Re: [asterisk-users] Divitas
From: EdPimentl [EMAIL PROTECTED] Date: Sun, 27 May 2007 16:12:09 -0400 There will be a number of companies set to offer similar services. In 3 months we will have a 24 port SIP-GSM-SKYPE gateway -E On 5/27/07, Dean Collins [EMAIL PROTECTED] wrote: I was cleaning through some old IT magazines this long weekend when I came across a company called Divitas in the April 30th edition of Network Computing. I've never heard of them but has anyone else heard of them? Basically they have a call control appliance that can deliver centrally held up calls between not only GSM but also redirect the call to a wifi hotspot if you are in range. It seems like a neat concept that shouldn't necessarily be beyond the capabilities of Asterisk (apart from the fact that the end Win Mobile 5 / Symbian handset would need some type of client). Any thoughts? At $550 per seat looks an expensive way to transfer calls between networks but I've never seen another CPE piece of equipment that can do this. According to another IT magzaine, Divitas indeed uses Asterisk. But Divitas does not seem to be a pure CPE solution. That may be why they could charge a premium. Yuan Liu http://www.divitas.com/products Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). [image: Call Button]http://click.mexuar.com/webuser/click/7/userurl/Cognationhttp://click.mexuar.com/webuser/nojs/7/userurl/Cognation ___ --Bandwidth 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] Local SMS how-to.
From: Anselm Martin Hoffmeister [EMAIL PROTECTED] Date: Tue, 22 May 2007 13:41:43 +0200 Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player: Hello, i just want to activate SMS service between my asterisk local sip accounts and between asterisk and local sip accounts. How can i do this thin? Also i tried smsq to an account but all i obtained is a error message: ---Cut Here--- May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1: Permission denied, deleting May 22 13:09:37 WARNING[4829] pbx_spool.c: Failed to scan service '/var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1' ---And Here--- Is necessary supplementary settings in /etc/asterisk/extensions.conf and /etc/asterisk/sip.conf ? Is necessary special module? I checked apps_sms.so is already loaded. Thank you for your support guys. No special change in sip.conf required. I've transmitted SMS over local SIP channel and it's be quire reliable - over LAN. Yuan Liu The SMSq stuff is for landline-type SMS, like those that never became really popular here in Europe ;-) I do not know of any SIP hardphone that supports them, but regular analog and ISDN handsets behind a SIP-to-analog/ISDN gateway work for me. The point of this SMS transfer method is calling the destination handset with a certain callerid set (which differs between countries - whatever number the telco prefers to choose - this can also be configured in the phone). The phone will not ring but instead immediately answer the call and receive the short message at 1200bps whatever modem standard they chose to use. For sending SMS, the handset will call a similarly telco-provided number (premium-rate numbers here in Germany - maybe that is the reason for the lack of popularity of this service) and do that 1200bps talk. If you still think you can make use of it, make sure to call smsq with the user id that asterisk is running as. That _might_ already do the trick. If you do not get it running, ask again - I might have a working setup somewhere around ;-) Nevertheless, for me, landline SMS is a PITA. The only great thing is you can upload Ringtones to Siemens gigaset phones. BR Anselm ___ --Bandwidth 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] xten will not send tones to * and i from sip phone
From: pedro noticioso [EMAIL PROTECTED] Date: Fri, 18 May 2007 20:36:59 -0700 (PDT) hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some sort of configuration out there to tell the xten softphone to work as expected? thanks! Seems to be a mismatch in dtmfmode between Asterisk and Xten. You may try dial ***7469 in Xten to bring up a magic menu. force_send_inband is enabled by default - not sure if this has any bearing, though, because my Xlite works with my Asterisks by default. Then another problem! I used the i extension, plus _X and _X. to make sure I catch everything that is not propperly dialed. If I take the regular phones that are connected through the sipura ata, then dial 'exten = 700,1,Goto(default,s,1)' so that I get the asking for an extension to reach, I dial a wrong number and walla, its caight by one of my magic numbers! BUT, if I pickup the same phone, and just dial the same wrong number? I just get a busy signal! and there is nothing registered at the CLI even though I added DEBIG to the configuration! :s Are you talking about the same context? (Specifically, does Sipura use [default]?) Snippets of sip.conf and extensions.conf would be helpful if you are not sure what to look. Yuan Liu What can I do to make sure I always send an error sound and never again a busy signal? 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] Call to an arbitrary outbound number by asterisk
From: Arpit Mehta [EMAIL PROTECTED] Date: Fri, 18 May 2007 02:31:22 -0400 Hi, I have a strange problem. I have a TE110p digium card. I want to dial 19173995791 when any incoming call comes in. What is happening is that when I dial 19173-995791. Asterisk picks up the first 5 digits assuming it is the extension and appends 212-85 (here in the university most numbers start with this) in front . Therefore I get connected to some random number 212-85-(19173) (where the voicemail is running). I cannot understand why asterisk is doing this whereas my dialplan says it needs to connect to other number exten = _.,1,Dial(Zap/g1/19173995791) Also any idea if this is an Asterisk problem or a telco problem. Any help/hints/suggestions would be most welcome If you are sure that your university doesn't have a PBX, that's a telco problem. Looks like that the switch has a dial plan that does not allow you to dial this sequence directly and interpret all dialed sequence as a local call. (This is usually the function of a PBX but ...) What is this number 19173995791, any way? (and what is 212-85?) If you attach a phone directly to a channel bank, would you be able to dial this sequence? Yuan Liu Here are my files. zapata.conf context=incoming switchtype=national signalling=pri_cpe group=1 channel=1-23 extension.conf [incoming] exten = _.,1,Dial(Zap/g1/19173995791) # I have added this line in the dialplan is because I want it to match the last 5 digit and simply dial the number 19173995791 such that a call leg is established between the calling party and the number 19173995791 CLI debug information -- Requested transfer capability: 0x00 - SPEECH -- Called g1/19173995791 -- Zap/1-1 is proceeding passing it to Zap/23-1 -- Zap/1-1 is making progress passing it to Zap/23-1 ### The call keeps ringing for sometime then it goes to voicemail. The message comes when the voicemail start. Note that I have not setup any voice mail -- Zap/1-1 answered Zap/23-1 ### Goes to the voicemail -- Native bridging Zap/23-1 and Zap/1-1 -- Channel 0/23, span 1 got hangup request -- Hungup 'Zap/1-1' == Spawn extension (incoming, 17689, 1) exited non-zero on 'Zap/23-1' -- Hungup 'Zap/23-1' Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth 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] Dealing with 2 SIP providers
From: Mike [EMAIL PROTECTED] Date: Fri, 11 May 2007 19:44:51 -0400 Yeah ok. That doesn't help. What I mean is I want a call to go out on ProviderA, UNLESS it's down and then go to ProviderB. ChanIsAvail() is supposed to allow this. Yuan Liu I want it to ring 30 seconds and then Hangup if nobody has answers. I DON'T want to dial both, only one or the other. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, May 11, 2007 17:03 To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Dealing with 2 SIP providers From: Mike [EMAIL PROTECTED] Date: Fri, 11 May 2007 11:06:35 -0400 Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten = 1234,1,Dial(SIP/providerA) exten = 1234,2,Dial(providerB) exten = 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a total of 60. I want to wait only 30 seconds before the hang up. Like put 15 seconds on each? It's quite hard to understand what exactly the requirements are. Yuan Liu Also, if ProviderA has a main server and a backup server, am I now forced to have 3 Dial commands, or can I setup ProviderA with host and backuphost in the same SIP entry? Mike ___ --Bandwidth 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] Confirmation key to answer -- for a queue
From: Yaakov Menken [EMAIL PROTECTED] Date: Sun, 13 May 2007 00:59:54 -0400 Hi, Pretty sure I'm missing something simple, but I've seen references to this feature but not found documentation for it: I have a queue set up so that many people are contacted (ringall) when a call comes in. I would like the answering party to confirm that he is a human being rather than cellphone voicemaill by pressing a digit. This is somewhat similar to the 2nd macro example found at http://www.voip-info.org/wiki-Asterisk+cmd+Dial Thought it would be chanspec 'c'. http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels Yuan Liu Is there a queues.conf option that I'm missing here? Thanks for any advice, Yaakov Menken ___ --Bandwidth 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] Dealing with 2 SIP providers
From: Mike [EMAIL PROTECTED] Date: Fri, 11 May 2007 11:06:35 -0400 Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten = 1234,1,Dial(SIP/providerA) exten = 1234,2,Dial(providerB) exten = 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a total of 60. I want to wait only 30 seconds before the hang up. Like put 15 seconds on each? It's quite hard to understand what exactly the requirements are. Yuan Liu Also, if ProviderA has a main server and a backup server, am I now forced to have 3 Dial commands, or can I setup ProviderA with host and backuphost in the same SIP entry? Mike ___ --Bandwidth 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 interruption
From: Andre Wangler [EMAIL PROTECTED] Date: Fri, 4 May 2007 07:35:38 +0200 Hello all Could someone tell me what happens with running calls when reloading the whole asterisk config files? I think SIP-calls are not Nothing. All calls are maintained according to documentation. Yuan Liu interrupted because of the protocol architecture (signalling vs. media) but what's with other kind of calls like h323 or over analogue interfaces? are they interrupted? I'm quite new with asterisk, so excuse this probably trivial question... Andre ___ --Bandwidth 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] Reinvite after DTMF?
From: Wilson Pickett [EMAIL PROTECTED] Date: Fri, 4 May 2007 11:37:41 +0200 Maybe I missed something here. In my understanding, the only parties in the call at DTMF stage are the originator and Asterisk. The destination is not in the picture yet. Is this correct? What is the purpose of the said DTMF sequence? Do you have a sample dial plan? No, the problem is to receive a call, to dial and send the DTMF to the new dialed number. The dial would normally then bridge the two While it is not possible to reinvite in the middle of a call (based on whatever event), I'm thinking more in the way of a workaround. Does this DTMF sequence absolutely have to be sent in the MIDDLE of the call or can it be sent at the beginning, i.e., before any conversation starts? Yuan Liu channels. I'm trying to figure out if there's a way to then remove asterisk from the RTP stream because of the needless distance (crossing the ocean twice is a waste). ___ --Bandwidth 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: Re[2]: [asterisk-users] Starting Asterisk on Ubuntu 7.04
From: James Texter [EMAIL PROTECTED] Date: Fri, 04 May 2007 12:28:39 -0500 If you do make config when compiling zaptel and asterisk, it should put the script in /etc/init.d, and add the relevant entries to the various start levels. Not with 1.4 at least. makefile is not looking in the right place and not the right script. Yuan Liu Thanks, James Texter On Fri, 2007-05-04 at 18:44 +0200, Christian wrote: Hi, I have already done: apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from the latest sources. So what should i do then? New to Ubuntu. many thanks, Christian On 2007-05-04 at 17:00 Tzafrir Cohen wrote: On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian apt-get install asterisk Look at the init.d scripts. Note that in Ubuntu, subdirectories under /var/run are deleted at boot, and hence that script generates /var/run/asterisk (with proper ownership) at boot time. -- 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] Reinvite after DTMF?
From: Wilson Pickett [EMAIL PROTECTED] Date: Thu, 3 May 2007 09:19:25 +0200 On 5/2/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Wilson Pickett [EMAIL PROTECTED] Date: Wed, 2 May 2007 15:30:21 +0200 Is there a way to do the following scenario? 1) my asterisk box receives an incoming call from a toll free number provider such as nufone, voicepulse, etc. 2) It then dials a number via SIP and outputs a DTMF sequence. At this point, I assume, the destination SIP has not been invited? The purpose of the DTMF is either determine which SIP destination to invite or to perform some other dial plan functions. ok, that part we do every day. 3) After DTMF though, is it possible to get the two SIP channels (original SIP caller plus SIP called) hooked together and have my pbx no longer in the call at all? tia If the above is true, then there shouldn't be a problem if all other conditions for reinvite are satisfied, because Asterisk will only execute Dial at this point, and that Dial could follow with reinvite. (I assume that the original SIP caller is in fact the toll free provider.) So what is in the dialplan once the DTMF is sent? The two channels are already bridged, how can asterisk then bow out? I don't see a way, Maybe I missed something here. In my understanding, the only parties in the call at DTMF stage are the originator and Asterisk. The destination is not in the picture yet. Is this correct? What is the purpose of the said DTMF sequence? Do you have a sample dial plan? Yuan Liu but I thought I'd ask if someone else did? ___ --Bandwidth 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] Get Asterisk to redirect a SIP INVITE
From: CSB [EMAIL PROTECTED] Date: Thu, 3 May 2007 21:51:02 +1200 I want to get Asterisk to redirect an incoming SIP INVITE to another SIP URI. I was looking at the Transfer application but it seems to You may want to elaborate the requirement. How is the incoming INVITE initiated? Is the originator a user in your system? Does the other URI represent a peer? etc. Yuan Liu be broken (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483). Is there an alternative way to do this on Asterisk 1.2.18? Regards Cameron ___ --Bandwidth 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] Called party identification - where to takecalledname?
