This improves handling of call limits ; and device states in certain situations. The user part ; of a type=friend will still be affected by the call ; limit, but Asterisk will only use one object for ; counting the simultaneous calls. ___ --Bandwidth 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] Somebody can help me?
Dear friends, I'm just subsribed at the mailing list. My name is Andrea, and i'm an Italian student. For my final exam for the secondary school i want to develop a PBX and configure it for a little LAN (my school's LAN). Please somebody can show me the minimal request for a 5pcs lan?? I've a Pentium Intel 300MHz available, can i use that? O i've to search another strongher??? Please answer me... Andrea Ps: Excuse me for the bad English but i'm an Italian Student ___ --Bandwidth 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] Somebody can help me?
On Sat, Feb 24, 2007 at 11:31:38AM +0100, Andrea Seghezzi wrote: Dear friends, I'm just subsribed at the mailing list. My name is Andrea, and i'm an Italian student. For my final exam for the secondary school i want to develop a PBX and configure it for a little LAN (my school's LAN). Please somebody can show me the minimal request for a 5pcs lan?? I've a Pentium Intel 300MHz available, can i use that? O i've to search another strongher??? It should do for basic things (for testing). If you want to use compressed voice-over-IP calls and such, you'll see the CPU limitation biting. BTW: please use a meaningful subject for your messages. It helps getting them answered more effectively. A useful subject line for this message could have been: asterisk minimal requirements or is PII 300MHz enough for asterisk? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CWI, call-limit and incominglimit
24 feb 2007 kl. 11.07 skrev Pavel Jezek: Olle E Johansson wrote: 23 feb 2007 kl. 12.42 skrev Steve Davies: Hi, In older versions of asterisk I used to be able to use incominglimit=1 to effectively disable call waiting on a specific SIP channel (Where broken phones do not allow this on the handset itself) In 1.2.x this became call-limit=1, but this prevents the phone from opening a 2nd line in order to transfer a call using attended transfer. The WiKi suggests using SetGroup() etc, but this does not cater for the case where you are Dialling several different phones simultaneously. You can still set one call-limit for the user and another for the peer. The peer call-limit would be used to prevent call waiting and the user limit could be set to a reasonable level so the phone can do transfers. it can be also usefull to use 'limitonpeer' option in sip.conf with this, it would not be needed to define separate type=user and type=peer for each phone, instead define one type=friend and apply limitonpeers=yes ;limitonpeers=no; Apply all call limits (limit=) only to peers, never ; to users. This improves handling of call limits ; and device states in certain situations. The user part ; of a type=friend will still be affected by the call ; limit, but Asterisk will only use one object for ; counting the simultaneous calls. Well, yes. That option does not exist in 1.2, it's someting I have implemented in svn trunk. And in this particular case, different call limits on the user and the peer seemed useful. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Test
On Fri, 23 Feb 2007 20:44:36 + --[ UxBoD ]-- [EMAIL PROTECTED] wrote: On Fri, 23 Feb 2007 19:09:40 + --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I have just setup inbound SIP and wonder if somebody would be so kind as to test that it works for me, and that my firewall is setup okay. sip:[EMAIL PROTECTED] Thank you Thank you very much to westcomuk for leaving the message, and to others who have tested for me :) A question though is that is shows the respondent as sip:@my sip server and not the actual originating caller. Why would that be ? Thanks, This is now resolved. Does help if you enable callerid in zapata.conf ! ;) -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // SIP:[EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Analogue Phone Problems on a TDM11B
Hi,
I received my shiny card this morning but I am having problems getting a
analogue phone to work. In zapata.conf I have :-
Code:
[channels]
usecallerid=no
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echotraining=yes
immediate=no
context=internal
signalling=fxo_ks
channel = 1
context=incoming
signalling=fxs_ks
channel = 4
and for my extensions.conf I have :-
Code:
[incoming]
exten = s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1)
exten = s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)
exten = s,3,Answer()
exten = s,4,Dial(Zap/1,20)
exten = s,5,Playback(cybermog)
exten = s,6,VoiceMail([EMAIL PROTECTED],s)
exten = s,7,Hangup()
[outofhours]
exten = s,1,Answer()
exten = s,2,Playback(cybermog)
exten = s,3,VoiceMail([EMAIL PROTECTED],s)
exten = s,4,Hangup()
[internal]
include = outbound-local
[outbound-local]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
What happens is that if I dial the FXO module, it picks up the call and fails
to pass it to the analogue (zap/1) phone. If I just try picking up the analogue
receiver the phone just rings and rings.
What have I done wrong ?