From: Dan Austin [EMAIL PROTECTED] Date: Thu, 3 May 2007 10:01:25 -0700 Yehavi wrote: I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? Short answer is that you cannot. Longer answer is that it is possible, but requires new functionality to be added to the core and a new API call be added that can check if the called party is a local endpoint and retrieve the caller-id values. It will depend on actual application. For some small sites, manually setting up an AstDB family should suffice. This can even be semi automated. Yuan Liu At least that was what I found when working on the patch. If anyone knows a way to lookup a peer/friend from the dialplan and collect such details, it would be possible to use the existing patch without any more changes in the core. BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. 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
Re: [asterisk-users] using Playback() to play a random sound file
From: Steve Edwards [EMAIL PROTECTED] Date: Tue, 1 May 2007 22:08:10 -0700 (PDT) On Tue, 1 May 2007, Yuan LIU wrote: From: Steve Edwards [EMAIL PROTECTED] Date: Tue, 1 May 2007 21:10:40 -0700 (PDT) On Tue, 1 May 2007, Jay Austad wrote: I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? I do this with an AGI. In 1.4, there's also a dial plan function RAND(). Doesn't RAND return a random number? How will that help to play back a random file in a directory? Well, any randomness algorithm starts with a random number. In Asterisk extension language, though, translating number to a file name takes a bit lifting - meaning lifting by you, not by a system command or by a well published procedure. One way to do this is in dial plan is to use a static AstDB table. (AEL may have better ways but I don't know.) Yuan Liu Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth 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] Returning different SIP Hangup Cause
From: Kristian Kielhofner [EMAIL PROTECTED] Date: Wed, 2 May 2007 11:55:06 -0400 On 5/2/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi, I would like to return different values/cause to another SIP Server with Hangup cmd. I tried to put different values in Hangup(xx) ... but it always returns the same value ! How can I send back different error cause ? Thanks, Jean-Marc I'm actually interested in something like this too... What I'd like is a way to indicate different SIP errors manually (4xx, 5xx, 6xx) something like the OP: Obviously Hangup doesn't have this - from a feature stand point, Hangup being a channel agnostic application, introducing an error code may not be desirable. I can think of one workaround in channels that support SendText: use SendText before Hangup. Today, you'd have to use AGI to ReceiveText, but it's a more manageable pain. However, how to invoke this AGI from the origination side can be very challenging. I can't think of a way right now. Yuan Liu Hangup(513) etc, etc. Anyone have any ideas? -- Kristian Kielhofner ___ --Bandwidth 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] Reinvite after DTMF?
From: Wilson Pickett [EMAIL PROTECTED] Date: Wed, 2 May 2007 15:30:21 +0200 Is there a way to do the following scenario? 1) my asterisk box receives an incoming call from a toll free number provider such as nufone, voicepulse, etc. 2) It then dials a number via SIP and outputs a DTMF sequence. At this point, I assume, the destination SIP has not been invited? The purpose of the DTMF is either determine which SIP destination to invite or to perform some other dial plan functions. ok, that part we do every day. 3) After DTMF though, is it possible to get the two SIP channels (original SIP caller plus SIP called) hooked together and have my pbx no longer in the call at all? tia If the above is true, then there shouldn't be a problem if all other conditions for reinvite are satisfied, because Asterisk will only execute Dial at this point, and that Dial could follow with reinvite. (I assume that the original SIP caller is in fact the toll free provider.) 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] IVR dictionary dial-plan
From: Steve Kennedy [EMAIL PROTECTED] Date: Mon, 30 Apr 2007 19:33:43 +0100 Does anyone know of an (E)AGI or program to develop a IVR dial-plan which will take a list of words and then do something when a unique branch has been found. i.e. Say there's 3 words demon deacon bishop On a phone they'd be represented as 33666 332266 247467 So if the user enters 2 we know they want bishop if they enter 336 they want demon and 332 they want deacon. There was a similar discussion in the forum, http://forums.digium.com/viewtopic.php?t=14559. Don't seem to have a ready answer. Yuan Liu Could run the dictionary through a script which could generate the dial-plan or do it via some script interactively. Any help appreciated. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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] My Sip Provider lacks Sip 2.0 183 (Ringing)information
From: Knud Müller [EMAIL PROTECTED] Date: Tue, 01 May 2007 15:19:17 +0200 Hi all, my sip provider does'nt send a 183 Message when the opposite party rings. It sends the ringing indication on the audio stream. Is there any chance that the asterisk can analyze this audio stream (meta) information. I saw there is a zaptel configuration entry that sound pretty close to what I need 'callprogress'. Set progressinband to yes in sip.conf. Yuan Liu Has someone already solved this problem? Knud ___ --Bandwidth 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: Wildcard TDM11B Wildcard TDM04B
From: bilal ghayyad [EMAIL PROTECTED] Date: Tue, 1 May 2007 14:56:14 -0700 (PDT) Hi Noah; ut TDM11B contains physically 4 ports, if it supports only 1 FXS and 1 FXO, then what shall we do in the other two ports already existed? You can populate two more interface modules. Yuan Liu Regards Bilal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Playback() to play a random sound file
From: Steve Edwards [EMAIL PROTECTED] Date: Tue, 1 May 2007 21:10:40 -0700 (PDT) On Tue, 1 May 2007, Jay Austad wrote: I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? I do this with an AGI. In 1.4, there's also a dial plan function RAND(). Yuan Liu Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth 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] can�t anserd the call
From: Josu Lazkano Lete [EMAIL PROTECTED]
Date: Fri, 27 Apr 2007 10:09:56 +0200
hello, I have instaled a analog line, and when I call on the console apears
that:
I want to redirect the call to 101 extension.
*CLI -- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still failed so falling back to
context 'default'
[good stuff sniffed]
mi configuration files are this:
extensions.conf:
[general]
static=yes
writeprotect=yes
;autofallthrough=yes
;clearglobalvars=no
;priorityjumping=no
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten = 102,2,Hangup
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,Dial(SIP/101,30,Ttm)
[outgoing]
exten =_94XXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =_94XXX,2,Hangup()
exten =_94XXX,102,Hangup()
zapata.conf:
You have not specified a particular context for your Zap channels in
zapata.conf, so any call initiated from Zap would go to [default]. You also
specified that all SIP channels should use context [default]. However, you
haven't created a [default] context in extensions.conf.
So either create a [default], or change contexts used by Zap and SIP to
something you have in extensions.conf.
Yuan Liu
[channels]
signalling=fxs_ks
usecallerid=yes
callwaiting=no
threewaycalling=no
transfer=yes
cancallforward=yes
; valores validos 256(32ms),512(64ms),1024(128ms)
echocancel=yes
echotraining=yes
echocancelwhenbridged=no
rxgain=0
txgain=0
group=1
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
;busydetect=yes
;busycount=10
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
polarityonanswerdelay=600
;callprogress=no
progzone=es
channel = 1
zaptel.conf:
loadzone=es
defaultzone=es
fxsks=1
sip.conf:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[101]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME
[102]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME
thanks for all!!!
___
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[asterisk-users] chan_bluetooth as FXS?
Any way to use chan_bluetooth as FXS? 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] dialplan / problem with extension-length 1
From: Steve Davies [EMAIL PROTECTED]
Date: Wed, 25 Apr 2007 15:58:25 +0100
On 4/25/07, Barton Fisher [EMAIL PROTECTED] wrote:
Michael Kamleitner wrote:
[good stuff sniffed]
A very simple workaround to achieve what you want might be to replace
WaitExten(5)
with
Background(silence/5)
I use this all over the place to great success as it prevents the need
for any overly-clever processing of the result of WaitExten.
Cheers,
Steve
A question here. I usually only use timeout to wait for any input, e.g.,
exten = s,1,Answer
exten = _ZX,1,Dial(Zap/g2/${EXTEN})
exten = t,1,Hangup
exten = i,1,Hangup
Am I missing some functionality from WaitExten if I do not plan to do
anything special after timeout?
Yuan Liu
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Re: [asterisk-users] dialplan / problem with extension-length 1
From: Michael Kamleitner [EMAIL PROTECTED] Date: Wed, 25 Apr 2007 17:47:34 +0200 thx for all of your suggestions... I'm learning more about asterisk every minute :) Barton, I tried to replace 'WaitExten' with 'Background' as you suggested, and at first was disappointed that didn't change the behavior. Than I tried Roberts suggestion, using 'Read' instead of 'WaitExten' - and again was disappointed - no change, or at least it seem asterisk didn't get every digit fo the extension, just some of them (f.e. I entered 1234, asterisk [at the console] complained that there is no voicebox 124 etc.). however, I've continued to experiment again and again, and strangely it seemed to work _some_ times, even when passing 4digit-extensions. now I think I got the solution: it seems I have to press the extension digits a little bit longer! let's say I hold each button at least 0.5sec, everything works great. if I do a quick dial, asterisk seems to loose digits. any ideas why this might be? From which channel do you make the call? (Zap? SIP?) Looks like a DTMF detection problem. If ZAP, you better use longer tone. You can try relaxeddtmf in zapata.conf, but people generally recommend against it. The card you use also matters. Heavy echo could also interfere with DTMF. If SIP, the symptom you described would happen only to inband DTMF. Try not to use inband if you can help it. Yuan Liu looking forward to your opinions... I really start to like toying around with asterisk :) michael On 4/25/07, Steve Davies [EMAIL PROTECTED] wrote: On 4/25/07, Barton Fisher [EMAIL PROTECTED] wrote: A very simple workaround to achieve what you want might be to replace WaitExten(5) with Background(silence/5) I use this all over the place to great success as it prevents the need for any overly-clever processing of the result of WaitExten. ___ --Bandwidth 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] auto dial out multiple destinations
From: Vieri [EMAIL PROTECTED] Date: Tue, 24 Apr 2007 05:13:53 -0700 (PDT) --- Doug Lytle [EMAIL PROTECTED] wrote: Vieri wrote: However, Asterisk doesn't wait for the destination to pick the phone up, so the playback ends prematurely This has been discussed many times. Search the archives. If you are using standard POTS lines, then Asterisk sees the call as being answered immediately. Sorry I didn't search enough. And thanks for the reply. I guess I'll have to loop when using POTS. Someone on the forum just pointed out that the c chanspec in Zap channel could be used for call confirmation, may not require loop - http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels Hope this helps. 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] Funky BIND/named errors
From: Brett Crapser [EMAIL PROTECTED] Date: Tue, 24 Apr 2007 20:17:41 -0500 I have been getting these for awhile now in my log files. Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_loopback.so' (in'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_spool.so' (in 'so'?): 205.166.226.38#53 Looks unrelated to Asterisk. More like one of DNS servers used by Asterisk. Yuan Liu Anyone else or am I looking at doing some serious memory testing? Brett ___ --Bandwidth 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] Softphone that supports central provisioning?
From: Salvatore Giudice [EMAIL PROTECTED] Date: Sat, 21 Apr 2007 01:46:20 -0400 A complete provisioning system for soft phones could impart some of the same authentication models used for popular IM clients. Imagine a large enterprise who wants to give out several thousand soft phones to employees in a turnkey fashion requiring the employee's network credentials to authenticate at the start of each session. Generally, it is not acceptable to use employee credentials to perform SIP digest authentication. Employee credentials are meant for employees, not devices or software that sets up a session on behalf of an employee. The solution to this kind of setup is to use a soft phone that can be downloaded on demand and presents the employee with a simple username/password/domain login box. In one such system that I worked on, the client would take the credentials from the employee and authenticate via HTTPS to a simple CGI script that authenticates the credentials against an Active Directory setup. Once the employee is authenticated, the CGI script sets a temporary password in a database that is accessible by a radius server and sends back all the provisioning information including the employee's office number and the temporary session password via XML in the HTTPS POST response. The client then logs into the SIP service using the session credentials. Thought the OP wanted the name of a soft phone that was capable of using CGI or whatever mechanism to pull such provisioning info, or one that could be reconfigured on demand (outside of itself). I'd like to know which one(s), too. Wouldn't imagine pushing user credentials to end points. Yuan Liu The employee is required to re-authenticate at the start of each soft phone session or after a timed interval when the temporary session password is expired from radius. The advantages to this kind of setup are: 1.) you don't have employee credentials stored in soft phones 2.) you avoid locking out employee credentials when policy-based password changes are required because of rapid authentication failures from a SIP device with stored credentials 3.) no SIP service credentials are stored in the soft phones 4.) in the event that the temporary session password is stolen from a soft phone installation, it is only good for a short period of time usually limited to 12 hours 5.) HTTPS is a significantly better provisioning method than TFTP (cough Cisco...) because it is encrypted and you have the opportunity to validate a cert from the provisioning server to ensure that the soft phone client is talking directly to the provisioning server. Man in the middle attacks suck. 6.) it's a lot easier to change provisioning information for all clients without requiring employees to download a new soft phone with hardcoded settings or trying to get employees to implement changes on their phones manually. For the same reason, it reduces initial setup complexity and also eliminates the bulk of setup related support calls We have put together implementations of this kind of system before for clients. Usually, this kind of scenario is not something we discuss outside our training classes or at conventions. Generally, this kind of system is commonly requested by enterprise and government customers when they seek to extend their phone system to employees for road warrior, pandemic, disaster recovery, or occasional work at home scenarios. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, April 20, 2007 9:01 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Softphone that supports central provisioning? On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote: Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? Why would you want to do that? There are well-known and established tools to provision (centrally configure) software running on computers in a entwork. Why should the soft phones be configured any differently? What OS do you use on the desktops? ___ --Bandwidth 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 stops responding to SIP/ZAP
From: Ken Williams [EMAIL PROTECTED] Date: Fri, 20 Apr 2007 07:27:05 -0600 About once a week or so my Asterisk box stops responding to all phones. I can pull up the console, do whatever I want at the CLI but the only way to get things working again is to restart Asterisk altogether. I finally cranked verbose debugging way up (and watched my log files go from 1mb/day to 100mb/day), but below I believe contains my problem. The next line is 1.5 minutes later where I restart Asterisk. As a general troubleshooting procedure, you want to ask yourself if you have made any changes before it stopped working. If not, and especially if you can restart and get it working again, I'd suspect some hardware failure. (Assuming the problem is reproduceable - I had times when TDM card stopped working with no trace of error.) Try installing on another box. Yuan Liu SIP/701 is a Grandstream GXP-2000 phone (we have about 30 of them in place here). Zap/3-1 is a Digium TDM400. I can't quite figure out where my problem is, is it the initial exception, is it not getting hung up completely, does it have to do with the call limit on the SIP channel, perhaps 'no provider found' statements? Any help would be appreciated, I have a relatively simple dial-plan, I can send over relevant bits of it if necessary. Thanks, Ken [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Exception on 12, channel 3 [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Got event On hook(1) on channel 3 (index 0) [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on channel 3 [Apr 19 13:51:13] DEBUG[27722] channel.c: Didn't get a frame from channel: Zap/3-1 [Apr 19 13:51:13] DEBUG[27722] channel.c: Bridge stops bridging channels SIP/701-08ee6120 and Zap/3-1 [Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel 'Zap/3-1' [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: zt_hangup(Zap/3-1) [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Hangup: channel: 3 index = 0, normal = 12, callwait = -1, thirdcall = -1 [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on channel 3 [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/3-1 [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Updated conferencing on 3, with 0 conference users [Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Hungup 'Zap/3-1' [Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state change to be queued on device/channel Zap/3-1 [Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension (from-internal,201,2) exited non-zero on 'SIP/701-08ee6120' [Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension (from-internal, 201, 2) exited non-zero on 'SIP/701-08ee6120' [Apr 19 13:51:13] DEBUG[27722] pbx.c: Launching 'Hangup' [Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup(SIP/701-08ee6120, ) in new stack [Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension (from-internal,h,1) exited non-zero on 'SIP/701-08ee6120' [Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/701-08ee6120' [Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel 'SIP/701-08ee6120' [Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Hangup call SIP/701-08ee6120, SIP callid [EMAIL PROTECTED]) [Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Updating call counter for incoming call [Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Call from peer '701' removed from call limit 6 [Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state change to be queued on device/channel SIP/701 [Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state change to be queued on device/channel SIP/701-08ee6120 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for Zap - 3 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for Zap/3 - state 0 (Unknown) [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for peer 701 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701 - state 1 (Not in use) [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for peer 701 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for peer 701 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701 - state 1 (Not in use) [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for peer 701 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] Softphone that supports central provisioning?
From: Steve Davies [EMAIL PROTECTED] Date: Fri, 20 Apr 2007 18:26:57 +0100 On 4/20/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 4/20/07, Olivier [EMAIL PROTECTED] wrote: Are you sure eyeBeam config are binary ? I thought it was just the case for XLite. Having looked into it further, you're right. For some inexplicable reason it's not putting the files where the manual says they should be - instead of a directory called eyeBeam n.n they're in a folder called 'RegNow Basic', but the .CPS files there are indeed in XML rather than binary format. When I last looked, I suspect I assumed that eyeBeam stored it's configs in the X-Lite directory and was thus looking at the configs for the free version that were no longer being accessed. I went around this loop with CounterPath a couple of months back. It seems that their idea of provisioning revolves around customising the software before selling it, so that it is locking the end-user into using your (the seller's) SIP server. They had trouble understanding that the user just paid money for this software, which they want to be provisioned by a server on their own network, and they do not support this. I gave up at this stage, but That's because mainstream service providers only want a branded client that indeed locks users in. Unless a reasonably powerful commercial entity (or even freelance org) exerts pressure, individual users and small companies can't do much. Does a Web deployed client such as JAIN SIP applet count? Yuan Liu perhaps if more people apply pressure, it will become possible to extend their current (quite useable) provisioning interface, but have a user-configurable setting to determine where the configuration is fetched from. At present the configuration server setting is fixed at compile-time by CounterPath. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CallerID Auth
From: Arun Kumar [EMAIL PROTECTED]
Date: Fri, 20 Apr 2007 17:58:10 +0400
Hi,
in my dial plan I've configured two trunks to make outbound calls (one for
national calls and other international). I want to allow only 2-3 extension
to make use of my international trunk to make outbound calls so I want some
kind of auth. based on their callerid . Please guide.
Just detect that a call is international, then branch out. e.g., if 011 is
the prefix required for international,
[outbound]
exten = _011.,1,Dial(Local/${EXTEN}/international)
exten = _X.,1,Dial(ZAP/g1/${EXTEN})
[international]
exten = _X.,1,GotoIf(${DBEXISTS(international/${CALLERID(NUMBER)})}?:deny)
exten = _X.,n,Dial(ZAP/g1/${EXTEN})
exten = _X.,n,Hangup; just in case
exten = _X.,n(deny),Playback(not-a-valid-numbertry-again)
exten = _X.,n,DISA(nopassword,outbound)
This is assuming AstDB contains a family international that includes
extensions/ID's allowed. Hope this helps.
Yuan Liu
thanks
arun
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RE: [asterisk-users] Re: Trigger for unavailable SIP peer
From: C F [EMAIL PROTECTED] Date: Thu, 19 Apr 2007 09:35:13 -0400 Thank you all for your response, but it appears that some of you didn't understand my question. I know I can schedule a cron to check the status (I can even use asterisk -rx sip show peers | grep UNREACHABLE if I use a cron) but that is not what I want. I want either a way that just as asterisk prints to the CLI the following: Peer '120' is now UNREACHABLE! Last qualify: 118 it should also be able to trigger whatever action from a conf file or the like. I think you can start a dial plan loop from a call file upon asterisk start just for this purpose. Then you should be able to use dial plan logic to take action. Still not out-of-box, but adds a little more flexibility than cron (in the sense of less programming, not in ultimate control). Yuan Liu Or if there is an available solution even that involves a cron job but already has all the options, so I don't have to reinvent the wheel. On 4/18/07, C F [EMAIL PROTECTED] wrote: I use qualify in sip.conf and need to setup a trigger when asterisk sees it as unreachable, so that I can either drop a call file, or send an email, or both. How can I do that? 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] sending an SMS via Asterisk?
From: Per Jessen [EMAIL PROTECTED] Date: Wed, 18 Apr 2007 14:48:45 +0200 Per Jessen wrote: Per Jessen wrote: OK, part of the confusion is now clearing up. But I'm not getting much further. When I try to send an SMS, I see the call going through, but no SMS is ever sent. This is a bit of what I see in the debug output: (this is sending a longer message, protocol 2): P[ 2] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 2] -- None -- mISDN/3-u54 answered Local/[EMAIL PROTECTED],2 Channel Local/[EMAIL PROTECTED],1 was answered. Launching SMS(062210|t) on Local/[EMAIL PROTECTED],1 P[ 2] * IND: Got Fixup State:CONNECTED L3id:50012 == Spawn extension (Internal, 062210, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' P[ 2] I IND :FACILITY oad:0434439000 dad:062210 pid:19 state:CONNECTED P[ 2] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:0062210 P[ 2] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 2] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 2] -- AOCD currency: currency:FR. amount:10 multiplier:1 typeOfChargingInfo:-1220842403 P[ 2] I IND :INFORMATION oad:0434439000 dad:062210 pid:19 state:CONNECTED P[ 2] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:0062210 P[ 2] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 2] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 2] -- None -- SMS[-1] RX 93 00 6D -- SMS[0] TX 10 98 96 00 10 01 00 00 11 06 00 00 00 00 00 00 00 12 03 00 02 00 04 13 65 00 53 65 63 75 72 69 74 79 20 72 65 73 65 61 72 63 68 65 72 73 20 68 61 76 65 20 74 72 61 63 65 64 20 73 70 61 6D 2D 73 65 6E 64 69 6E 67 20 62 6F 74 6E 65 74 20 63 6C 69 65 6E 74 73 20 62 61 63 6B 20 74 6F 20 6E 65 74 77 6F 72 6B 73 20 72 75 6E 20 62 79 20 74 68 65 20 55 53 20 6D 69 6C 69 74 61 72 79 2E 17 01 00 01 18 0A 00 30 34 33 34 34 33 39 30 30 30 1B 01 00 01 1C 03 00 00 00 00 E8 P[ 2] I IND :DISCONNECT oad:0434439000 dad:062210 pid:19 state:CONNECTED P[ 2] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:0062210 P[ 2] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 2] -- caps:Speech pi:8 keypad: sending_complete:0 P[ 2] -- org:1 nt:0, inbandavail:1 state:10 P[ 2] -- queue_hangup In all the other examples I've come across on the 'net, there are multil lines beginning SMS[x] RX/TX .. The operator seems to hang up on you. Good thing is, the operator is at least responding to your call and sending you that initial answer. This may sound bizarre but try the s option and operate in mttx mode. I vaguely remember seeing a comment about one operator does some role reversal. (May not be due to protocol 2.) If you have an extra channel to spare with (seems you do), can also try to set up a context to receive SMS so you know all your commands/dial plan are working before testing against operator. (I always test via SIP channel to simplify my debugging. You can do so, too.) Yuan Liu /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] incoming SIP call
From: Jean Marc Le Fevre [EMAIL PROTECTED] Date: Wed, 18 Apr 2007 18:14:41 +0200 Hello all, I'm having a quite simple configuration like: SIP provider = asterisk SIP = lan Everythings works fine but sometime I can't get incoming call. Define sometimes and from where the income call you can't get? here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance [good stuff sniffed] Where do you suspect the error message is? --- Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Does this message make sense, not registered? Yuan Liu Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI -- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: asterisk sip:[EMAIL PROTECTED];tag=as372da2cb To: sip:freephonie.net;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all allow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register = 09:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=6 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXX username=09XXX dtmfmode=inband qualify=6 fromdomain=freephonie.net [freephonie_inbound] type=peer context=incoming host=freephonie.net qualify=6 allow=all deny=0.0.0.0/0.0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten = s,1,Ringing exten = s,2,Noop(I receive a sip call); exten = s,n,Goto(home,1000,1) exten = s,n,Congestion ; ... !DSPAM:462643f450705772331342! ___ --Bandwidth 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] Trigger for unavailable SIP peer
From: dave cantera [EMAIL PROTECTED] Date: Thu, 19 Apr 2007 00:11:09 -0400 cf, I haven' t used the * manager... but from research, that is how I would expect to do it... I would have a cron job fire off every 5 minutes (or so, probably configurable) and connect to * via the manager, request the status, then send an email based on the result... would be pretty easy... could look into it if you would like... email me off list... daveC If you are inclined to use cron, manager interface would be an overkill. You can easily query either AstDB (database show, database showkey) or sip show user via asterisk -rx and parse the result. Yuan Liu C F wrote: I use qualify in sip.conf and need to setup a trigger when asterisk sees it as unreachable, so that I can either drop a call file, or send an email, or both. How can I do that? Thank you -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth 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] Timestamp in recorded calls filename
From: Ricardo Melendez [EMAIL PROTECTED]
Date: Wed, 18 Apr 2007 17:39:40 -0500
Hi, I need to add the timestamp to the recorded call filename, I use this
variable ${TIMESTAMP} in the Monitor() function, but when I look for this
call, the TIMESTAMP is missing in the filename.
Maybe you can show us how you used ${TIMESTAMP} in Monitor()?
Yuan Liu
I try to export this as a environment variable but nothing changes.
Any help is welcome, thanks.
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RE: [asterisk-users] sending an SMS via Asterisk?
From: Per Jessen [EMAIL PROTECTED]
Date: Tue, 17 Apr 2007 09:40:18 +0200
I've been googling and reading a lot, but I'm not getting any closer to
getting an SMS sent via Asterisk.
Prior to switching to asterisk, I used sms_client on an ISDN line to
dial one of two Swisscom SMS centers: 0900900941 or 0794998990.