--
--[ UxBoD ]--
// PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import
// Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8
// Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8
// SIP:[EMAIL PROTECTED]
--
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and is
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[asterisk-users] dial a pager and enter DTMF
Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly 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] Asterisk and Cisco PRI gateway config
do you have some info, about what platforms supports qsig decode? I found, that first supported in 12.4.(9)T, but I don't know, if only on 28xx or also in older routers (like 3660) with NM-HDV-E1... I assume that it is a hardware independent but I do not have the hardware to test... I see refernce to this command in 12.3T documentation, so it should be inside the normal 12.4 version. I moved to 12.4T due to TAC request before I tried the above command, so I did not test it on 12.4. __yehavi: ___ --Bandwidth 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] dial a pager and enter DTMF
I don't think you need the pipe in there. I've used this with the w option before, which adds a wait. Then continues .5 seconds later. RTM: http://www.voip-info.org/wiki-Asterisk+cmd+dial Try these: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212D12345678) or exten = s,2,Dial(SIP/TelaSip-gw4/5198881212www12345678) also, you were missing a right parenth. -Original Message- From: [EMAIL PROTECTED] on behalf of Supa Sent: Sat 2/24/2007 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dial a pager and enter DTMF Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly 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] cisco sip firmware update for cisco 7970
You can buy smartnet on a single phone for something like $8 a year. This will get you in legally. -Original Message- From: [EMAIL PROTECTED] on behalf of David Parcerisa Sent: Fri 2/23/2007 1:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] cisco sip firmware update for cisco 7970 I'm trying to buy the cisco firmware update but it seems that i cannot order online because I bought my 7970 on ebay. Is there any other chance to get this update? ... anyone can make me a favour and send it to me by email? 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 ___ --Bandwidth 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] ReceiveText()?
From: Olle E Johansson [EMAIL PROTECTED] Date: Sat, 24 Feb 2007 10:52:22 +0100 24 feb 2007 kl. 03.15 skrev Yuan LIU: How do I receive text sent from SendText() application? Asterisk lists text capability, so SendText() is successful. But I don't see an application to actually use it. EyeBeam and several SIP phones does receive those messages. We need to make sure that the application and the parser supports UTF8 messages, as both SIP and IAX2 is standardized on UTF8 text messaging. /O Thanks for the explanation, Olle. So I gather that Asterisk itself does not currently have an application or function to intercept text? Yuan Liu ___ --Bandwidth 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] dial a pager and enter DTMF
Thanks for your reply,but with both your suggestions, I get the following error: Called TelaSip-gw4/5198881212D12345678 -- Got SIP response 484 Address Incomplete back from 4.79.19.59 == No one is available to answer at this time (1:0/0/0) I aslo tried sticking a , in between the number and D command, however CLI reads it as a pipe On 2/24/07, Tim Connolly [EMAIL PROTECTED] wrote: I don't think you need the pipe in there. I've used this with the w option before, which adds a wait. Then continues .5 seconds later. RTM: http://www.voip-info.org/wiki-Asterisk+cmd+dial Try these: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212D12345678) or exten = s,2,Dial(SIP/TelaSip-gw4/5198881212www12345678) also, you were missing a right parenth. -Original Message- From: [EMAIL PROTECTED] on behalf of Supa Sent: Sat 2/24/2007 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dial a pager and enter DTMF Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly 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 ___ --Bandwidth 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] CWI, call-limit and incominglimit
Pavel Jezek wrote: it can be also usefull to use 'limitonpeer' option in sip.conf with this, it would not be needed to define separate type=user and type=peer for each phone, instead define one type=friend and apply limitonpeers=yes ;limitonpeers=no; Apply all call limits (limit=) only to peers, never ; to users. This improves handling of call limits ; and device states in certain situations. The user part ; of a type=friend will still be affected by the call ; limit, but Asterisk will only use one object for ; counting the simultaneous calls. I'm a little confused about the comments shown above, which I assume are from sip.conf. limitonpeers=yes would seem to imply that the limit= value would only apply to the peer portion of the sip user. But the included comments say, The user part of a type=friend call will still be affected by the call limit Those seem to be in conflict, but perhaps it's just my parser :-) Could someone clueful explain? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco sip firmware update for cisco 7970
Tim Connolly wrote: You can buy smartnet on a single phone for something like $8 a year. This will get you in legally. Any idea about how specifically to get such a contract? It is rumored to be pretty tricky. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] To use asterisk or proprietary hardware, that is the question
Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a set it and forget it type of thing like a lot of small business proprietary systems. At the same time they would like to be able to do minor dial plan changes themselves so I have determine that a GUI like FreePBX or similar alternative (free or commercial) is appropriate. I have some concerns about using Asterisk for this. As much as I am in support of the whole Asterisk revolution, I just do not feel confident enough in Asterisk on a Hard Drive as a set it and forget it setup running month after month, year after year. I am hoping someone can convince me otherwise. I'm concerned about hard drive corruptions/failures, memory leaks, software bugs etc. I have the budget to buy good quality hardware so if I was to go with Asterisk I would go industrial grade fanless computer, power conditioned UPS etc. I am not concerned about the reliability of most of the hardware. It's the hard drive and the software that runs on it that worries me. I will obviously use a mature stable Asterisk release and the most stable Linux version which I won't bother naming just to keep the discussion focussed. I have other Asterisk installs that went well but they were in environments where there were IT people around who were prepared to deal with some Linux administration and I could provide ongoing support for more major things. That is not the case here. Some of those sites have been running for months untouched, some needed some updates and reboots for various issues. I don't think this customer would look very favorably on me having to come in and add patches or have to reboot once a month or whatever. Their expection is the same as they would have with any other phone system that mounts on the wall and just works for years. I think that is a reasonable expectation. I am looking at putting in an Epygi proprietary VoIP system in instead. It is mostly hardware based although apparently runs Linux. It has a GUI, is supposedly plug and play most of the time, and most importantly, does not use a Hard Drive. I have heard good things about them so for arguments sake, let's assume voice quality, features, and the enduser experience are approximately the same as using an Asterisk/Analog FXO Card/hardware echo cancel solution. Flexibility, scalability, upgradeability are non-issues because the requirements are fixed. The Eqygi will end up costing a few hundred dollars more but for arguments sake let's assume cost's are approximately the same. Astlinux would work except it does not currently meet some key requirements (GUI, Sangoma Analog card support). Otherwise it would be a GREAT distribution for set it and forget it running without a Hard Drive IMHO. Anyways, I am hoping I can get enough positive feedback about set it and forget it experiences to convince me to use Asterisk/FreePBX instead of a more proprietary VoIP solution. Either way I will be using the same SIP phones so that is a non-issue as well. Basic Requirements are as follows: Wall Mount *6 local network SIP extensions *4 remote SIP extension over ADSL or cable *4 incoming analog phone lines in a hunt group *features such as auto attendant, voicemail to email, forward to pager for after hours emergency etc. Nothing too special Any help, advice, experiences etc. would be greatly 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] Accessible documentation vor blind users
If you are a Mac user, Apple's speech synthesis works well. Open the PDF in Preview, not Acrobat. Then choose 'Start Speaking' from the Services-Speech menu. I'm not sure what tools Windows has for reading PDF files though I expect there's something... Might want to ask on a Windows mailing list. Todd On Feb 24, 2007, at 1:53 AM, [EMAIL PROTECTED] wrote: Hi Hi Is there any accessible ocumentation, ie plain text or html, how to configure Asterisk. The book 'Asterisk: The Future of Telephony'' is availablly only as and pdf document and is thus unreadable for a blind user. Any pointers welcome. ___ --Bandwidth 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] CWI, call-limit and incominglimit
Brian Capouch wrote: But the included comments say, The user part of a type=friend call will still be affected by the call limit Those seem to be in conflict, but perhaps it's just my parser :-) Could someone clueful explain? I interpret this that asterisk _internally_ still counting calls for both user and peer, but actually limits calls only for peers... :-\ ___ --Bandwidth 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] Fax with T.38
Hi Bill, I'm in exactly the same boat with T.38 and OpenPBX. I too think the Cisco-T.38 Gateway is the most practical at this moment. Where are you on testing this and can you share the 3660 config? In researching the CIsco/voice, there is a TON of hardware options you need, or so it seems. Jon -Original message- From: Bill Gibbs [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 15:02:18 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [asterisk-users] Fax with T.38 Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. So my call flow will be PRI - Asterisk 1.2.x Out the 2nd PRI to the 3660 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 pass through to my ATA. I have the 3660 there to take the call via TDM and convert to T38. I only have a single PRI which is why I don't want to have to purchase other lines dedicated to a T38 faxserver, and this will give me the ability to use my DIDs already assigned. That's how I plan to set it up. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Jackson Sent: Wednesday, February 21, 2007 10:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax with T.38 Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Cheers, Ray PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. Thomas Deillon wrote: Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajnish Jain *Envoyé :* lundi, 19. février 2007 16:25 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, *Thomas Deillon* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine-
Re: [asterisk-users] ReceiveText()?