My dialplan looks like this:
exten = 0900900941,1,Goto(smsmotx,${CALLERIDNUM},1)
exten = 0794998990,1,Goto(smsmotx,${CALLERIDNUM},1)
How do callers get into these extensions?
; outgoing SMS
[smsmotx]
exten = _X.,1,Set(smsFrom=${CALLERIDNUM})
exten = _X.,n,SMS(${smsFrom},,${EXTEN},${CALLERIDNAME}) ; Create an SMS
exten = _X.,n,SMS(${smsFrom}) ; Send queued SMS
exten = _X.,n,Hangup()
When I attempt to send an SMS using smsq, Asterisk appears to be
behaving normally, a call is made etc., but the SMS never arrives ...
I'm a bit confused about your procedures. On one hand, if you use smsq, you
don't need to use SMS application (unless you are in the receiving end,
where you don't use smsq, either). You'll need to show the actual smsq
command line.
On the other hand, based on the dial plan snippet you provided, you are
dialing whoever dialed you (CALLERIDNUM), not Swisscom SMS centers. Not
sure why your caller never complained about spam calls, if Asterisk indeed
made the calls.
A quick fix would be (untested)
exten = 0900900941,1,Goto(smsmotx,${EXTEN},1)
exten = 0794998990,1,Goto(smsmotx,${EXTEN},1)
Hope this helps.
Yuan Liu
What am I doing wrong? Let me know what diagnostics I need to provide
if anyone wants to take a closer look.
thanks
/Per Jessen, Zürich
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RE: [asterisk-users] Having trouble figuring this out...
From: Brian William Kaplan [EMAIL PROTECTED] Date: Mon, 16 Apr 2007 23:29:14 -0400 Hello, I am using FastAGI and the AMI with Asterisk 1.2. Using the AMI I originate a call to a person. I would like the person to be able to press * to hangup or *9 to perform a certain function. I was thinking I could make it hangup on * and then WAIT FOR DIGIT to see if 9 is pressed. I'm not able to get this to work. Does anyone have any ideas on how I can make this happen? Basically, I want the callee to be able to press *9 to block a caller from calling the toll free line. I think the logic is botched in a normal bridged call. You cannot hang up a call and expect one party to continue interact with the sytem, because Asterisk is not a party. But I imagine it might be possible if you bring both parties into a conference, then use * to disconnect the one you want to rid, and let the other party press more buttons. Hope this helps. Yuan Liu This might be double posted because I'm not sure if my first posting went through. Sorry. ___ --Bandwidth 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 Asterisk + Hylafax
From: Tzafrir Cohen [EMAIL PROTECTED] Date: Mon, 16 Apr 2007 08:45:38 +0300 On Sun, Apr 15, 2007 at 10:10:34PM -0700, Yuan LIU wrote: (But if Zaptel and Hylafax can share an X100P driver ...) Where can you find a modem driver for a X100P? Kinda my question, too. Motorola used to have an SM56 Linux driver, but removed from their site. Now, there are some references to this, such as http://www.motorola.com/softmodem/public_download/Linux/ReadMe_Legacy_SM56.txt and http://www.angelfire.com/linux/sm56/, but if the original driver is nowhere to be downloaded, there might be a chance you can hack the URL based on the Motorola document. No knowledge about X100P/Intel and other. Yuan Liu I recently asked about it in the linmodemds.org mailing list, and aparantly none is available. -- 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] Passing a variable from one Asterisk boxtoanother
From: Jesus Mogollon [EMAIL PROTECTED]
Date: Mon, 16 Apr 2007 13:33:16 -0400
Hi Craig
I've been developing a Recording Server app (which I will be giving back
to the community) and one of the requirements is for the recording to be
offloaded to several machines. Because of the filename is being set prior
to
the recording, I need to pass this variable to the slave server. I'm using
1.2.13 (heavily patched) and I came across your email. Any chance of
getting
your port? Thanks for your help...
If there are only a limited number of variables to pass, you may as well do
this in dial plan using SIPHEADER.
Yuan Liu
Jesus Mogollon
On 2/22/07, Craig Guy [EMAIL PROTECTED] wrote:
Hi Richard,
there was a thread regarding this a while ago on the dev list which
resulted
in a patch being made to allow variable passing via IAX2 channels. See
http://bugs.digium.com/view.php?id=7619 for the patch which I think is in
SVN or anyhow, is not in 1.2
I have recently backported this patch to 1.2 and have a patch which is
tested against 1.2.12, 1.2.12.1 and 1.2.15, but should work against at
least
1.2.13 and 1.2.14. The patch introduces a new dialplan function called
IAXVAR, Email me if interested.
Craig
- Original Message -
From: Richard Lyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 21, 2007 7:27 AM
Subject: Re: [asterisk-users] Passing a variable from one Asterisk box
toanother
Richard Lyman wrote:
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All
works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get
passed...
Does anyone have any clever ideas?
as noted in asterisk/docs/README.variables (iirc)
you should see that variable inheritance can occur by prefacing the
variable with '_' or '__'
also, depending on the age of your asterisk you might want to start
using
'Set' vice 'SetVar'
also, having ${EXTEN:0} , the :0 doesn't do anything, so you should
not
use it and just have ${EXTEN}
i hope this helps
sadly replying to my own post, but, i forgot to mention that
passing variables with IAX2 can be an issue sometimes when you use
user and peer (the user side can pass vars the peer side can not, or
doesn't accept them iirc)
this does not happen using friend, but that has its own issues... check
the wiki for more thoughts about this.
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RE: [asterisk-users] Fax with Asterisk + Hylafax
From: Jose Limeres [EMAIL PROTECTED] Date: Sun, 15 Apr 2007 19:10:58 +0100 Hi, Anybody lucky with this config inside an Asterisk server for dealing with FAX ? FXO_LINE ASTERISK 1.4.2 --- IAXMODEM -- HYLAFAX TDM400PZAPTEL 4.3.1 1 FXO port 1.41 Search Asterisk forum. Yes, somebody posted positive results. Yuan Liu I know Fax is not officially supported on TDM400P cards but I did not expect not being able of sending one single Fax. Actually when I try to send a Fax, the call is established between my * server and the remote Fax but after 30 secs Asterisk disconnects the call and Hylafax reports NO CARRIER DETECTED. Tried playing around with a few parameters such as no echocancellation, alaw (also slinear) codec, faxdetection =incoming in zaptel but with no luck. Regards, Jose Limeres ___ --Bandwidth 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 Asterisk + Hylafax
From: Steve Totaro [EMAIL PROTECTED] Date: Sun, 15 Apr 2007 22:36:15 -0400 Stephen Bosch wrote: Steve Totaro wrote: You could try to get it working but it may never be 100%. If your needs are 100% then I suggest using a standard fax and get an analog line and do it the old fashioned way. If you need Hylafax type features then buy a modem that is compatible with Hylafax and run it on a different box. It's not entirely clear to me why people continue to cling to the idea that Asterisk should handle faxing also. What's the benefit? Hylafax is great, and you can even use it on the same machine. On same machine is a bit exaggerated, considering there is a Zaptel card on it. (But if Zaptel and Hylafax can share an X100P driver ...) Yuan Liu -Stephen- I could have sworn that is what I just said. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring audio file legth
From: Suity Zsolt [EMAIL PROTECTED] Date: Fri, 13 Apr 2007 08:43:33 +0200 Stephen Bosch wrote: Bob Smither wrote: On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote: Hi, I have to set call length to 3min, but before hangup have to warn caller. There are many IVRmenu and submenu options with different warning audio. I have to measure somehow the audio file length and subtract it from 3 minutes. I have not tried this, so I may be off - but do you really have to do this? The documentation I have indicates that if there is an extension T in the context, that extension is used at the absolute timeout. So, would: exten = T,1,play your warning message exten = T,n,Hangup What if he wants to warn the caller with 30 seconds remaining? Then 15? Then 5? It's not my goal this time, but good question. When a global timeout is reached and jumps to the T extension, can I change the timeout (Set(TIMEOUT(absolute) again) and go back somehow? When we dialing this isn't a problem use L option. Two questions. First, what is the application you are trying to limit duration if not Dial()? IVR? Second, I couldn't seem to get L to work with y and z. Dial() seems to simply ignore these. Anyone experiencing similar? Yuan Liu # L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c) * LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the caller. * LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee. * LIMIT_TIMEOUT_FILE - File to play when time is up. * LIMIT_CONNECT_FILE - File to play when call begins. * LIMIT_WARNING_FILE - File to play as warning if 'y' is defined. If LIMIT_WARNING_FILE is not defined, then the default behaviour is to announce (You have [XX minutes] YY seconds). -- Suich ___ --Bandwidth 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: number to names
From: Ronaldo Zacarias Afonso [EMAIL PROTECTED] Date: Fri, 13 Apr 2007 08:06:04 -0300 OK Yuan, What I wanted to know is if the extension I've created is right. exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) OK, the syntax is a bit off. exten = 101,1,Dial(SIP/[EMAIL PROTECTED]) will send the call to [EMAIL PROTECTED] (at least that's what I'm using); whether that user (more precisely, the server that hosts this user) accepts the call is up to the server. Yuan Liu Will my asterisk bridge a SIP phone that dialed 101 to the SIP user: [EMAIL PROTECTED] Do I need some think more in order for it to work? Do you have or know any documentation that explains me that? Regards Ronaldo. On 4/13/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Ronaldo Zacarias Afonso [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 11:54:51 -0300 Hi all, Is it possible to configure an extension number to dial a sip address? Nothing prevents you from doing this. Yuan Liu For example: exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) That way I can dial to a sip name using my Hardphone that is not able to dial using names just numbers. Thanks in advance. Ronaldo. (I hope putting my sip address soon here) ___ --Bandwidth 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] SIP: number to names
From: Ronaldo Zacarias Afonso [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 11:54:51 -0300 Hi all, Is it possible to configure an extension number to dial a sip address? Nothing prevents you from doing this. Yuan Liu For example: exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) That way I can dial to a sip name using my Hardphone that is not able to dial using names just numbers. Thanks in advance. Ronaldo. (I hope putting my sip address soon here) ___ --Bandwidth 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] Catch all undefined numbers to play a nice messageand resta
From: pedro noticioso [EMAIL PROTECTED]
Date: Thu, 12 Apr 2007 12:02:52 -0700 (PDT)
Hi there list!
I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like
May be you can explain what is the difference between an undefined extention
(number) and an extention (number) that doesn't exist. You can certainly
use extension i to catch wrong numbers and play nice messages instead of
busy.
If you only want to catch numbers that matches a certain pattern but don't
exist in your system, and want to give busy signal to all other dialed
numbers, you can match the pattern and transfer to another context, then use
i in that context. For example, suppose your extensions should start with
2,3,4 and must be 3 digits, but you have only defined 200-242, 320-350, and
400-420, you can do (untested)
[incoming]
exten = _[2-4]XX,1,Goto(valid,${EXTEN},1)
exten = i,1,Congestion; give busy to any other dialed number
[valid]
exten = _2[0-4]X,1,Dial(SIP/${EXTEN})
exten = _24[12],1,Dial(SIP/${EXTEN})
exten = _3[2-4]X,1,Dial(SIP/${EXTEN})
exten = _4[0-1]X,1,Dial(SIP/${EXTEN})
exten = 420,1,Dial(SIP/${EXTEN})
exten = 350,1,Dial(SIP/${EXTEN})
exten = i,1,Answer(); if
exten = i,n,Playback(nice-message)
exten = i,n,DISA(nopassword,incoming)
Hope this helps.
Yuan Liu
You can search for the word irc to see my comments,
the line above is my latest unsuccessful test, thanks!
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RE: [asterisk-users] DTMF problem with inbound calls on Toll-Free number
From: ismir saljic [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 07:42:13 -0700 (PDT) Hi all, I have asterisk 1.2.13 and problem is about DTMF.When i have incoming call on Toll-Free number asterisk accept DTMF digits but dial only first in context. Per instance: When i press 1 it is OK,but when i try to dial extension 700 asterisk dial only first digit(1) and i receive from asterisk invalid extension 7 in context...Extensions 700 exists.It seems asterisk dial only first digit. Several possibilities come to mind. But you haven't indicated what kind of incoming trunk you use for the toll-free and toll numbers? Are you receiving the call from VoIP? Yuan Liu When i dial ordinary(not Toll-Free)number everyting is OK. Please help. Regards! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missing chan_zap.so
From: Tzafrir Cohen [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 09:18:46 +0300 On Wed, Apr 11, 2007 at 08:09:16PM -0700, Yuan LIU wrote: From: Sanjay Rajdev [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 01:29:51 +0530 (IST) [good stuff sniffed] and downloaded zaptel 1.4.1, after that executed the following commands ./configure make clean make make install Went to asterisk folder ./configure make clean make make upgrade But could not get chan_zap.so then did the make install of asterisk. still missing the chan_zap.so Have you loaded wctdm? Whatever kernel modules are loaded does not matter to the build of chan_zap.so Tzafrir, In my experience, the background menuselect (1.4) seems to decide that chan_zap.so is unnecessary if a Zap drive is not loaded. I also manually ran menuselect, and found chan_zap selection greyed out - unselectable. Loading a Zap module seems to solve the problem. Yuan Liu Do you have: Should be generated by 'make': channels/chan_zap.so # under the asterisk build directory Should be copied by 'make install': /usr/lib/modules/chan_zap.so -- 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] Automatic Hang
From: LKS GMAIL [EMAIL PROTECTED]
Date: Thu, 12 Apr 2007 13:02:24 +0200
Hi guys!
Im using Asterisk 1.2 with mISDN support.
I have problems with Pickup calls with my Grandstream Buttons . I set up on
Dial Plan this:
Exten = _**XXX,1,Pickup(SIP/{EXTEN:2}) but it doesnt work if the call
comes from mISDN. So, I wanna do something to this:
Exten = _**XXX,1,SendDtmf(*8#) because if I introduce *8# into my
telephone
i can pickup a call from everywhere. BUT the problem is that I cannot dial
automatically *8#. Does anybody know how to do it?
It is not clear what do you mean by introduce *8# into. Are you referring
to the pickupexten feature (default set to *8)? What if you change
pickupexten = *8 to pickupexten = ** in features.conf?
Yuan Liu
THANKS
Saludos, Lukassky.
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RE: [asterisk-users] missing chan_zap.so
From: Sanjay Rajdev [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 01:29:51 +0530 (IST) [good stuff sniffed] and downloaded zaptel 1.4.1, after that executed the following commands ./configure make clean make make install Went to asterisk folder ./configure make clean make make upgrade But could not get chan_zap.so then did the make install of asterisk. still missing the chan_zap.so Have you loaded wctdm? Just make install zaptel doesn't load it. 'modprobe wctdm'. (You may even want to ztconfig at this time.) Then remake Asterisk. You may need to make menuselect and select chan_zap first as a selection may have been made for you when zaptel wasn't loaded. Yuan Liu Can someone please help. Regards, Sanjay Rajdev ___ --Bandwidth 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 is your Backup Strategy?
From: Alex Balashov [EMAIL PROTECTED] Date: Wed, 11 Apr 2007 20:17:37 -0400 (EDT) On Wed, 11 Apr 2007, Forrest Beck said something to this effect: 1) Using hearbeat and drbd to monitor the servers. When the primary fails the backup will assign itself the virtual ip used between the two, and then mount the drbd disk which has the asterisk configs and voicemail. The biggest con to this is hearbeat just monitors a ping response either over IP or a COM port. So if the asterisk service dies, heartbeat will not fail over. Although I think there are work arounds for this. The newest version is suppose to have support for monitoring a TCP port as well This seems like a good approach, if you've got any stability and/or filesystem-related quirks ironed out -- I've heard of some. I don't know much about heartbeat, but I don't imagine it'd be hard to hack in a SIP polling event either internally or externally. You are right. It shouldn't be hard to just require the primary server to register with the backup, monitor this registration from backup; when Asterisk on primary fails, run a script to request primary to shutdown and take over. Yuan Liu You could use SIP Swiss Army Knife (sipsak) or some other SIP testing tool to send a periodic OPTIONS ping to the SIP service and trigger a protection switch to the secondary server if it's down. Even if you can't hack this into the heartbeat setup itself (can't it use external scripts for monitoring?), you can certainly do something like run it on the primary server and if the SIP service dies, enact a firewall rule that drops ICMP responses and thus artificially trigger a failure. -- Alex -- Alex Balashov [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 ___ --Bandwidth 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] Verizon-Vonage Lawsuit
From: Kenneth Padgett [EMAIL PROTECTED] Date: Mon, 9 Apr 2007 23:49:31 -0400 [good stuff sniffed] I'm not doubting that patents exist, I'm just betting that you'd have to have some seriously drunken vision to interpret them as the exact business processes Vonage uses. I think if Verizon thought for a second they had solid ground to stand on, they would disclose which patents they're referencing so the public could decide. I bet you can access court records under some public information access laws. 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] Adding Noise or background noise
From: Arun Kumar [EMAIL PROTECTED] Date: Sun, 8 Apr 2007 05:25:58 -0700 Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding some noise /echo /latency or something. Please guide me. This is got to be the strangest requirement I've seen - a penalty box. But if you must, one way to add noise could be to bring the parties to a conference, then add a third party to the conf. Another possibility is to use frequent announcements (don't have to be real announcements, but could be simple, brief noise) with L option in Dial(). I haven't seen L announcements working properly, though. Yuan Liu thanks arun ___ --Bandwidth 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] Audio Gain Settings
From: Bob Smither [EMAIL PROTECTED] Date: Fri, 06 Apr 2007 20:22:34 -0500 Warning - novice question ahead! Dear List, I have installed Asterisk 1.4.2 on an AMD dual core x86-64 box running CentOS 4.4. Compilation and installation were straightforward. The box only supports IAX connections so I have no zap hardware. My question is this - where do I set the txgain and rxgain parameters for the IAX channels? With a previous setup I used settings in zapata.conf, but I believe these are not used with the IAX connections (?). You are right. zapata.conf is not used in IAX connections. My reading has led me to believe that manipulating gain on an IP PBX is neither necessary nor practical in VoIP channels, so Asterisk does not devise such settings. Yuan Liu Thanks for any insight. -- Bob Smither [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] hox to connecte two asterisk server
From: hind habaoui [EMAIL PROTECTED]
Date: Fri, 6 Apr 2007 18:01:11 +
hi lee.
I see your problem with trunk iax, probably i don't have the solution but i
don't knew if you can help me to solve mine.
Can't seem to see what the problem you have? Errors? Incorrect result?
(What is expected and what is the result?) Also, you need to clarify the
settings on two servers more clearly - your sip-calls context seems to
suggest that server B uses IAX with its users, but uses SIP to connect to
server A? The iax.conf seems to suggest SIP rather than IAX.
Yuan Liu
i want to connecte two asterisk server: server A and server B. i want make
possible calls betwen all asterisk users.: users in server A with sip
number
022100 can phone another sip user in server B with number 037100.
this is my config:
*
iax.conf for server A:
**
register = serveur_rabat:[EMAIL PROTECTED]
[serveur_casa]
type=peer
host=dynamic
username=serveur_casa
secret=casa
disallow=all
allow=ulaw
allow=gsm
;context=sip-calls
[serveur_casa]
type=user
host=dynamic
username=serveur_casa
secret=casa
disallow=all
allow=ulaw
allow=gsm
my extension.conf
**
[sip-calls]
exten=_022[1-8]XX,1,macro(Bienvenu)
exten=_022[1-8]XX,2,SetGlobalVar(BOITE=${CDR(src)})
exten=_022[1-8]XX,3,Dial(SIP/sip-${EXTEN},${TP_MAX_APPEL})
exten=_022[1-8]XX,4,macro(BoiteVocale,${BOITE})
exten=_022[1-8]XX,5,hangUp()
;
;
;lecture des boites vocales
exten=_[1-8]XX,1,macro(lecture_boite)
exten=_[1-8]XX,2,PlaBack(vm-num-i-have)
exten=_[1-8]XX,3,HangUp()
;
;
; on donne accès au service du standard
exten=022999,1,Wait(5)
exten=022999,2,Dial(${TEL1},,t)
exten=022999,3,HangUp()
include=parkedcalls
include=iax-calls
[iax-calls]
exten=_037XXX,1,macro(Bienvenu)
exten=_037XXX,2,Dial(IAX2/serveur_rabat/${EXTEN},${TP_MAX_APPEL},r)
exten=_037XXX,3,macro(BoiteVocale)
**
file config for the second server looks like the server's A file.
thank you in advance
hind
GTR 2007
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RE: [asterisk-users] disabling authentication
From: Mark Price [EMAIL PROTECTED] Date: Wed, 4 Apr 2007 10:07:31 -0400 Is there a way to cause asterisk to accept all calls without any authentication? Mark Yes - not to set up a user/peer section in sip.conf. The context in [general] section will be used. 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] ZAP device reference in Zaptel 1.4
From: Devraj Mukherjee [EMAIL PROTECTED] Date: Wed, 4 Apr 2007 11:46:11 +1000 Hi Eric, Thanks for your suggestion I just reinstalled Asterisk, it still doesn't seem to know anything about Zaptel. I am using CentOS and installed Asterisk using yum from ATrpms. Anything else I can try? Try lsmod to confirm that zaptel is indeed installed. I'm not familiar with CentOS or yum, but I assume you installed a binary package, so chan_zap.so is probably included. Hope this helps. Yuan Liu On 4/4/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Devraj Mukherjee wrote: Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing [EMAIL PROTECTED]:1] SetCDRUserField(SIP/103-b7802230, Telstra) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/103-b7802230, ZAP/4/69223139) in new stack [Apr 4 10:47:43] WARNING[5659]: channel.c:3024 ast_request: No channel type registered for 'ZAP' [Apr 4 10:47:43] WARNING[5659]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) You need to reinstall Asterisk. You installed Asterisk before installing Zaptel so Asterisk did not build anything that requires 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] Weird extension behavior
From: Mark Hennessy [EMAIL PROTECTED] Date: Sun, 01 Apr 2007 06:15:40 -0400 Hi, I'm using Asterisk with two Cisco 7960 phones using SIP. I'm seeing the following weird behavior: SIP Phome 1 is extension 4002 SIP Phone 2 is extension 4003 I call 4002 from 4003 and that works fine. I call 4003 from 4002, and it rings locally to 4002, never gets to 4003. I'm able to send a config query packet to 4003 from the asterisk console and get a response, when I send one to 4002 there is no respone. I know that both phones pull down their config via TFTP properly, I look in the network settings and see that 4002 has been given an IP of x.y.z.201 and 4003 has been given an IP of x.y.z.202 and the asterisk box is running on x.y.z.74. I combed through all of the config files in both Asterisk's config and the TFTP-downloaded configs for the phones looking for any possible instance of 4003 being transposed for 4002 or vice versa and was not able to find any. What additional information is necessary to provide to trace down and resolve this issue? Corresponding entries in sip.conf may help. Yuan Liu AFAICT, the server is using Asterisk 1.2.x and beyond the 7960 phones, no other specialized hardware is in use. -- Mark P. Hennessy ___ --Bandwidth 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: Paging
From: Forrest Beck [EMAIL PROTECTED]
Date: Fri, 30 Mar 2007 16:52:39 -0400
Forgot to mention.
We are using Polycom phones on asterisk 1.4.2
I tried the allpage agi, but it checks for all SIP peers connected to
the server.
On 3/30/07, Forrest Beck [EMAIL PROTECTED] wrote:
First off, A lot of thanks to this list. I have learned ton from
reading through the posts this past year.
I need some advise.
I have two group of phones connected to a single server.
Group1= SIP/2503SIP/2504
Group2=SIP/3501SIP/3502
I'd like to be able to dial an extension and page a certain group of
phones only if ChanIsAvail returns 1.
I am not sure how to go about programming this. I though to write a
AGI script that reads a list of phones (one list per group), checks
ChanIsAvail then Pages the phone. I will have about 60 extensions
per group to Page. Will there be lag until all the phones get paged
and the script finishes?
The lag shouldn't be too large. Yet you don't have to use AGI to build a
list, and you don't even have to wait for all channels to be checked if I
understand the objective correctly. For example (untested),
exten = _Z.,1,ChanIsAvail(SIP/${EXTEN},j)
exten = _Z.,n,Dial(SIP/${EXTEN})
exten =
_Z.,101,Set(group=$[$[${GROUP1}=~SIP/${EXTEN}]?${GROUP1}::${GROUP2}])
exten = _Z.,n,While(${group})
exten = _Z.,n(check),ChanIsAvail(${group},j)
exten = _Z.,n,Set(page=${${page}${AVAILORIGCHAN}}); tweak if empty not
acceptable
exten = _Z.,n,Dial(${page}); start dialing before list completes
exten = _Z.,n,Set(group=$[${group}=~${AVAILORIGCHAN}*(.*)])
exten = _Z.,n,Endwhile
exten = _Z.,check+101,Congestion; or however way you want to handle no
channel available
You may need to tweak a bit to get it working but that's the spirit. Hope
this helps.
Yuan Liu
Then I thought maybe a Macro in the dialplan to dial a global var of
the group of phones, but that won't work. If phone isn't available,
none will get paged.
Has anyone done this before? I just don't know where to start.