Olle E Johansson a écrit : 24 feb 2007 kl. 03.15 skrev Yuan LIU: How do I receive text sent from SendText() application? Asterisk lists text capability, so SendText() is successful. But I don't see an application to actually use it. EyeBeam and several SIP phones does receive those messages. IAX softphones also can display text messages and MozPhone has a chat functionnality . Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth 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] dial a pager and enter DTMF
is there a way to pipe the dial command with SendDTMF(123456) What I am trying to do is dial an extension and have it page a group of pagers with the same number. Saving a lot of time over dial each one manually by hand. ___ --Bandwidth 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] To use asterisk or proprietary hardware, that is the question
shadowym wrote: Astlinux would work except it does not currently meet some key requirements (GUI, Sangoma Analog card support). Otherwise it would be a GREAT distribution for set it and forget it running without a Hard Drive IMHO. Kristian is working with Sangoma to get wanpipe supported once again in Asterisk. There have been some recent (last few days) changes in svn that indicate we are very close to having this working. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk
Please post extensions.conf based on the error I am guessing that you didnt set up extensions.conf properly. - Original Message - From: Pedro Santos To: asterisk-users@lists.digium.com Sent: Friday, February 23, 2007 10:44 PM Subject: [asterisk-users] asterisk Hi i install Asterisk can register softphones on clients computers but when i make a call to a extencion this error apear Call Failed: not found in the asterisk machine i do commannd sip show peers and i can see the clients there can you help me thanks -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To use asterisk or proprietary hardware, that is the question
If the harddrive is the only thing you are concerned about... dont' be. Many proprietary phone systems use hard drives.Toshiba, Samsung, and Nortel to name a few run some of their features off hard drives. We had a Nortell system some years ago that ran the entire ACD system off a hard disk. Glad we got rid of that thing before the hard drive died! That was always a fear of mine... Now.. back to your issue. Setup a crontab to restart asterisk every night. Use a version of asterisk you know well (I like 1.2.6) and know is stable.Finally, setup RAID-5 on the hard drives. That way if one dies, you can still replace it without data loss. You aren't going to suffer corruption of anything, and if somehow you DO get corruption of a hard disk, I don't see why that couldn't happen to the flash of a fanless system, or a proprietary system. What the heck are you doing that you need fanless... and a system that can never have any maintence, anyway?Even a Nortel or Samsung may need maintence from time to time. On 2/24/07, shadowym [EMAIL PROTECTED] wrote: Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a set it and forget it type of thing like a lot of small business proprietary systems. At the same time they would like to be able to do minor dial plan changes themselves so I have determine that a GUI like FreePBX or similar alternative (free or commercial) is appropriate. I have some concerns about using Asterisk for this. As much as I am in support of the whole Asterisk revolution, I just do not feel confident enough in Asterisk on a Hard Drive as a set it and forget it setup running month after month, year after year. I am hoping someone can convince me otherwise. I'm concerned about hard drive corruptions/failures, memory leaks, software bugs etc. I have the budget to buy good quality hardware so if I was to go with Asterisk I would go industrial grade fanless computer, power conditioned UPS etc. I am not concerned about the reliability of most of the hardware. It's the hard drive and the software that runs on it that worries me. I will obviously use a mature stable Asterisk release and the most stable Linux version which I won't bother naming just to keep the discussion focussed. I have other Asterisk installs that went well but they were in environments where there were IT people around who were prepared to deal with some Linux administration and I could provide ongoing support for more major things. That is not the case here. Some of those sites have been running for months untouched, some needed some updates and reboots for various issues. I don't think this customer would look very favorably on me having to come in and add patches or have to reboot once a month or whatever. Their expection is the same as they would have with any other phone system that mounts on the wall and just works for years. I think that is a reasonable expectation. I am looking at putting in an Epygi proprietary VoIP system in instead. It is mostly hardware based although apparently runs Linux. It has a GUI, is supposedly plug and play most of the time, and most importantly, does not use a Hard Drive. I have heard good things about them so for arguments sake, let's assume voice quality, features, and the enduser experience are approximately the same as using an Asterisk/Analog FXO Card/hardware echo cancel solution. Flexibility, scalability, upgradeability are non-issues because the requirements are fixed. The Eqygi will end up costing a few hundred dollars more but for arguments sake let's assume cost's are approximately the same. Astlinux would work except it does not currently meet some key requirements (GUI, Sangoma Analog card support). Otherwise it would be a GREAT distribution for set it and forget it running without a Hard Drive IMHO. Anyways, I am hoping I can get enough positive feedback about set it and forget it experiences to convince me to use Asterisk/FreePBX instead of a more proprietary VoIP solution. Either way I will be using the same SIP phones so that is a non-issue as well. Basic Requirements are as follows: Wall Mount *6 local network SIP extensions *4 remote SIP extension over ADSL or cable *4 incoming analog phone lines in a hunt group *features such as auto attendant, voicemail to email, forward to pager for after hours emergency etc. Nothing too special Any help, advice, experiences etc. would be greatly 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] To use asterisk or proprietary hardware, that is the question