Thanks
--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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Re: [asterisk-users] Question on Priorities
From: Rizwan Hisham [EMAIL PROTECTED] Date: Sat, 31 Mar 2007 17:01:51 +0500 [inbound-sip] exten = uxbod,1,Dial(sip/1001,20,jt) exten = uxbod,n,Hangup exten = uxbod,102,PlayBack(uxbod) exten = uxbod,103,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,104,Hangup() here if dial fails then n+101 =102 extension will get executed unless you use j option in dial application and priority jumping has to be set to priorityjumping=yes in the general section of your extensions.conf file. In your dialplan i dont know y you r forcing the caller to goto voicemail even if the call has already answered. I hope you understand my modification in your dialplan. On 3/31/07, --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my extensions.conf I have a simple plan for testing :- There is a simpler way, by using label. [inbound-sip] exten = uxbod(ntest),1,Dial(sip/1001,20,t) exten = uxbod,n,PlayBack(uxbod) exten = uxbod,n,Hangup() exten = uxbod,ntest+101,PlayBack(uxbod) exten = uxbod,n,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,n,Hangup() Yuan Liu [inbound-sip] exten = uxbod,1,Dial(sip/1001,20,t) exten = uxbod,n,PlayBack(uxbod) exten = uxbod,n,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,n,Hangup() exten = uxbod,103,PlayBack(uxbod) exten = uxbod,104,VoiceMail([EMAIL PROTECTED],s) exten = uxbod,105,Hangup() So when the extension has to add 101 do I just do n+101 ? TIA -- 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 -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth 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] bugetone 200's
From: [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 18:29:07 +0100 how do these phones perform? ok for office use? work well with asterisk? any info would be appreciated. I have a bugetone 100 at home. It appears to work with Asterisk and basic quality seems to be OK (G.711) including speaker phone. But I use none of its built-in features such as transfer or even hold so the elaborative right-hand side buttons are pretty much useless. One annoying problem about buttons: it doesn't have a Redial key, but has a Send key to please some annoying VoIP system. And if you need to redial, press that Send key. The designer must be out of his or her mind. A lesser button issue: the mute key is marked as Mute/Del. As such, could easily be overlooked by a casual user looking for Mute key. I would much appreciate a stand-alone Mute key, and won't mind having a combined Flash/Del key. You guessed it: Flash stands alone. (I do appreciate the corner location of the Mute/Del key. But Del is really not that useful to qualify for this premier location.) 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] Setting rxgain per channel
From: Delca [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 18:39:37 -0300 How do I set rxgain per channel on zapata.conf? I've a TDM400 with 2 FXS. Does FXS even use rxgain? To set rxgain for an FXO channel, simply put the entry before saying channel =. Hope this helps. Yuan Liu Thank you! Santiago del Castillo ___ --Bandwidth 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] maximum simultaneous calls
From: Mark Quitoriano [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 23:05:57 +0800 Hi, what could be the maximum simultaneous calls can asterisk do? i read about the asterisk business edition review[1] and it can only handle 120 simultaneous calls? i'm using 1.2.x branch of asterisk and i use more or less 90 simultaneous calls. [1] http://www.voiptalk.org/products/Asterisk+Business+Edition What about http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning? People reports all kinds of numbers above 120. The answer partially depends on your hardware. Simultaneous calls can also mean very different things under different circumstances, as the page will tell you. If there is no transcoding/NAT'ing/in-band signaling, simultaneous calls can mean SIP set-ups only. You can see extremely high numbers even on ancient equipment. If everything is in-band and you are using CPU-intensive CODECs, the number will drop sharply. It also varies with types of channels, i.e., whether you use PSTN, IAX, SIP, H.323. But still, I don't think 120 is any limit. 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] Unsetting Global Vars
From: Johann Hoehn [EMAIL PROTECTED] Date: Wed, 28 Mar 2007 16:45:28 -0500 How do I clear a global variable for good? I have a situation of needing to use global variables to aide in channel communication, but will be changing the name within a defined scope. Not sure if I understand what clear a variable mean. I don't think there is a concept like unset in Asterisk. If you want to make sure a used variable does not cause side effects, simply set it to null string. Additional Background... I want to get a variable from a channel (child) that is created by another channel (parent), however the execution of the parent channel does not continue until the child channel is gone. So I want to use a global variable as 'scratch' space and later the parent to grab it. Basically I need to be able to do the opposite of variable inheritance. I need to propagate a variable status up the channel chain instead of down. I feel the need to propagate a variable up the chain from time to time. But I still don't understand why this is necessary in your case, much less how this relates to the need to unset. Maybe you can give more specifics, even pseudo code. Yuan Liu -- Johann Hoehn Project Coordinator, Administration Direct: 270-707-2040 x 4011 Ecommerce Corporation (www.ecommerce.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] web based sip phone
From: Pezhman Lali [EMAIL PROTECTED] Date: Fri, 30 Mar 2007 02:05:35 -0700 (PDT) hello is any web based sip phone? The easy answer is yes. Search for Java SIP phone. Some of them can be deployed on the Web. Yuan Liu for example: a user after logining in, view a configured sip phone, and .. best MAni ___ --Bandwidth 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] just on my LAN
From: Josu Lazkano Lete [EMAIL PROTECTED] Date: Wed, 28 Mar 2007 09:59:16 +0200 hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 Depends on what you use it for. You certainly don't need Libpri. You may need Zaptel (specifically the ztdummy driver) if you want to run meetme, i.e., conference (some document also mention music on hold). Not sure what sounds and addons are for. (Basic sound files are included with distribution at least up to 1.2.16.) Yuan Liu 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] Can I generate random SIP traffic?
From: [EMAIL PROTECTED] [EMAIL PROTECTED] Date: Wed, 28 Mar 2007 09:53:14 +0100 Hello, I would like to generate a peer-to-peer or a server/client SIP traffic between two or more Openwrt access point, to make some statistics about QoS. I tried some SIP traffic generators for OpenWrt, but I didn't find nothing of satisfactory. Now I wonder if asterisk can help me generating random SIP traffic. I'm googling since yesterday without results. Can you help me plz? Thanks and sorry for the disturb. Since no one seems to have specific information, let me try generic. You can certainly program Asterisk to generate random SIP traffic. Or maybe you really mean SIP+RTP traffic. Either way, Asterisk can do it, just like you can program C or Perl to do so. The real question is: what is unsatisfactory about SIP traffic generators you have tried that you hope Asterisk to help? 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: Fwd: [asterisk-users] Multi-registration ?
From: Drew Gibson [EMAIL PROTECTED] Date: Mon, 26 Mar 2007 15:08:51 -0400 Olivier wrote: I tried using multiple accounts from one phone to separate call centre traffic but the phones (Aastra 480i) would default all calls from the phone to the account with the highest line number. This made it impractical for my purposes. Drew Do think this limitation comes the phone or from Asterisk ? Cheers The phone, it selects the outgoing account to use. Logistically, there is no way for the phone to know which outgoing account YOU want it to use, unless you press extra buttons like on old style PBX phones or multi-line phones. Short of having custom made phones, you can play with dial plan and use, for example, a special prefix or postfix to indicate which personality you want to present when outgoing. Is this practical? Yuan Liu regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.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] how to define a pilot number
From: Lito Lampitoc [EMAIL PROTECTED] Date: Tue, 27 Mar 2007 14:28:25 +0800 Hello all, is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Thanks. Lito Telco calls this line rollover. No it cannot be done with Asterisk or any PBX. It can only be configured on the telco side. 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] web based sip phone
From: Pezhman Lali [EMAIL PROTECTED] Date: Fri, 30 Mar 2007 02:34:24 -0700 (PDT) thanks Yuan I was search the best result is sipfoundary.org but it's client is not spesific for my purpose, but it will be. is any better answer for this searching? Have you tried JAIN SIP applet? It requires an application server to deploy (JBOSS does fine). But if you are desperate :-) (Well it didn't fit my need then but my requirements were rather bizarre.) Part of the answer also depends on your requirements. For some, a CGI/AGI Web interface constitutes a Web based phone. (Think Jahjah.) Such does not require any remote deployment and can be made very sophisticated. (You can even write a streaming Applet without running anything SIP on client machine, and let server do the SIP work.) On the other hand, with appropriate Active-X permissions, you can also deploy nearly any thick application. Yuan Liu best Mani --- Yuan LIU [EMAIL PROTECTED] wrote: From: Pezhman Lali [EMAIL PROTECTED] Date: Fri, 30 Mar 2007 02:05:35 -0700 (PDT) hello is any web based sip phone? The easy answer is yes. Search for Java SIP phone. Some of them can be deployed on the Web. Yuan Liu for example: a user after logining in, view a configured sip phone, and .. best MAni ___ --Bandwidth 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] Refresher course needed!
From: Brad Sumrall [EMAIL PROTECTED]
Date: Tue, 27 Mar 2007 00:06:13 -0500
Hello everyone
My name is Brad, I am an old Asterisk Vet of the very early days just
coming
back to join the group.
Ok, for starters, I feel like the monkey with the light bulb looking at
extensions.conf and sip.conf.
It has been some time.
A friend ask me to set up a asterisk server that records phone calls.
FC4
Asterisk 1.4
And all the latest and greatest
Problem number 1
Some good get back into the grove literature.
I work CLI only, never much for graphics and gui's
Asterisk 1.4 still has CLI. I don't think many people here use GUI.
voip-info.org is a good starter. Another really good restarter? CLI help!
Problem number 2
We have asterisk logged into teliax but cannot see the inbound call come up
on the CLI
Tethereal says this;
1660 3.829799 207.174.202.4 - 66.109.17.92 SIP Status: 100 Trying(1
bindings)
1661 3.831357 207.174.202.4 - 66.109.17.92 SIP Status: 200 OK(1
bindings)
Asterisk says this;
*CLI
Nothing, notta!
How did you start Asterisk or remote console? Have you tried core set
verbose 10? (Just kidding. Most often I go 3.) Have you tried
sip set debug?
My extensions.conf
(yes, I loaded the samples)
[general]
static=yes
writeprotect=no
clearglobalvars=no
;#include filename.conf
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;From here is brads stuff
exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten = YOURNUMBER,1,Answer()
exten = YOURNUMBER,1,DIAL(SIP/user,20)
Getting more confused about what inbound call you did not see after reading
the sample conf. Did you put a context title before brads stuff? What is
your sip.conf/user.conf if you expect incoming call from SIP?
Ah. Feels good to teach grandma cook milk:-)
Yuan Liu
Thanks to all!
Brad
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RE: [asterisk-users] outbound call
From: Karthik Arumugam [EMAIL PROTECTED] Date: Mon, 26 Mar 2007 21:35:45 +0530 HI All, I am new to asterisk. i want to make outbound calls from asterisk. I tried with many times with the given settings but in vain In vain says vain. Exactly what does not work? Any messages? Errors? This is my scenario: I have a *user A* who has registered with sip server(ONDO), I Is *user A*'s user name with the server 'test' as your dial plan suggested? made asterisk to register as a sip client with ONDO, I want to make a call to user A from an extension. What is an extension's context? Is this extension dexter as your config suggested? You can get much better response if you can help others understand what your problem is. Yuan Liu My configurations sip.config [general] context=default register = raja:[EMAIL PROTECTED]/1234 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to ( 0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls [authentication] auth=raja:[EMAIL PROTECTED] [*192.xxx.xxx.xxx*-out] type=peer ; we only want to call out, not be called secret=adsi6677 username=raja ; Authentication user for outbound proxies fromuser=raja ; Many SIP providers require this! fromdomain=*192.xxx.xxx.xxx* host=*192.xxx.xxx.xxx* - Ignored: context=outgoing [dexter] type=friend username=dexter secret=password host=dynamic context=outgoing extensions.conf [outgoing] exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]) Here *192.xxx.xxx.xxx* is my sip server host ip (ONDO). Please correct me where i am going wrong in this scenario. I was able to receive incoming calls to dexter from user A, Thanks in advance! Regards karthik ___ --Bandwidth 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] cutting hash in dial app
From: René Enskat [EMAIL PROTECTED] Date: Mon, 26 Mar 2007 20:29:16 +0200 hello, isit possible to cut off the hash behind a dial string? coz we have a provider who gives us an error 600 Declined if ther is a hash in dial command. for example: Dial(SIP/x.x.x.x-b7d2d870, SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED] x) and i have to cut out: -b7d2d870 regards rene Cutting out part of a string is very easy to do - CLI show function CUT. But the dial command you cited looks really strange. Don't look like correct syntax at all. So maybe you need to fix that first. CLI show application dial 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] Need help to strip variable
From: van Leen, Phil [EMAIL PROTECTED] Date: Fri, 30 Mar 2007 11:52:34 +0800 Hi all, I have a need to strip some characters from a variable to get the right data but have only found how to strip all but the last or middle stuff, need to keep the beginning. EG: With $(SIPURI) I want to keep just the sip number and delete the remainder '@server.com'. Ideally I'd like to use 'SayDigits($([EMAIL PROTECTED])' You can certainy use CUT(), or regular expression. Yuan Liu All replies greatfully accepted. Phil ___ --Bandwidth 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] Multi-registration ?