On Saturday 24 February 2007 6:48 pm, Matt wrote: Now.. back to your issue. Setup a crontab to restart asterisk every night. Use a version of Nonsense. Set up proper monitoring of system resources (memory is only one resource you should be watching) and help the community out if you're detecting memory leaks. restarting every night is bad bad bad. asterisk you know well (I like 1.2.6) and know is stable.Finally, setup RAID-5 on the hard drives. That way if one dies, you can still replace it Again, nonsense. software RAID1 is more than adequate, but personally I far prefer to use CompactFlash. There's absolutely no reason to have three+ drives in small office PBX; Hell I'd be hard-pressed to justify two (RAID1) in such an install. -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] Voicemeup @ 0.008 per minute USA /CAN
Just to let you all know we are offering 0.019 down to 0.008 automatic pricings on volume.. TDM termination/origination Unlimited SIP/IAX accounts g729/ulaw/alaw/gsm/etc 15 channels opened per account to start with Toll Free numbers / Local numbers Reseller rates Wholesale Rates Whitelabel -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth 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] Wildcard Testing
Greetings and I thank you in-advance, I have a system installed with both Trixbox 1.x and 2.x installed and I am trying to utilize a wildcard x100p but it does seem hang-up or release the line. I understand that the Wildcards are not recommended, but I need to get a test system up for evaluation. Here are the steps followed: Run genzaptelconf and red alarm appears in zttool Run again and it comes up ok Dial from SIP soft phone works great until hang up Card or asterisk does not seem to release the line. Any help would greatly be appreciated. /metta -Diagnostics [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. [EMAIL PROTECTED] ~]# Current Alarms: Red Alarm â â â âSync Source:Internally clocked â â â â âIRQ Misses: 0 â â â â âBipolar Viol: 0 â â â â âTx/Rx Levels: 0/ 0 â â â â âTotal/Conf/Act: 1/ 1/ 0 [EMAIL PROTECTED] ~]# lspci 00:00.0 Host bridge: Intel Corporation 82810E DC-133 GMCH [Graphics Memory Controller Hub] (rev 03) 00:01.0 VGA compatible controller: Intel Corporation 82810E DC-133 CGC [Chipset Graphics Controller] (rev 03) 00:1e.0 PCI bridge: Intel Corporation 82801AA PCI Bridge (rev 02) 00:1f.0 ISA bridge: Intel Corporation 82801AA ISA Bridge (LPC) (rev 02) 00:1f.1 IDE interface: Intel Corporation 82801AA IDE (rev 02) 00:1f.3 SMBus: Intel Corporation 82801AA SMBus (rev 02) 01:07.0 Communication controller: Motorola Wildcard X100P 01:0c.0 Ethernet controller: 3Com Corporation 3c905C-TX/TX-M [Tornado] (rev 78) [EMAIL PROTECTED] ~]# ___ --Bandwidth 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] To use asterisk or proprietary hardware, that is the question
At 11:53 AM 2/24/2007, you wrote: Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a set it and forget it type of thing like a lot of small business proprietary systems. There is no such thing as set and forget. Businesses change. They either grow or shrink, they don't stand still. They will add and remove phones. So they will call you at that time. Or are you expecting them to shop for their own phones on Ebay? At the same time they would like to be able to do minor dial plan changes themselves so I have determine that a GUI like FreePBX or similar alternative (free or commercial) is appropriate. We take a different approach. We don't want a GUI. We don't want the limits. We work with the business to design their dial plan. Then we write it. We do not give them a GUI because we don't want them making changes and then asking for support. We sell them a minor service agreement and remote in for any changes. We also handle professional voice recording and basic training on phone use. And we handle backups and service if needed. Once they understand that we can do that without a service call, they are quite receptive to the idea. Conventional PBXs come with service agreements so customers are used to that but surprised at the low cost from you. I have some concerns about using Asterisk for this. As much as I am in support of the whole Asterisk revolution, I just do not feel confident enough in Asterisk on a Hard Drive as a set it and forget it setup running month after month, year after year. I am hoping someone can convince me otherwise. Hard drives are reliable. But I have similar feelings so we are working on a flash solution. Were running it beta in our office right now. It only uses the hard drive for daily voicemail, boots from flash and runs from RAM. I'm concerned about hard drive corruptions/failures, memory leaks, software bugs etc. Conventional systems have bugs too. I have the budget to buy good quality hardware so if I was to go with Asterisk I would go industrial grade fanless computer, power conditioned UPS etc. You don't really need fanless. Make it cheap enough that it can easily be replaced. Like a $500 PC. I am not concerned about the