From: dave cantera [EMAIL PROTECTED] Date: Mon, 26 Mar 2007 11:44:41 -0500 olivier, soft phones on a PC require a port to connect to the server... haven't tried multiple soft phones, simultaneously, connecting to one server or multiple servers but if you can configure the outgoing port, it should be possible... NAT might get quite confusing so I would try it before making any commitments.. as for hard phones, I register my sip phones with two or three servers, one server per extension or two+ extensions with one server... one or more extensions to one server... if you have 3 extensions you can register those to 1-3 servers as you wish... daveC Olivier wrote: Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? If this is for testing purposes, yes. You can, for example, run multiple VM's to run several soft phones. Some soft phones does not check running status, so it is possible to run several on different virtual IP interfaces (assuming you already know how to set up virtual interfaces), in addition to running them on different ports as daveC mentioned. Of course it's possible to run different soft phones on different virtual interfaces. 2. Is possible to do the same with SIP hardphones ? This is a totally different question, and daveC gave the answer. If the only purpose is to have different personalities, some soft phones can have multiple personalities as well so you don't have to run multiple soft phones. 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] Interconnexion d'un serveur Asterisk � des PABX LG ( IP LDK
From: khawla khawla [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 13:20:57 + bounjour je dispose de differents commutateurs de LG (IP LDK) sur differents sites. je voudrais savoir comment je pourrais interconnecter ces differents IP LDK a un serveur Asterisk via IP ( ceci sous entend que chacun de ces commutateurs dispose déjà d'une carte VOIBE). Mecri d'avance pour l'aide Pas toujours facile d'obtenir de réponse d'une liste anglais, n'est-ce pas:-) Vous pourvez enrégistrer un commutateur contre un serveur Asterisk à l'aide d'un username et un secret via SIP. Il faut que le serveur les sache. Vous pouvez poser des questions plus spécifiques à l'égard d'Asterisk. Mais je ne sais rien de carte VOIBE (Google ne donne pas plus que des millions de millions de documents ruisses), donc je ne peux pas vous en aider. Au regard. 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] Asterisk with Dialplan or TrixBox for this case?
From: Brian McEntire [EMAIL PROTECTED] Date: Sat, 24 Mar 2007 13:57:38 -0400 Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be open to other suggestions. I have a Wildcat TDM400 (IIRC) with 2 FXS and 2 FXO ports. Currently I'm terminating a POTS line and a VoicePulse VOIP line (via the supplied adapter) into the FXS ports (forgive me if I confused the FXO/FXS it gets me every time.) I have the dialplan set up to ring all extensions when either incoming line rings. Ring available extensions if one is in use. For dial out, it only dials out the VOIP line unless I override by dialing 9 first (because we pay per call on the POTS line so I want to know I'm doing it rather than have asterisk do it for me if the VOIP line is already in use.) - - - What I'm looking to do is keep the functionality above but drop the POTS line and add a SunRocket line also terminated with a VOIP adapter just like the VoicePulse line. Although the net connection will be a single point of failure, at least I'll have two different VOIP providers for some redundancy. I'd like to: - ring all extensions when a call comes in either VOIP line. - distinctive ring for calls coming in the SunRocket line (which Asterisk will know by the port that the line comes in on.) - do not disturb functionality to disable all extensions from ringing by dialing a *XX number from any phone in the house. Ability to toggle ringing back on easily. - dial out any available line (now that both are VOIP) Easy to do with TrixBox or better off installing the latest Asterisk and doing it through the command line and configuration file interface? If your box has the power to run extra stuff that come with TrixBox and you are sure that doing what you need is easier in TrixBox, there's not much difference. (From your description, the requirements are easily implementable with plain config files.) Thanks! PS - Oddly, the SunRocket VOIP adapter doesn't seem to give a dialtone but a regular old phone works fine when connected to it. Will this cause problems for Asterisk? Asterisk does not have to check dial tone. (But it's a really oddball adapter.) However, if you are going all VoIP, why bother providers that require adapters (thus TDM card)? You can get better result by using providers that transmits voice over IP into your Asterisk and get rid of the TDM card. 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] asterisk: error while loading shared libraries: libiksemel.
From: Dmitri Smirnoff [EMAIL PROTECTED] Date: Sat, 24 Mar 2007 21:11:17 -0400 How I can disable Gtalk Jabber module?Thanks# asterisk -vcasterisk: error while loading shared libraries: libiksemel.so.3: cannot open shared objectfile: No such file or directory===Centos4.4 2.6.9-34.0.2.ELzaptel 1.4.1asterisk 1.4.2iksemel 1.2Dmitri Smirnoff msn: [EMAIL PROTECTED]: 613 693 1299 ext 120 Rerun make menuselect? 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] Noob question regarding PCI 2.x TDM400P Card
From: Barton Fisher [EMAIL PROTECTED] Date: Fri, 23 Mar 2007 09:26:44 -0800 I have some old PC's I want to build as a test box - It's up and running OK now. Now I installed a TDM400P and there is nothing I can do to get the card to come up. My guess is the box is not PCI 2.2 compliant or does it need to be to see the card? I had a similar situation. What I found was: the CMOS setup program had an option to turn PCI 2.2 on or off - default was off. Later motherboards no longer have this. Yuan Liu Thanks, Bart Here's what I know: Processors 1 Model Pentium III (Katmai) CPU Speed 551.37 MHz Cache Size 512 KB System Bogomips 1103.57 PCI Devices - Bridge: Intel Corporation 82371AB/EB/MB PIIX4 ACPI - Ethernet controller: Intel Corporation 82557/8/9 [Ethernet Pro 100] - Host bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/DX Host bridge - IDE interface: Intel Corporation 82371AB/EB/MB PIIX4 IDE - ISA bridge: Intel Corporation 82371AB/EB/MB PIIX4 ISA - PCI bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/DX AGP bridge - USB Controller: Intel Corporation 82371AB/EB/MB PIIX4 USB - VGA compatible controller: Chips and Technologies F69000 HiQVideo ___ --Bandwidth 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] Limit call duration
From: Suity Zsolt [EMAIL PROTECTED] Date: Wed, 21 Mar 2007 14:09:20 +0100 Robert Lister wrote: On Wed, Mar 21, 2007 at 12:56:55PM +0100, Suity Zsolt wrote: Hi everyone, I'm new to Asterisk, but I like it ;o) Have a question to you; How can I limit the incoming call duration? You could use L() flag in when dialing the physical end point. Yuan Liu I think you can say something like: AbsoluteTimeout (or in 1.2x, Set(TIMEOUT(absolute) = seconds) ) See: http://www.voip-info.org/wiki/view/Asterisk+cmd+AbsoluteTimeout Thank you, I will try later today, but I think this is what I looking for. (If I can set it only for external calls) -- Suich ___ --Bandwidth 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] Which parameters of a live Asterisk server wouldyou monitor
From: Olivier [EMAIL PROTECTED] Date: Tue, 20 Mar 2007 09:49:34 +0100 Hi, Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? I was thinking of : - telco lines status (make sure every is up) - registered hardphones If you use VoIP, add data network status (and possibly quality). Yuan Liu - config files backup (compare live and saved configuration files, if files differ, notifies the administration team) - systems variables (disk and CPU) - log files (trigger an alarm for every ERROR or NOTICE message in full logs) What do you think ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Automated Outbound Messaging
From: Rob Schall [EMAIL PROTECTED] Date: Tue, 20 Mar 2007 16:00:01 -0500 Cory Andrews wrote: These folks have 6-8 T's worth of outbound they do on a daily basis, I need an interface that would allow them to stick a comma delimited file or file(s) in every day via FTP, the file would contain call #'s, and some additional variables, and then the Asterisk box would schedule the calls. It would pull a voice file locally and deliver to answering machines or live call recipients. Looks like user interface is not a concern - if they are thinking of FTP text files. In this case, a simple script to kick off some call files should suffice. Won't take a week. (Search for call file.) But having to deal with answering machines is always tricky for any automation. Yuan Liu Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Tuesday, March 20, 2007 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Automated Outbound Messaging Cory Andrews wrote: I have a client application looking for an Asterisk based solution. Client wants to deliver pre-recorded messages for a variety of clients. Wondering if anyone is offering an middleware for Asterisk for management of outbound messaging? Someone can correct me if I'm wrong, but I think a friend of mine mentioned that TrixBox has a gab cast function. It also shouldn't be that difficult to put together a script to do this. I actually have plans to do this myself, but no need for it just yet... If they want a decent interface to see the next caller before calling, you might want to have a database that reads in all the numbers, then users that grab the next non-checked number from the database. This also gives you the option of leaving notes with that call (such as calling back, etc). Then when ready, press the call button which creates a call file. Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] no special context for sip peer
From: Christophorus Laube [EMAIL PROTECTED] Date: Mon, 19 Mar 2007 13:23:34 +0100 Hi list, I want to set up special contexts for every sip user. But a context=XYZ does not help in the perr definition as I have to provide a context in the general section of sip.conf. This is my sip.conf: [general] port=5060 bindaddr=192.168.0.75 disallow=all allow=ulaw allow=alaw context=SIP maxexprirey=3600 defaultexpirey=120 language=de pritrustusercid=yes callerid=asreceived [bob] type=peer A peer can only receive calls from your Asterisk, so there is no way to really invoke that context. A user or friend will use the context to call others through your Asterisk, therefore needs a context. Yuan Liu username=bob host=dynamic secret=nothing context=BOB_SIP qualify=yes canreinvite=yes callingpres=allowed_passed_screen So what am I doing wrong? What do I have to change in order to get my BOB_SIP extensions to work when I am doing a call from this peer? Now * always takes the default context SIP. Regards, Christophorus ___ --Bandwidth 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] zttool always reports OK on TDM400P
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] Date: Mon, 19 Mar 2007 11:26:56 -0500 Yuan LIU wrote: Just noticed that no matter what the line condition is, zttool always reports OK, so it's pretty useless. (In contrast, I'd get Red alert if I unplug the line connecting to an X100P.) I'm using zaptel 1.2.15 on Linux 2.6.15-28 (also tested on 2.6.10). Correct. The TDM400P does not do line detection. Is there an easy way to tell if line is on? When I brought my box to another location, even though CLI says execute Dial(Zap/g1/5551212), call was not delivered. That made me curious. Thanks. 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
[asterisk-users] zttool always reports OK on TDM400P
Just noticed that no matter what the line condition is, zttool always reports OK, so it's pretty useless. (In contrast, I'd get Red alert if I unplug the line connecting to an X100P.) I'm using zaptel 1.2.15 on Linux 2.6.15-28 (also tested on 2.6.10). 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] Only secretary can call the boss, all othersonly reach the
From: Ricardo Carvalho [EMAIL PROTECTED]
Date: Fri, 16 Mar 2007 13:41:49 +
With Ioan suggestion it still doesn't work, because Asterisk still thinks
that the INVITE sent as consequence of the REFER message isn't correlated
with a transferred call coming from the secretary.
Your requirement that the secretary only press the phone's built-in transfer
key makes Asterisk out of the loop, as SIP requires that a proxy do not act
on REFER, and that the agent receiving REFER do not treat REFER-to in any
special way.
To force the secretary to use an Asterisk-defined key sequence, you can
disable the phone's feature keys. Alas! Only if hard phone manufacturers
allow us to reprogram these keys.
On the other hand, you can still use Jonathan's method, to change the boss'
real extension. If there's a special reason why this can't be done, I can
think of a really crooked method to do what you want:
. Retrain your secretary to transfer call to boss to a special extension,
say boss_xfer, instead of to boss_extension.
. In boss_xfer, set some special variable or SIP header before
Dial(Local/[EMAIL PROTECTED]).
. In boss_extension, check for this special viable or SIP header before
really dialing boss' phone.
. Keep boss_xfer a top secret.
Yuan Liu
I've also tried to do it using different contexts, but it still doesn't
work. I've done like this:
[default]
exten = secretary_extension,1,Dial(SIP/secretary_extension)
exten = boss_extension,1,Dial(SIP/secretary_extension)
[secretary]
include = default
exten = boss_extension,1,Dial(SIP/boss_extension)
The problem seems to be that in either case, Asterisk doesn't keep the
state of the call, to know that if transferred from the secretary, the
server should let it pass to the boss and not redirecting it back to the
secretary.
May this be solved with Transfer([Tech/]dest[|options])? And is it the only
way to do it? Can't it be done with normal transfer key that the phones
I've deployed have?
Any other ideas?!
Thanks,
Ricardo.
Ioan Indreias wrote:
Maybe you could use something like:
exten =
boss_ext,1,GotoIf($[${CALLERID(number)}=secretary_ext]?boss:secretary)
exten = boss_ext,n(boss),Dial(SIP/boss_ext)
exten = boss_ext,n(secretary),Dial(SIP/secretary_ext)
## nini @ www.modulo.ro ##
Jonathan k. Creasy wrote:
Why don't you just give the secretary the boss' REAL extension and give a
different extension to the world that just rings the secretary?
-jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho
Sent: Friday, January 26, 2007 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Only secretary can call the boss, all others
only reach the secretary when dial the boss extension
Dear all,
How may I configure my extensions.conf so that only the boss's secretary
can call the boss through his extension, all others when dial his
extension only makes the boss's secretary phone ring, not his. If she
wants, she can transfer the incoming call to the boss dialling his
extension.