reliability of most of the hardware. It's the hard drive and the software that runs on it that worries me. I will obviously use a mature stable Asterisk release and the most stable Linux version which I won't bother naming just to keep the discussion focussed. Asterisk is pretty darn stable. I have other Asterisk installs that went well but they were in environments where there were IT people around who were prepared to deal with some Linux administration and I could provide ongoing support for more major things. That is not the case here. Some of those sites have been running for months untouched, some needed some updates and reboots for various issues. I don't think this customer would look very favorably on me having to come in and add patches or have to reboot once a month or whatever. So do it from home. And how often do you really need to upgrade a minimal read only flash based system with no dev tools running from RAM? Does the latest kernel really matter? Their expection is the same as they would have with any other phone system that mounts on the wall and just works for years. I think that is a reasonable expectation. Agreed. And if it breaks, you replace it quickly and at a low cost. I am looking at putting in an Epygi proprietary VoIP system in instead. It is mostly hardware based although apparently runs Linux. It has a GUI, is supposedly plug and play most of the time, and most importantly, does not use a Hard Drive. I have heard good things about them so for arguments sake, let's assume voice quality, features, and the enduser experience are approximately the same as using an Asterisk/Analog FXO Card/hardware echo cancel solution. Flexibility, scalability, upgradeability are non-issues because the requirements are fixed. The Eqygi will end up costing a few hundred dollars more but for arguments sake let's assume cost's are approximately the same. Are you selling them service or passing them off to someone else? Who will set up and support Egypi? If you are servicing them then that is one more system that you have to learn, stock and support. If you don't stock it, can they afford to be down for a day or longer waiting for a replacement? Astlinux would work except it does not currently meet some key requirements (GUI, Sangoma Analog card support). Otherwise it would be a GREAT distribution for set it and forget it running without a Hard Drive IMHO. Anyways, I am hoping I can get enough positive feedback about set it and forget it experiences to convince me to use Asterisk/FreePBX instead of a
Re: [asterisk-users] To use asterisk or proprietary hardware, that is the question
Agreed. Monitor yes.But why let the system run.. only to find out it is going to go down after being up for 100 days?Yes, it should be able to run continually with no issues, but unfortunately asterisk seems to have memory leaks. 1.2.6 is the only one we've found that will run and run and run. Our phone system we don't restart, but our voip switches restart nightly because we can't have them go down.. so a 'restart when convenient' once a day is protective. On 2/24/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 24 February 2007 6:48 pm, Matt wrote: Now.. back to your issue. Setup a crontab to restart asterisk every night. Use a version of Nonsense. Set up proper monitoring of system resources (memory is only one resource you should be watching) and help the community out if you're detecting memory leaks. restarting every night is bad bad bad. asterisk you know well (I like 1.2.6) and know is stable.Finally, setup RAID-5 on the hard drives. That way if one dies, you can still replace it Again, nonsense. software RAID1 is more than adequate, but personally I far prefer to use CompactFlash. There's absolutely no reason to have three+ drives in small office PBX; Hell I'd be hard-pressed to justify two (RAID1) in such an install. -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
RE: [asterisk-users] Fax with T.38
I am waiting for the powers that be to get a dual port PRI card at this time. I think a dial-peer will only need to look similar to this on the Cisco: dial-peer voice 10 voip destination-pattern WHATEVER session protocol sipv2 session target ipv4:openpbx ip dtmf-relay sip-notify rtp-nte fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco Since that's basically what you need to do voice, all this adds is the T38 line. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jbebeau Sent: Saturday, February 24, 2007 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Fax with T.38 Hi Bill, I'm in exactly the same boat with T.38 and OpenPBX. I too think the Cisco-T.38 Gateway is the most practical at this moment. Where are you on testing this and can you share the 3660 config? In researching the CIsco/voice, there is a TON of hardware options you need, or so it seems. Jon -Original message- From: Bill Gibbs [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 15:02:18 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [asterisk-users] Fax with T.38 Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. So my call flow will be PRI - Asterisk 1.2.x Out the 2nd PRI to the 3660 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 pass through to my ATA. I have the 3660 there to take the call via TDM and convert to T38. I only have a single PRI which is why I don't want to have to purchase other lines dedicated to a T38 faxserver, and this will give me the ability to use my DIDs already assigned. That's how I plan to set it up. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Jackson Sent: Wednesday, February 21, 2007 10:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax with T.38 Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Cheers, Ray PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. Thomas Deillon wrote: Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajnish Jain *Envoyé :* lundi, 19. février 2007 16:25 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is
Re: [asterisk-users] dial a pager and enter DTMF
Buy a cap code from the paging provider and program that cap into the group of pagers that way when you page that cap code all of the pagers will trip. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Supa wrote: is there a way to pipe the dial command with SendDTMF(123456) What I am trying to do is dial an extension and have it page a group of pagers with the same number. Saving a lot of time over dial each one manually by hand. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000716-3, 02/23/2007 - 2/24/2007 9:12:09 PM ___ --Bandwidth 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] Sending SMS
Is there anyone sending SMS with Asterisk? -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email ___ --Bandwidth 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] dial a pager and enter DTMF
Supa wrote: Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) I think what you are missing is the timeout parameter. The dial command is... Dial(type/identifier, timeout, options) So you would want something like: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212|120|D(12345678)) which would give you a two minute timeout Also, you are indeed missing a right paren on the end, which I added in the line above. /edg ___ --Bandwidth 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] 1.4.0 spews garbage on CLI, crashes
Hi, I just installed asterisk 1.4.0 on my mac. I compiled from source with no issues. I installed the sample config files, and basically just added a register line to sip.conf (to register with a Free World Dialup account). Then I called my asterisk system from a different computer (using x-lite softphone on windows xp, registered to an ekiga.net account). Asterisk answers, and I can hear the prompts etc. However, after a while, (regardless of what I do with the demo menu options) asterisk starts spewing garbage to the console, then some error messages, then eventually dies. Here is a condensed output: snip startup messages Asterisk Ready. *CLI sip show registry HostUsername Refresh State Reg.Time fwd.pulver.com:5060 829262 105 Registered Sat, 24 Feb 2007 16:05:17 *CLI -- Executing [EMAIL PROTECTED]:1] Wait(SIP/5060-01828e00, 1) in new stack -- Executing [EMAIL PROTECTED]:2] Answer(SIP/5060-01828e00, ) in new stack -- Executing [EMAIL PROTECTED]:3] Set(SIP/5060-01828e00, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing [EMAIL PROTECTED]:4] Set(SIP/5060-01828e00, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing [EMAIL PROTECTED]:5] BackGround(SIP/5060-01828e00, demo-congrats) in new stack -- Playing 'demo-congrats' (language 'en') -- Executing [EMAIL PROTECTED]:6] BackGround(SIP/5060-01828e00, demo-instruct) in new stack -- Pla;ing 'demo-instruct' (language 'en') : v\243\250\330\252z[\344\207.\233\343\2224\333l^K^W\230S\307t\316IJ\303\254\253^S\263\264G$\2161[\327$\2037\212fVW$l *CLI VW$l;~en#DA;fblBYY5Bjd snip tons of garbage *CLI : S;\262\335T\214k\363\2204\233`\212J\341B\271\\242^F\373\260I\354)[EMAIL PROTECTED]'\326X.\371\222\327X\244\357\221$J\332^A\233\372\331\343$^L+L\362\333\204D\234\243Tt\325\227F\375\221\223\243G^_\211-\213\335\243I=,\354\\225d\263'\212I\341#\207oj\272\352\325^T\206\275D\343$\214\363\262KQ\233^D^K#u\334\225\352\242\355\\227\346[Feb 24 16:07:20] WARNING[777]: codec_gsm.c:140 gsmtolin_framein: Invalid GSM data (1) [Feb 24 16:07:20] WARNING[777]: translate.c:197 framein: gsmtolin did not update samples 0 [Feb 24 16:07:20] WARNING[777]: codec_gsm.c:140 gsmtolin_framein: Invalid GSM data (1) [Feb 24 16:07:20] WARNING[777]: translate.c:197 framein: gsmtolin did not update samples 0 snip tons more of this [Feb 24 16:07:23] WARNING[777]: codec_gsm.c:140 gsmtolin_framein: Invalid GSM data (1) [Feb 24 16:07:23] WARNING[777]: translate.c:197 framein: gsmtolin did not update samples 0 [Feb 24 16:07:23] WARNING[777]: codec_gsm.c:140 gsmtolin_framein: Invalid GSM data (1) [Feb 24 16:07:23] WARNING[777]: translate.c:197 framein: gsmtolin did not update samples 0 Bus error d66-183-116-232:/Users/swebster admin$ *** That's it, the process has died. Any ideas on the cause of this? Or solutions? Thanks, Scott ___ --Bandwidth 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] ERROR: relation cc_ui_authen does not exist
This is related to asterisk database and in the process of installing a2billing, am still in the install stages and not able to logon but know what the problem. When I create the database and try to verify it, this what I get a2billing= SELECT * FROM cc_ui_authen; ERROR: relation cc_ui_authen does not exist I am suppose to get this: a2billing= SELECT * FROM cc_ui_authen; userid | login | password | groupid | perms | confaddcust | name | direction | zipcode | state | phone | fax | datecreation +---++-+---+-+--+---+-+---+---+-+--- 2 | admin | mypassword | 0 | 1023 | | | | | | | | 2005-02-27 04:14:05.391501+02 1 | root | myroot | 0 | 1023 | | | | | | | | 2005-02-27 03:33:27.691314+02 (2 rows) made changes to pg_hba.conf made sure that it ends with local all all trust host all all 127.0.0.1 255.255.255.255 trust host all all localip 255.255.255.255 trust made changes also under connection section in postgresql.conf added: #tcpip_socket = True port = 5432 I also commented out #tcpip_socket = True becuase the postgres will not start when its set to true. But that is not the problem, problem is not being able to verify the database.___ --Bandwidth 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] Sellvoip configuration....Please Help!!!!