I've tried the following, but it doesn't work:
exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)
exten = _boss_extension,1,Dial(SIP/secretary_extension)
This doesn't work because when the secretary tries to transfer the call
to the boss (using her phone's transfer key, not #), one REFER SIP
message is sent back to the caller's phone providing him the new address
for whom the next INVITE should be sent. That INVITE is sent, but when
reaches Asterisk, that INVITE matches this line:
exten = _boss_extension,1,Dial(SIP/secretary_extension)
and not this one:
exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)
Any ideas of how may I solve this issue?
Regards,
Ricardo.
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Re: [asterisk-users] Nomination for Coolest App in 2007
From: Brad Templeton [EMAIL PROTECTED] Date: Fri, 16 Mar 2007 13:37:55 -0700 On Wed, Mar 14, 2007 at 09:37:45AM -0500, Steve Totaro wrote: Another interesting (from an American's perspective anyways) is that inbound calls on cell phones are free. Even if you buy a SIM with a little pre-paid time and use up the time, you can still receive inbound calls for free for a couple months. Inbound calls on cell phones outside North America are alas, not free, though people pretend they are free. They are caller pays for airtime. The only free incoming call systems I have seen are some mobile to mobile free call plans, and a small number of North American mobile plans that, for a flat monthly or daily fee, offer free incoming. Several carriers in Canada offer first-minute incoming free. Quite an interesting concept from consumer's perspective. The caller-pays system found outside North America is, in my view -- though I know some differ -- one of the last, great curses of old world telephony on our new environment. This debate came up in several places and the verdict is not crystal clear. Do you realize that caller-pays system also effectively reduces spam - or at least make it less painful? Especially with SMS, people who carries a mobile phone could easily be targeted by marketers and PAY for it! (I'm starting to see voice telemarketers calling to people's cell phones these days.) Considering per-minute cost in a mobile network is still much higher than that in PSTN, you can't deny advantages of a caller-pays system. Yuan Liu With my VoIP terminators, I can call most of the world's landline's for a price so low I think of it as free, with one exception -- the damn caller-pays cell phones which cost over an order of mangitude more because the fact that the payer doesn't negotiate the price removes the competition that would normally drive the price down. (And has driven it down in the receiver-pays countries.) However, for people in those countries, the bluetooth module does seem like a good idea. Obviously in places with no landlines, but also in places with these bizarre prices, so that if you call one mobile from another mobile, it's cheap, but if you call from a SIP terminator, it's 25 cents/minute. ___ --Bandwidth 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: Which SIP method/option to display a shorttext message
From: Olivier [EMAIL PROTECTED] Date: Thu, 15 Mar 2007 15:21:15 +0100 Hi, After further research, it seems SIP MESSAGE rfc3428) and SIP INFO (rfc2976) methods could be the more relevant for this feature. I'm still wondering whether SIP hardphones or Asterisk implement these methods in such a way you could make a welcome message, for example, appear on you contact phone screen. Cheers There was a thread indicating that you can do that with SendText() with capable hard phones. 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] DNIS/DNID
From: Mark Quitoriano [EMAIL PROTECTED] Date: Thu, 15 Mar 2007 11:59:30 +0800 Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying to send the DNID/DNIS to the PBX here's my dialplan exten = 888111,1,Dial(ZAP/g2) I thought you'd get an error message about the syntax above? If the PBX is configured to take DNIS as DTMF string, D() flag could be used. Yuan Liu exten = 888111,n,Hangup() The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or ZAP/g1 the PBX get the number 1. What should i add to send the extension number as DNID/DNIS? 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] Zaptel version for asterisk 1.2.16
From: Wilson Pickett [EMAIL PROTECTED] Date: Wed, 14 Mar 2007 15:18:35 +0100 I'm used to seeing the same versioning (maybe I've been gone too long) Is zaptel 1.2.15 the right one for asterisk 1.2.16 ? It works. I've tried some other mixes and they also work. 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] Compiling smsq in 1.2
From: Yuan LIU [EMAIL PROTECTED] Date: Tue, 06 Mar 2007 13:58:00 -0800 How to compile smsq in 1.2? It is compile in 1.4 by default. It is included in 1.2.13, but not compiled. Any rule or method to make it? Problem solved after upgrading to 1.2.16. Two points: 1) There was indeed a rule to make smsq - in fact it should be built in a normal compilation. But it did not. To force compilation, use $ make utils/smsq 2) The reason it did not was two folded. First, my Ubuntu installation did not include popt.h, so the above command complained this first. To obtain this header file, package libpopt-dev is required. But even after installing the required package, 1.2.13 still won't compile smsq due to numerous symbol reference errors. To fully install smsq, I upgraded to 1.2.16. Boom! All compiled with no problem. Even on a system that did not have libpopt-dev during first compilation, make utils/smsq did the trick after installing the package. - The compilation process (1.2.13 and 1.2.16) gives no error or warning message when it detects missing popt.h; it very quietly ignored the problem and happily reports a successful build even though a documented component is missing. Hope these notes can help someone else. 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] Noob Question
From: Thomas Patterson [EMAIL PROTECTED]
Date: Mon, 12 Mar 2007 19:03:12 +1300
I have setup my Asterisk server to have 3 outbound routes
1 being for local calls
2 being for toll calls
3 being international call
What I am wanting to do is automaticly setup if you dial a local number it
goes out on the local interface
If you dial a toll call it will go out on the tall provider.
Now for the 3 option I want it to pick up the slack of the other eg if I
have not put the dialing prefix in it will default to this trunk
Just match your local numbering plan. Don't know your country's, but in
North America (NANP), you can do
[general]
NANP = NXXNXX; 10-digit phone number starting with area code
NA_LOCAL = NXX; a North American local #
[outgoing]
exten = _${NA_LOCAL},1,NoOp(Got local number ${EXTEN})
exten = _${NA_LOCAL},n,Dial(Zap/g1/${EXTEN}); G1 is for local
exten = _${NANP),1,Goto(1${EXTEN}); for the lazy people
exten = _1${NANP},1,NoOp(Got toll number ${EXTEN})
exten = _1${NANP},n,Dial(Zap/g2/${EXTEN}); G2 is for toll
exten = _X.,1,NoOp(Likely international number ${EXTEN})
exten = _X.,1,Dial(Zap/g3/${EXTEN}); G3 is for international
Of course the real thing is a bit more complicated, if you want to count for
local toll and toll-free numbers, etc.
Yuan Liu
Any help would be greatfull
Thomas Patterson
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Re: [asterisk-users] play file and action only stop if one definedkey has been p
From: Thomas Winter [EMAIL PROTECTED]
Date: Sat, 10 Mar 2007 09:47:26 +0100
Am Friday 09 March 2007 23:51 schrieb Steve Murphy:
On Fri, 2007-03-09 at 23:01 +0100, Thomas Winter wrote:
I didnt see the option.
The number can be different and is stored in mySQL
exten = ${tmp_var},1,NoOp(INFO key pressed)
exten = ${tmp_var},n,GoTo(s,restart)
Woa! can you really do that? I would have to check the code, but I have
the strong impression that you cannot use a variable in the extension
name field, they are not evaluated, nor are they really evaluatable. All
the extensions in a context are compared when looking for a match to a
target location, but
I know that goto's etc, can use a variable in a reference, but not in a
definition like this.
I can do this, but it is not working as I wrote before.
Then there must be a reason:-) No, Asterisk will not complain about the
syntax - that's probably why you say you can. But you can use CLI show
dialplan to examine the actual dialplan entered into Asterisk's memory.
You'll see that all the lines you used a variable as extension declaration
contains a null string as extension. Or better, use show dialplan
[EMAIL PROTECTED] and realize that it either matches nothing, or
matches some unexpected item. (Replace test-extension with a real value such
as 1234, and your-context with your context name.)
Asterisk does not do dynamic extension assignment (maybe in AEL, but
definitely not in extensions.conf). It interprets all extensions upon
reading extensions.conf.
Hope this helps.
Yuan Liu
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[asterisk-users] Compiling smsq in 1.2
How to compile smsq in 1.2? It is compile in 1.4 by default. It is included in 1.2.13, but not compiled. Any rule or method to make it? 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
[asterisk-users] When does local leg in call file start?
For a simple call file like Channel: Zap/g1/XXX RetryTime: 60 WaitTime: 30 Context: from-file Extension: s Priority: 1 I noticed that [EMAIL PROTECTED] started to execute regardless of the state of the outgoing call. Is this supposed to be? So far I can only set a Wait() in the local leg and hope the remote party picks up soon enough. I thought call file extension will start execution only when the outgoing leg is answered. Or is there some way to detect this? 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] When does local leg in call file start?
From: Doug Lytle [EMAIL PROTECTED] Date: Sun, 04 Mar 2007 13:56:35 -0500 Yuan LIU wrote: I noticed that [EMAIL PROTECTED] started to execute regardless of the state of the outgoing call. Is this supposed to be? So far I can only set a Wait() in the local leg and hope the remote party picks up soon enough. I thought call file extension will start execution only when the outgoing leg is answered. Or is there some way to detect this? If you are dialing via a PRI or a device that supports call supervision, this is the case. If you are using a standard POTS line, the call is assumed answered immediately. This has been covered many times on this list, search the archives for code fragments on how to deal with such a situation. Doug Thanks for the explanation, Doug and Eric. Yes I came across those threads several times, just didn't quite relate to call files. 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
[asterisk-users] Read() status?
Does application Read() return a status? Console displays stuff, but show application read doesn't mention any status variable. 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] Asterisk - e164 (enum) lookup confused
From: Joseph [EMAIL PROTECTED] Date: Sat, 03 Mar 2007 00:30:42 -0700 I would like to implement enum lookup in my dial plan but searching for solution / implementation I'm getting confused what is current standard. On some pages I read that the ENUMLOOKUP is not in development anymore and suggesting on using Enumlookup.agi scrip , some are saying that Asterisk 1.2.0 comes with a new powerful ENUMLOOKUP. So there is probably no need to use this script anymore; so I'm confused as to what should I use. Have you read doc/README.enum (or enum.txt if 1.4) in your source tree? Internet documents could be quite confusing, considering that Google knows little about the age of them. What's been said is that Enumlookup application is being deprecated, and replaced by a powerful ENUMLOOKUP function. Likewise, many other applications are being or have been replaced by functions of similar names. AGI is definitely not the choice in Asteriskland. Yuan Liu Where can I find good Howto (with good explanation)? -- #Joseph ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to fail an AGI
From: Lenz [EMAIL PROTECTED] Date: Sat, 03 Mar 2007 11:37:29 +0100 You could set a dialplan variable in the AGI so that it's pretty easy to tell what happened in the AGI. About the code 0, the funny part is that you see AGI Script completed, returning 0 even if the AGI does not exist, or is not executable. This should be a good candidate for improvement :-) l. Thanks for the enlightenment. Now I know where to look, and found the following from 1.2 show application agi: Returns -1 on hangup (except for DeadAGI) or if application requested hangup, or 0 on non-hangup exit. Apparently this AGISTATUS is a 1.4 thing, and probably still very simplistic. Just wonder why all AGI commands carry sophisticated return codes. Yuan Liu On Sat, 03 Mar 2007 06:28:23 +0100, Yuan LIU [EMAIL PROTECTED] wrote: I mean how do I set failure condition in AGI? My script exits with code 0 upon success, and non-zero when problems occur - the standard *nix way. But Asterisk always report AGI Script completed, returning 0, and AGISTATUS is always SUCCESS. Yuan Liu -- Loway Research - Home of QueueMetrics http://queuemetrics.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] DTMF from TDM400P and X100P
With one IVR payment system, I noticed quite a difference in DTMF transmission between these two cards. The IVR missed nearly all digits from X100P, while receiving digits from TDM fine. Since neither card process or synthesize audio, what can the difference be? (This particular IVR has problem with some regular phone devices, too.) 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] DTMF from TDM400P and X100P
From: Tzafrir Cohen [EMAIL PROTECTED] Date: Fri, 2 Mar 2007 22:14:09 +0200 On Fri, Mar 02, 2007 at 09:36:28AM -0800, Yuan LIU wrote: With one IVR payment system, I noticed quite a difference in DTMF transmission between these two cards. The IVR missed nearly all digits from X100P, while receiving digits from TDM fine. Since neither card process or synthesize audio, what can the difference be? (At least the TDM400P actually has a hardware TDMF detector, but it is not used, AFAIK) (This particular IVR has problem with some regular phone devices, too.) audio quality? You mean from DAC? That could make sense - altough the MODEM function X100P is designed for would require it to be fairly accurate. DTMF itself is one of basic MODEM functions. listen to the audio with e.g. ztmonitor. Thanks for the input. Have yet to put a sound card with TDM. (The IVR in question is not my machine.) I suspect that ztmonitor listens to the digital output, though. Yuan Liu -- 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