On Fri, Feb 23, 2007 at 09:11:18AM +0100, [EMAIL PROTECTED] wrote: hi guy, i have a problem, i have an sellvoip account and i want configure asterisk for outbound calls. Alas, the best sellvoip configuration, I eventually had to conclude, was not to use sellvoip. They have good quality service, which makes this even more frustrating, but they are woefully understaffed, and can take months -- yes months, not hours, not days, not weeks -- to respond to support requests and tickets.They really are a good value when they work, but I had to abandon them, because problems can appear and you have no idea when they will be fixed. ___ --Bandwidth 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] To use asterisk or proprietary hardware, that is the question
On 02/25/07 06:26 Darrick Hartman said the following: Kristian is working with Sangoma to get wanpipe supported once again in Asterisk. is there a reason why wanpipe stopped working with asterisk ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth 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] dial a pager and enter DTMF
From: Supa [EMAIL PROTECTED] Date: Sat, 24 Feb 2007 10:05:06 -0500 Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Indeed simply syntax issueS. First, the above is missing a right parenthesis as others pointed out. Second, you need to leave a blank timeout field if you don't use timeout. Third, check the syntax to call TelaSip-gw4; some providers use [EMAIL PROTECTED] to specify number 5198881212. So Dial(SIP/TelaSip-gw4/5198881212,,D(12345678)) or Dial(SIP/[EMAIL PROTECTED],,D(12345678)) I did a little test and D() tag sent DTMF correctly so it's the right tool for you. Hope this helps. Yuan Liu Any help would be greatly 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
AW: [asterisk-users] ReceiveText()?
Here is how you can send/receive text in the DialPlan using an AGI script: print STDERR 1. Testing 'sendtext'...; print SEND TEXT \hello world\\n; my $result = STDIN; checkresult($result); print STDERR 2. Receiving Text 'receivetext'...; print RECEIVE TEXT 3000\n; my $result = STDIN; checkresult($result); Greetz, Roland. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Olle E Johansson Gesendet: 24 February 2007 10:52 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] ReceiveText()? 24 feb 2007 kl. 03.15 skrev Yuan LIU: How do I receive text sent from SendText() application? Asterisk lists text capability, so SendText() is successful. But I don't see an application to actually use it. EyeBeam and several SIP phones does receive those messages. We need to make sure that the application and the parser supports UTF8 messages, as both SIP and IAX2 is standardized on UTF8 text messaging. /O ___ --Bandwidth 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: AW: [asterisk-users] ReceiveText()?
From: Roland Ndaka Fru [EMAIL PROTECTED] Date: Sun, 25 Feb 2007 07:45:57 +0100 Here is how you can send/receive text in the DialPlan using an AGI script: print STDERR 1. Testing 'sendtext'...; print SEND TEXT \hello world\\n; my $result = STDIN; checkresult($result); print STDERR 2. Receiving Text 'receivetext'...; print RECEIVE TEXT 3000\n; my $result = STDIN; checkresult($result); Greetz, Roland. That's cool. Thanks for the pointer, Roland. Gotta go back to test-agi again. Now, if only one can pass value back into dial plan... Yuan Liu -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Olle E Johansson Gesendet: 24 February 2007 10:52 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] ReceiveText()? 24 feb 2007 kl. 03.15 skrev Yuan LIU: How do I receive text sent from SendText() application? Asterisk lists text capability, so SendText() is successful. But I don't see an application to actually use it. EyeBeam and several SIP phones does receive those messages. We need to make sure that the application and the parser supports UTF8 messages, as both SIP and IAX2 is standardized on UTF8 text messaging. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
